AudioFlinger.cpp revision d96c5724818fb47917bb5e7abe37799735e1ec0e
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// ---------------------------------------------------------------------------- 165 166#ifdef ADD_BATTERY_DATA 167// To collect the amplifier usage 168static void addBatteryData(uint32_t params) { 169 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 170 if (service == NULL) { 171 // it already logged 172 return; 173 } 174 175 service->addBatteryData(params); 176} 177#endif 178 179static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 180{ 181 const hw_module_t *mod; 182 int rc; 183 184 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 185 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 186 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 187 if (rc) { 188 goto out; 189 } 190 rc = audio_hw_device_open(mod, dev); 191 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 192 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 193 if (rc) { 194 goto out; 195 } 196 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 197 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 198 rc = BAD_VALUE; 199 goto out; 200 } 201 return 0; 202 203out: 204 *dev = NULL; 205 return rc; 206} 207 208// ---------------------------------------------------------------------------- 209 210AudioFlinger::AudioFlinger() 211 : BnAudioFlinger(), 212 mPrimaryHardwareDev(NULL), 213 mHardwareStatus(AUDIO_HW_IDLE), 214 mMasterVolume(1.0f), 215 mMasterVolumeSW(1.0f), 216 mMasterVolumeSupportLvl(MVS_NONE), 217 mMasterMute(false), 218 mNextUniqueId(1), 219 mMode(AUDIO_MODE_INVALID), 220 mBtNrecIsOff(false) 221{ 222} 223 224void AudioFlinger::onFirstRef() 225{ 226 int rc = 0; 227 228 Mutex::Autolock _l(mLock); 229 230 /* TODO: move all this work into an Init() function */ 231 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 232 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 233 uint32_t int_val; 234 if (1 == sscanf(val_str, "%u", &int_val)) { 235 mStandbyTimeInNsecs = milliseconds(int_val); 236 ALOGI("Using %u mSec as standby time.", int_val); 237 } else { 238 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 239 ALOGI("Using default %u mSec as standby time.", 240 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 241 } 242 } 243 244 mMode = AUDIO_MODE_NORMAL; 245} 246 247AudioFlinger::~AudioFlinger() 248{ 249 while (!mRecordThreads.isEmpty()) { 250 // closeInput() will remove first entry from mRecordThreads 251 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 252 } 253 while (!mPlaybackThreads.isEmpty()) { 254 // closeOutput() will remove first entry from mPlaybackThreads 255 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 256 } 257 258 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 259 // no mHardwareLock needed, as there are no other references to this 260 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 261 delete mAudioHwDevs.valueAt(i); 262 } 263} 264 265static const char * const audio_interfaces[] = { 266 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 267 AUDIO_HARDWARE_MODULE_ID_A2DP, 268 AUDIO_HARDWARE_MODULE_ID_USB, 269}; 270#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 271 272audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 273{ 274 // if module is 0, the request comes from an old policy manager and we should load 275 // well known modules 276 if (module == 0) { 277 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 278 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 279 loadHwModule_l(audio_interfaces[i]); 280 } 281 } else { 282 // check a match for the requested module handle 283 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 284 if (audioHwdevice != NULL) { 285 return audioHwdevice->hwDevice(); 286 } 287 } 288 // then try to find a module supporting the requested device. 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 295 return NULL; 296} 297 298status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 299{ 300 const size_t SIZE = 256; 301 char buffer[SIZE]; 302 String8 result; 303 304 result.append("Clients:\n"); 305 for (size_t i = 0; i < mClients.size(); ++i) { 306 sp<Client> client = mClients.valueAt(i).promote(); 307 if (client != 0) { 308 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 309 result.append(buffer); 310 } 311 } 312 313 result.append("Global session refs:\n"); 314 result.append(" session pid count\n"); 315 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 316 AudioSessionRef *r = mAudioSessionRefs[i]; 317 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 318 result.append(buffer); 319 } 320 write(fd, result.string(), result.size()); 321 return NO_ERROR; 322} 323 324 325status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 326{ 327 const size_t SIZE = 256; 328 char buffer[SIZE]; 329 String8 result; 330 hardware_call_state hardwareStatus = mHardwareStatus; 331 332 snprintf(buffer, SIZE, "Hardware status: %d\n" 333 "Standby Time mSec: %u\n", 334 hardwareStatus, 335 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 336 result.append(buffer); 337 write(fd, result.string(), result.size()); 338 return NO_ERROR; 339} 340 341status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 342{ 343 const size_t SIZE = 256; 344 char buffer[SIZE]; 345 String8 result; 346 snprintf(buffer, SIZE, "Permission Denial: " 347 "can't dump AudioFlinger from pid=%d, uid=%d\n", 348 IPCThreadState::self()->getCallingPid(), 349 IPCThreadState::self()->getCallingUid()); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352 return NO_ERROR; 353} 354 355static bool tryLock(Mutex& mutex) 356{ 357 bool locked = false; 358 for (int i = 0; i < kDumpLockRetries; ++i) { 359 if (mutex.tryLock() == NO_ERROR) { 360 locked = true; 361 break; 362 } 363 usleep(kDumpLockSleepUs); 364 } 365 return locked; 366} 367 368status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 369{ 370 if (!dumpAllowed()) { 371 dumpPermissionDenial(fd, args); 372 } else { 373 // get state of hardware lock 374 bool hardwareLocked = tryLock(mHardwareLock); 375 if (!hardwareLocked) { 376 String8 result(kHardwareLockedString); 377 write(fd, result.string(), result.size()); 378 } else { 379 mHardwareLock.unlock(); 380 } 381 382 bool locked = tryLock(mLock); 383 384 // failed to lock - AudioFlinger is probably deadlocked 385 if (!locked) { 386 String8 result(kDeadlockedString); 387 write(fd, result.string(), result.size()); 388 } 389 390 dumpClients(fd, args); 391 dumpInternals(fd, args); 392 393 // dump playback threads 394 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 395 mPlaybackThreads.valueAt(i)->dump(fd, args); 396 } 397 398 // dump record threads 399 for (size_t i = 0; i < mRecordThreads.size(); i++) { 400 mRecordThreads.valueAt(i)->dump(fd, args); 401 } 402 403 // dump all hardware devs 404 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 405 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 406 dev->dump(dev, fd); 407 } 408 if (locked) mLock.unlock(); 409 } 410 return NO_ERROR; 411} 412 413sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 414{ 415 // If pid is already in the mClients wp<> map, then use that entry 416 // (for which promote() is always != 0), otherwise create a new entry and Client. 417 sp<Client> client = mClients.valueFor(pid).promote(); 418 if (client == 0) { 419 client = new Client(this, pid); 420 mClients.add(pid, client); 421 } 422 423 return client; 424} 425 426// IAudioFlinger interface 427 428 429sp<IAudioTrack> AudioFlinger::createTrack( 430 pid_t pid, 431 audio_stream_type_t streamType, 432 uint32_t sampleRate, 433 audio_format_t format, 434 audio_channel_mask_t channelMask, 435 int frameCount, 436 IAudioFlinger::track_flags_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 pid_t tid, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 // check if an effect chain with the same session ID is present on another 472 // output thread and move it here. 473 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 474 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 475 if (mPlaybackThreads.keyAt(i) != output) { 476 uint32_t sessions = t->hasAudioSession(*sessionId); 477 if (sessions & PlaybackThread::EFFECT_SESSION) { 478 effectThread = t.get(); 479 break; 480 } 481 } 482 } 483 lSessionId = *sessionId; 484 } else { 485 // if no audio session id is provided, create one here 486 lSessionId = nextUniqueId(); 487 if (sessionId != NULL) { 488 *sessionId = lSessionId; 489 } 490 } 491 ALOGV("createTrack() lSessionId: %d", lSessionId); 492 493 track = thread->createTrack_l(client, streamType, sampleRate, format, 494 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 495 496 // move effect chain to this output thread if an effect on same session was waiting 497 // for a track to be created 498 if (lStatus == NO_ERROR && effectThread != NULL) { 499 Mutex::Autolock _dl(thread->mLock); 500 Mutex::Autolock _sl(effectThread->mLock); 501 moveEffectChain_l(lSessionId, effectThread, thread, true); 502 } 503 504 // Look for sync events awaiting for a session to be used. 505 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 506 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 507 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 508 if (lStatus == NO_ERROR) { 509 track->setSyncEvent(mPendingSyncEvents[i]); 510 } else { 511 mPendingSyncEvents[i]->cancel(); 512 } 513 mPendingSyncEvents.removeAt(i); 514 i--; 515 } 516 } 517 } 518 } 519 if (lStatus == NO_ERROR) { 520 trackHandle = new TrackHandle(track); 521 } else { 522 // remove local strong reference to Client before deleting the Track so that the Client 523 // destructor is called by the TrackBase destructor with mLock held 524 client.clear(); 525 track.clear(); 526 } 527 528Exit: 529 if (status != NULL) { 530 *status = lStatus; 531 } 532 return trackHandle; 533} 534 535uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("sampleRate() unknown thread %d", output); 541 return 0; 542 } 543 return thread->sampleRate(); 544} 545 546int AudioFlinger::channelCount(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("channelCount() unknown thread %d", output); 552 return 0; 553 } 554 return thread->channelCount(); 555} 556 557audio_format_t AudioFlinger::format(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("format() unknown thread %d", output); 563 return AUDIO_FORMAT_INVALID; 564 } 565 return thread->format(); 566} 567 568size_t AudioFlinger::frameCount(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("frameCount() unknown thread %d", output); 574 return 0; 575 } 576 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 577 // should examine all callers and fix them to handle smaller counts 578 return thread->frameCount(); 579} 580 581uint32_t AudioFlinger::latency(audio_io_handle_t output) const 582{ 583 Mutex::Autolock _l(mLock); 584 PlaybackThread *thread = checkPlaybackThread_l(output); 585 if (thread == NULL) { 586 ALOGW("latency() unknown thread %d", output); 587 return 0; 588 } 589 return thread->latency(); 590} 591 592status_t AudioFlinger::setMasterVolume(float value) 593{ 594 status_t ret = initCheck(); 595 if (ret != NO_ERROR) { 596 return ret; 597 } 598 599 // check calling permissions 600 if (!settingsAllowed()) { 601 return PERMISSION_DENIED; 602 } 603 604 float swmv = value; 605 606 Mutex::Autolock _l(mLock); 607 608 // when hw supports master volume, don't scale in sw mixer 609 if (MVS_NONE != mMasterVolumeSupportLvl) { 610 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 611 AutoMutex lock(mHardwareLock); 612 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 613 614 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 615 if (NULL != dev->set_master_volume) { 616 dev->set_master_volume(dev, value); 617 } 618 mHardwareStatus = AUDIO_HW_IDLE; 619 } 620 621 swmv = 1.0; 622 } 623 624 mMasterVolume = value; 625 mMasterVolumeSW = swmv; 626 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 627 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 628 629 return NO_ERROR; 630} 631 632status_t AudioFlinger::setMode(audio_mode_t mode) 633{ 634 status_t ret = initCheck(); 635 if (ret != NO_ERROR) { 636 return ret; 637 } 638 639 // check calling permissions 640 if (!settingsAllowed()) { 641 return PERMISSION_DENIED; 642 } 643 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 644 ALOGW("Illegal value: setMode(%d)", mode); 645 return BAD_VALUE; 646 } 647 648 { // scope for the lock 649 AutoMutex lock(mHardwareLock); 650 mHardwareStatus = AUDIO_HW_SET_MODE; 651 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 652 mHardwareStatus = AUDIO_HW_IDLE; 653 } 654 655 if (NO_ERROR == ret) { 656 Mutex::Autolock _l(mLock); 657 mMode = mode; 658 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 659 mPlaybackThreads.valueAt(i)->setMode(mode); 660 } 661 662 return ret; 663} 664 665status_t AudioFlinger::setMicMute(bool state) 666{ 667 status_t ret = initCheck(); 668 if (ret != NO_ERROR) { 669 return ret; 670 } 671 672 // check calling permissions 673 if (!settingsAllowed()) { 674 return PERMISSION_DENIED; 675 } 676 677 AutoMutex lock(mHardwareLock); 678 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 679 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 680 mHardwareStatus = AUDIO_HW_IDLE; 681 return ret; 682} 683 684bool AudioFlinger::getMicMute() const 685{ 686 status_t ret = initCheck(); 687 if (ret != NO_ERROR) { 688 return false; 689 } 690 691 bool state = AUDIO_MODE_INVALID; 692 AutoMutex lock(mHardwareLock); 693 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 694 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 695 mHardwareStatus = AUDIO_HW_IDLE; 696 return state; 697} 698 699status_t AudioFlinger::setMasterMute(bool muted) 700{ 701 // check calling permissions 702 if (!settingsAllowed()) { 703 return PERMISSION_DENIED; 704 } 705 706 Mutex::Autolock _l(mLock); 707 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 708 mMasterMute = muted; 709 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 710 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 711 712 return NO_ERROR; 713} 714 715float AudioFlinger::masterVolume() const 716{ 717 Mutex::Autolock _l(mLock); 718 return masterVolume_l(); 719} 720 721float AudioFlinger::masterVolumeSW() const 722{ 723 Mutex::Autolock _l(mLock); 724 return masterVolumeSW_l(); 725} 726 727bool AudioFlinger::masterMute() const 728{ 729 Mutex::Autolock _l(mLock); 730 return masterMute_l(); 731} 732 733float AudioFlinger::masterVolume_l() const 734{ 735 if (MVS_FULL == mMasterVolumeSupportLvl) { 736 float ret_val; 737 AutoMutex lock(mHardwareLock); 738 739 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 740 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 741 (NULL != mPrimaryHardwareDev->get_master_volume), 742 "can't get master volume"); 743 744 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 745 mHardwareStatus = AUDIO_HW_IDLE; 746 return ret_val; 747 } 748 749 return mMasterVolume; 750} 751 752status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 753 audio_io_handle_t output) 754{ 755 // check calling permissions 756 if (!settingsAllowed()) { 757 return PERMISSION_DENIED; 758 } 759 760 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 761 ALOGE("setStreamVolume() invalid stream %d", stream); 762 return BAD_VALUE; 763 } 764 765 AutoMutex lock(mLock); 766 PlaybackThread *thread = NULL; 767 if (output) { 768 thread = checkPlaybackThread_l(output); 769 if (thread == NULL) { 770 return BAD_VALUE; 771 } 772 } 773 774 mStreamTypes[stream].volume = value; 775 776 if (thread == NULL) { 777 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 778 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 779 } 780 } else { 781 thread->setStreamVolume(stream, value); 782 } 783 784 return NO_ERROR; 785} 786 787status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 788{ 789 // check calling permissions 790 if (!settingsAllowed()) { 791 return PERMISSION_DENIED; 792 } 793 794 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 795 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 796 ALOGE("setStreamMute() invalid stream %d", stream); 797 return BAD_VALUE; 798 } 799 800 AutoMutex lock(mLock); 801 mStreamTypes[stream].mute = muted; 802 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 803 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 804 805 return NO_ERROR; 806} 807 808float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 809{ 810 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 811 return 0.0f; 812 } 813 814 AutoMutex lock(mLock); 815 float volume; 816 if (output) { 817 PlaybackThread *thread = checkPlaybackThread_l(output); 818 if (thread == NULL) { 819 return 0.0f; 820 } 821 volume = thread->streamVolume(stream); 822 } else { 823 volume = streamVolume_l(stream); 824 } 825 826 return volume; 827} 828 829bool AudioFlinger::streamMute(audio_stream_type_t stream) const 830{ 831 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 832 return true; 833 } 834 835 AutoMutex lock(mLock); 836 return streamMute_l(stream); 837} 838 839status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 840{ 841 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 842 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 843 // check calling permissions 844 if (!settingsAllowed()) { 845 return PERMISSION_DENIED; 846 } 847 848 // ioHandle == 0 means the parameters are global to the audio hardware interface 849 if (ioHandle == 0) { 850 Mutex::Autolock _l(mLock); 851 status_t final_result = NO_ERROR; 852 { 853 AutoMutex lock(mHardwareLock); 854 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 855 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 856 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 857 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 858 final_result = result ?: final_result; 859 } 860 mHardwareStatus = AUDIO_HW_IDLE; 861 } 862 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 863 AudioParameter param = AudioParameter(keyValuePairs); 864 String8 value; 865 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 866 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 867 if (mBtNrecIsOff != btNrecIsOff) { 868 for (size_t i = 0; i < mRecordThreads.size(); i++) { 869 sp<RecordThread> thread = mRecordThreads.valueAt(i); 870 RecordThread::RecordTrack *track = thread->track(); 871 if (track != NULL) { 872 audio_devices_t device = (audio_devices_t)( 873 thread->device() & AUDIO_DEVICE_IN_ALL); 874 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 875 thread->setEffectSuspended(FX_IID_AEC, 876 suspend, 877 track->sessionId()); 878 thread->setEffectSuspended(FX_IID_NS, 879 suspend, 880 track->sessionId()); 881 } 882 } 883 mBtNrecIsOff = btNrecIsOff; 884 } 885 } 886 String8 screenState; 887 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 888 bool isOff = screenState == "off"; 889 if (isOff != (gScreenState & 1)) { 890 gScreenState = ((gScreenState & ~1) + 2) | isOff; 891 } 892 } 893 return final_result; 894 } 895 896 // hold a strong ref on thread in case closeOutput() or closeInput() is called 897 // and the thread is exited once the lock is released 898 sp<ThreadBase> thread; 899 { 900 Mutex::Autolock _l(mLock); 901 thread = checkPlaybackThread_l(ioHandle); 902 if (thread == 0) { 903 thread = checkRecordThread_l(ioHandle); 904 } else if (thread == primaryPlaybackThread_l()) { 905 // indicate output device change to all input threads for pre processing 906 AudioParameter param = AudioParameter(keyValuePairs); 907 int value; 908 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 909 (value != 0)) { 910 for (size_t i = 0; i < mRecordThreads.size(); i++) { 911 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 912 } 913 } 914 } 915 } 916 if (thread != 0) { 917 return thread->setParameters(keyValuePairs); 918 } 919 return BAD_VALUE; 920} 921 922String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 923{ 924// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 925// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 926 927 Mutex::Autolock _l(mLock); 928 929 if (ioHandle == 0) { 930 String8 out_s8; 931 932 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 933 char *s; 934 { 935 AutoMutex lock(mHardwareLock); 936 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 937 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 938 s = dev->get_parameters(dev, keys.string()); 939 mHardwareStatus = AUDIO_HW_IDLE; 940 } 941 out_s8 += String8(s ? s : ""); 942 free(s); 943 } 944 return out_s8; 945 } 946 947 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 948 if (playbackThread != NULL) { 949 return playbackThread->getParameters(keys); 950 } 951 RecordThread *recordThread = checkRecordThread_l(ioHandle); 952 if (recordThread != NULL) { 953 return recordThread->getParameters(keys); 954 } 955 return String8(""); 956} 957 958size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 959 audio_channel_mask_t channelMask) const 960{ 961 status_t ret = initCheck(); 962 if (ret != NO_ERROR) { 963 return 0; 964 } 965 966 AutoMutex lock(mHardwareLock); 967 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 968 struct audio_config config = { 969 sample_rate: sampleRate, 970 channel_mask: channelMask, 971 format: format, 972 }; 973 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 974 mHardwareStatus = AUDIO_HW_IDLE; 975 return size; 976} 977 978unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 979{ 980 Mutex::Autolock _l(mLock); 981 982 RecordThread *recordThread = checkRecordThread_l(ioHandle); 983 if (recordThread != NULL) { 984 return recordThread->getInputFramesLost(); 985 } 986 return 0; 987} 988 989status_t AudioFlinger::setVoiceVolume(float value) 990{ 991 status_t ret = initCheck(); 992 if (ret != NO_ERROR) { 993 return ret; 994 } 995 996 // check calling permissions 997 if (!settingsAllowed()) { 998 return PERMISSION_DENIED; 999 } 1000 1001 AutoMutex lock(mHardwareLock); 1002 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1003 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1004 mHardwareStatus = AUDIO_HW_IDLE; 1005 1006 return ret; 1007} 1008 1009status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1010 audio_io_handle_t output) const 1011{ 1012 status_t status; 1013 1014 Mutex::Autolock _l(mLock); 1015 1016 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1017 if (playbackThread != NULL) { 1018 return playbackThread->getRenderPosition(halFrames, dspFrames); 1019 } 1020 1021 return BAD_VALUE; 1022} 1023 1024void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1025{ 1026 1027 Mutex::Autolock _l(mLock); 1028 1029 pid_t pid = IPCThreadState::self()->getCallingPid(); 1030 if (mNotificationClients.indexOfKey(pid) < 0) { 1031 sp<NotificationClient> notificationClient = new NotificationClient(this, 1032 client, 1033 pid); 1034 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1035 1036 mNotificationClients.add(pid, notificationClient); 1037 1038 sp<IBinder> binder = client->asBinder(); 1039 binder->linkToDeath(notificationClient); 1040 1041 // the config change is always sent from playback or record threads to avoid deadlock 1042 // with AudioSystem::gLock 1043 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1044 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1045 } 1046 1047 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1048 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1049 } 1050 } 1051} 1052 1053void AudioFlinger::removeNotificationClient(pid_t pid) 1054{ 1055 Mutex::Autolock _l(mLock); 1056 1057 mNotificationClients.removeItem(pid); 1058 1059 ALOGV("%d died, releasing its sessions", pid); 1060 size_t num = mAudioSessionRefs.size(); 1061 bool removed = false; 1062 for (size_t i = 0; i< num; ) { 1063 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1064 ALOGV(" pid %d @ %d", ref->mPid, i); 1065 if (ref->mPid == pid) { 1066 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1067 mAudioSessionRefs.removeAt(i); 1068 delete ref; 1069 removed = true; 1070 num--; 1071 } else { 1072 i++; 1073 } 1074 } 1075 if (removed) { 1076 purgeStaleEffects_l(); 1077 } 1078} 1079 1080// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1081void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1082{ 1083 size_t size = mNotificationClients.size(); 1084 for (size_t i = 0; i < size; i++) { 1085 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1086 param2); 1087 } 1088} 1089 1090// removeClient_l() must be called with AudioFlinger::mLock held 1091void AudioFlinger::removeClient_l(pid_t pid) 1092{ 1093 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1094 mClients.removeItem(pid); 1095} 1096 1097// getEffectThread_l() must be called with AudioFlinger::mLock held 1098sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1099{ 1100 sp<PlaybackThread> thread; 1101 1102 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1103 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1104 ALOG_ASSERT(thread == 0); 1105 thread = mPlaybackThreads.valueAt(i); 1106 } 1107 } 1108 1109 return thread; 1110} 1111 1112// ---------------------------------------------------------------------------- 1113 1114AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1115 uint32_t device, type_t type) 1116 : Thread(false), 1117 mType(type), 1118 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1119 // mChannelMask 1120 mChannelCount(0), 1121 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1122 mParamStatus(NO_ERROR), 1123 mStandby(false), mDevice((audio_devices_t) device), mId(id), 1124 mDeathRecipient(new PMDeathRecipient(this)) 1125{ 1126} 1127 1128AudioFlinger::ThreadBase::~ThreadBase() 1129{ 1130 mParamCond.broadcast(); 1131 // do not lock the mutex in destructor 1132 releaseWakeLock_l(); 1133 if (mPowerManager != 0) { 1134 sp<IBinder> binder = mPowerManager->asBinder(); 1135 binder->unlinkToDeath(mDeathRecipient); 1136 } 1137} 1138 1139void AudioFlinger::ThreadBase::exit() 1140{ 1141 ALOGV("ThreadBase::exit"); 1142 { 1143 // This lock prevents the following race in thread (uniprocessor for illustration): 1144 // if (!exitPending()) { 1145 // // context switch from here to exit() 1146 // // exit() calls requestExit(), what exitPending() observes 1147 // // exit() calls signal(), which is dropped since no waiters 1148 // // context switch back from exit() to here 1149 // mWaitWorkCV.wait(...); 1150 // // now thread is hung 1151 // } 1152 AutoMutex lock(mLock); 1153 requestExit(); 1154 mWaitWorkCV.signal(); 1155 } 1156 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1157 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1158 requestExitAndWait(); 1159} 1160 1161status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1162{ 1163 status_t status; 1164 1165 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1166 Mutex::Autolock _l(mLock); 1167 1168 mNewParameters.add(keyValuePairs); 1169 mWaitWorkCV.signal(); 1170 // wait condition with timeout in case the thread loop has exited 1171 // before the request could be processed 1172 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1173 status = mParamStatus; 1174 mWaitWorkCV.signal(); 1175 } else { 1176 status = TIMED_OUT; 1177 } 1178 return status; 1179} 1180 1181void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1182{ 1183 Mutex::Autolock _l(mLock); 1184 sendConfigEvent_l(event, param); 1185} 1186 1187// sendConfigEvent_l() must be called with ThreadBase::mLock held 1188void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1189{ 1190 ConfigEvent configEvent; 1191 configEvent.mEvent = event; 1192 configEvent.mParam = param; 1193 mConfigEvents.add(configEvent); 1194 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1195 mWaitWorkCV.signal(); 1196} 1197 1198void AudioFlinger::ThreadBase::processConfigEvents() 1199{ 1200 mLock.lock(); 1201 while (!mConfigEvents.isEmpty()) { 1202 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1203 ConfigEvent configEvent = mConfigEvents[0]; 1204 mConfigEvents.removeAt(0); 1205 // release mLock before locking AudioFlinger mLock: lock order is always 1206 // AudioFlinger then ThreadBase to avoid cross deadlock 1207 mLock.unlock(); 1208 mAudioFlinger->mLock.lock(); 1209 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1210 mAudioFlinger->mLock.unlock(); 1211 mLock.lock(); 1212 } 1213 mLock.unlock(); 1214} 1215 1216status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1217{ 1218 const size_t SIZE = 256; 1219 char buffer[SIZE]; 1220 String8 result; 1221 1222 bool locked = tryLock(mLock); 1223 if (!locked) { 1224 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1225 write(fd, buffer, strlen(buffer)); 1226 } 1227 1228 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1229 result.append(buffer); 1230 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1231 result.append(buffer); 1232 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1233 result.append(buffer); 1234 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1235 result.append(buffer); 1236 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1237 result.append(buffer); 1238 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1239 result.append(buffer); 1240 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1241 result.append(buffer); 1242 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1243 result.append(buffer); 1244 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1245 result.append(buffer); 1246 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1247 result.append(buffer); 1248 1249 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1250 result.append(buffer); 1251 result.append(" Index Command"); 1252 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1253 snprintf(buffer, SIZE, "\n %02d ", i); 1254 result.append(buffer); 1255 result.append(mNewParameters[i]); 1256 } 1257 1258 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1259 result.append(buffer); 1260 snprintf(buffer, SIZE, " Index event param\n"); 1261 result.append(buffer); 1262 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1263 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1264 result.append(buffer); 1265 } 1266 result.append("\n"); 1267 1268 write(fd, result.string(), result.size()); 1269 1270 if (locked) { 1271 mLock.unlock(); 1272 } 1273 return NO_ERROR; 1274} 1275 1276status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1277{ 1278 const size_t SIZE = 256; 1279 char buffer[SIZE]; 1280 String8 result; 1281 1282 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1283 write(fd, buffer, strlen(buffer)); 1284 1285 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1286 sp<EffectChain> chain = mEffectChains[i]; 1287 if (chain != 0) { 1288 chain->dump(fd, args); 1289 } 1290 } 1291 return NO_ERROR; 1292} 1293 1294void AudioFlinger::ThreadBase::acquireWakeLock() 1295{ 1296 Mutex::Autolock _l(mLock); 1297 acquireWakeLock_l(); 1298} 1299 1300void AudioFlinger::ThreadBase::acquireWakeLock_l() 1301{ 1302 if (mPowerManager == 0) { 1303 // use checkService() to avoid blocking if power service is not up yet 1304 sp<IBinder> binder = 1305 defaultServiceManager()->checkService(String16("power")); 1306 if (binder == 0) { 1307 ALOGW("Thread %s cannot connect to the power manager service", mName); 1308 } else { 1309 mPowerManager = interface_cast<IPowerManager>(binder); 1310 binder->linkToDeath(mDeathRecipient); 1311 } 1312 } 1313 if (mPowerManager != 0) { 1314 sp<IBinder> binder = new BBinder(); 1315 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1316 binder, 1317 String16(mName)); 1318 if (status == NO_ERROR) { 1319 mWakeLockToken = binder; 1320 } 1321 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1322 } 1323} 1324 1325void AudioFlinger::ThreadBase::releaseWakeLock() 1326{ 1327 Mutex::Autolock _l(mLock); 1328 releaseWakeLock_l(); 1329} 1330 1331void AudioFlinger::ThreadBase::releaseWakeLock_l() 1332{ 1333 if (mWakeLockToken != 0) { 1334 ALOGV("releaseWakeLock_l() %s", mName); 1335 if (mPowerManager != 0) { 1336 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1337 } 1338 mWakeLockToken.clear(); 1339 } 1340} 1341 1342void AudioFlinger::ThreadBase::clearPowerManager() 1343{ 1344 Mutex::Autolock _l(mLock); 1345 releaseWakeLock_l(); 1346 mPowerManager.clear(); 1347} 1348 1349void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1350{ 1351 sp<ThreadBase> thread = mThread.promote(); 1352 if (thread != 0) { 1353 thread->clearPowerManager(); 1354 } 1355 ALOGW("power manager service died !!!"); 1356} 1357 1358void AudioFlinger::ThreadBase::setEffectSuspended( 1359 const effect_uuid_t *type, bool suspend, int sessionId) 1360{ 1361 Mutex::Autolock _l(mLock); 1362 setEffectSuspended_l(type, suspend, sessionId); 1363} 1364 1365void AudioFlinger::ThreadBase::setEffectSuspended_l( 1366 const effect_uuid_t *type, bool suspend, int sessionId) 1367{ 1368 sp<EffectChain> chain = getEffectChain_l(sessionId); 1369 if (chain != 0) { 1370 if (type != NULL) { 1371 chain->setEffectSuspended_l(type, suspend); 1372 } else { 1373 chain->setEffectSuspendedAll_l(suspend); 1374 } 1375 } 1376 1377 updateSuspendedSessions_l(type, suspend, sessionId); 1378} 1379 1380void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1381{ 1382 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1383 if (index < 0) { 1384 return; 1385 } 1386 1387 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1388 mSuspendedSessions.editValueAt(index); 1389 1390 for (size_t i = 0; i < sessionEffects.size(); i++) { 1391 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1392 for (int j = 0; j < desc->mRefCount; j++) { 1393 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1394 chain->setEffectSuspendedAll_l(true); 1395 } else { 1396 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1397 desc->mType.timeLow); 1398 chain->setEffectSuspended_l(&desc->mType, true); 1399 } 1400 } 1401 } 1402} 1403 1404void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1405 bool suspend, 1406 int sessionId) 1407{ 1408 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1409 1410 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1411 1412 if (suspend) { 1413 if (index >= 0) { 1414 sessionEffects = mSuspendedSessions.editValueAt(index); 1415 } else { 1416 mSuspendedSessions.add(sessionId, sessionEffects); 1417 } 1418 } else { 1419 if (index < 0) { 1420 return; 1421 } 1422 sessionEffects = mSuspendedSessions.editValueAt(index); 1423 } 1424 1425 1426 int key = EffectChain::kKeyForSuspendAll; 1427 if (type != NULL) { 1428 key = type->timeLow; 1429 } 1430 index = sessionEffects.indexOfKey(key); 1431 1432 sp<SuspendedSessionDesc> desc; 1433 if (suspend) { 1434 if (index >= 0) { 1435 desc = sessionEffects.valueAt(index); 1436 } else { 1437 desc = new SuspendedSessionDesc(); 1438 if (type != NULL) { 1439 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1440 } 1441 sessionEffects.add(key, desc); 1442 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1443 } 1444 desc->mRefCount++; 1445 } else { 1446 if (index < 0) { 1447 return; 1448 } 1449 desc = sessionEffects.valueAt(index); 1450 if (--desc->mRefCount == 0) { 1451 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1452 sessionEffects.removeItemsAt(index); 1453 if (sessionEffects.isEmpty()) { 1454 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1455 sessionId); 1456 mSuspendedSessions.removeItem(sessionId); 1457 } 1458 } 1459 } 1460 if (!sessionEffects.isEmpty()) { 1461 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1462 } 1463} 1464 1465void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1466 bool enabled, 1467 int sessionId) 1468{ 1469 Mutex::Autolock _l(mLock); 1470 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1471} 1472 1473void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1474 bool enabled, 1475 int sessionId) 1476{ 1477 if (mType != RECORD) { 1478 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1479 // another session. This gives the priority to well behaved effect control panels 1480 // and applications not using global effects. 1481 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1482 // global effects 1483 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1484 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1485 } 1486 } 1487 1488 sp<EffectChain> chain = getEffectChain_l(sessionId); 1489 if (chain != 0) { 1490 chain->checkSuspendOnEffectEnabled(effect, enabled); 1491 } 1492} 1493 1494// ---------------------------------------------------------------------------- 1495 1496AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1497 AudioStreamOut* output, 1498 audio_io_handle_t id, 1499 uint32_t device, 1500 type_t type) 1501 : ThreadBase(audioFlinger, id, device, type), 1502 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1503 // Assumes constructor is called by AudioFlinger with it's mLock held, 1504 // but it would be safer to explicitly pass initial masterMute as parameter 1505 mMasterMute(audioFlinger->masterMute_l()), 1506 // mStreamTypes[] initialized in constructor body 1507 mOutput(output), 1508 // Assumes constructor is called by AudioFlinger with it's mLock held, 1509 // but it would be safer to explicitly pass initial masterVolume as parameter 1510 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1511 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1512 mMixerStatus(MIXER_IDLE), 1513 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1514 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1515 mScreenState(gScreenState), 1516 // index 0 is reserved for normal mixer's submix 1517 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1518{ 1519 snprintf(mName, kNameLength, "AudioOut_%X", id); 1520 1521 readOutputParameters(); 1522 1523 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1524 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1525 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1526 stream = (audio_stream_type_t) (stream + 1)) { 1527 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1528 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1529 } 1530 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1531 // because mAudioFlinger doesn't have one to copy from 1532} 1533 1534AudioFlinger::PlaybackThread::~PlaybackThread() 1535{ 1536 delete [] mMixBuffer; 1537} 1538 1539status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1540{ 1541 dumpInternals(fd, args); 1542 dumpTracks(fd, args); 1543 dumpEffectChains(fd, args); 1544 return NO_ERROR; 1545} 1546 1547status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1548{ 1549 const size_t SIZE = 256; 1550 char buffer[SIZE]; 1551 String8 result; 1552 1553 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1554 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1555 const stream_type_t *st = &mStreamTypes[i]; 1556 if (i > 0) { 1557 result.appendFormat(", "); 1558 } 1559 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1560 if (st->mute) { 1561 result.append("M"); 1562 } 1563 } 1564 result.append("\n"); 1565 write(fd, result.string(), result.length()); 1566 result.clear(); 1567 1568 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1569 result.append(buffer); 1570 Track::appendDumpHeader(result); 1571 for (size_t i = 0; i < mTracks.size(); ++i) { 1572 sp<Track> track = mTracks[i]; 1573 if (track != 0) { 1574 track->dump(buffer, SIZE); 1575 result.append(buffer); 1576 } 1577 } 1578 1579 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1580 result.append(buffer); 1581 Track::appendDumpHeader(result); 1582 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1583 sp<Track> track = mActiveTracks[i].promote(); 1584 if (track != 0) { 1585 track->dump(buffer, SIZE); 1586 result.append(buffer); 1587 } 1588 } 1589 write(fd, result.string(), result.size()); 1590 1591 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1592 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1593 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1594 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1595 1596 return NO_ERROR; 1597} 1598 1599status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1600{ 1601 const size_t SIZE = 256; 1602 char buffer[SIZE]; 1603 String8 result; 1604 1605 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1606 result.append(buffer); 1607 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1608 result.append(buffer); 1609 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1610 result.append(buffer); 1611 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1612 result.append(buffer); 1613 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1614 result.append(buffer); 1615 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1616 result.append(buffer); 1617 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1618 result.append(buffer); 1619 write(fd, result.string(), result.size()); 1620 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1621 1622 dumpBase(fd, args); 1623 1624 return NO_ERROR; 1625} 1626 1627// Thread virtuals 1628status_t AudioFlinger::PlaybackThread::readyToRun() 1629{ 1630 status_t status = initCheck(); 1631 if (status == NO_ERROR) { 1632 ALOGI("AudioFlinger's thread %p ready to run", this); 1633 } else { 1634 ALOGE("No working audio driver found."); 1635 } 1636 return status; 1637} 1638 1639void AudioFlinger::PlaybackThread::onFirstRef() 1640{ 1641 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1642} 1643 1644// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1645sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1646 const sp<AudioFlinger::Client>& client, 1647 audio_stream_type_t streamType, 1648 uint32_t sampleRate, 1649 audio_format_t format, 1650 audio_channel_mask_t channelMask, 1651 int frameCount, 1652 const sp<IMemory>& sharedBuffer, 1653 int sessionId, 1654 IAudioFlinger::track_flags_t flags, 1655 pid_t tid, 1656 status_t *status) 1657{ 1658 sp<Track> track; 1659 status_t lStatus; 1660 1661 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1662 1663 // client expresses a preference for FAST, but we get the final say 1664 if (flags & IAudioFlinger::TRACK_FAST) { 1665 if ( 1666 // not timed 1667 (!isTimed) && 1668 // either of these use cases: 1669 ( 1670 // use case 1: shared buffer with any frame count 1671 ( 1672 (sharedBuffer != 0) 1673 ) || 1674 // use case 2: callback handler and frame count is default or at least as large as HAL 1675 ( 1676 (tid != -1) && 1677 ((frameCount == 0) || 1678 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1679 ) 1680 ) && 1681 // PCM data 1682 audio_is_linear_pcm(format) && 1683 // mono or stereo 1684 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1685 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1686#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1687 // hardware sample rate 1688 (sampleRate == mSampleRate) && 1689#endif 1690 // normal mixer has an associated fast mixer 1691 hasFastMixer() && 1692 // there are sufficient fast track slots available 1693 (mFastTrackAvailMask != 0) 1694 // FIXME test that MixerThread for this fast track has a capable output HAL 1695 // FIXME add a permission test also? 1696 ) { 1697 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1698 if (frameCount == 0) { 1699 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1700 } 1701 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1702 frameCount, mFrameCount); 1703 } else { 1704 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1705 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1706 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1707 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1708 audio_is_linear_pcm(format), 1709 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1710 flags &= ~IAudioFlinger::TRACK_FAST; 1711 // For compatibility with AudioTrack calculation, buffer depth is forced 1712 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1713 // This is probably too conservative, but legacy application code may depend on it. 1714 // If you change this calculation, also review the start threshold which is related. 1715 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1716 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1717 if (minBufCount < 2) { 1718 minBufCount = 2; 1719 } 1720 int minFrameCount = mNormalFrameCount * minBufCount; 1721 if (frameCount < minFrameCount) { 1722 frameCount = minFrameCount; 1723 } 1724 } 1725 } 1726 1727 if (mType == DIRECT) { 1728 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1729 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1730 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1731 "for output %p with format %d", 1732 sampleRate, format, channelMask, mOutput, mFormat); 1733 lStatus = BAD_VALUE; 1734 goto Exit; 1735 } 1736 } 1737 } else { 1738 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1739 if (sampleRate > mSampleRate*2) { 1740 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1741 lStatus = BAD_VALUE; 1742 goto Exit; 1743 } 1744 } 1745 1746 lStatus = initCheck(); 1747 if (lStatus != NO_ERROR) { 1748 ALOGE("Audio driver not initialized."); 1749 goto Exit; 1750 } 1751 1752 { // scope for mLock 1753 Mutex::Autolock _l(mLock); 1754 1755 // all tracks in same audio session must share the same routing strategy otherwise 1756 // conflicts will happen when tracks are moved from one output to another by audio policy 1757 // manager 1758 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1759 for (size_t i = 0; i < mTracks.size(); ++i) { 1760 sp<Track> t = mTracks[i]; 1761 if (t != 0 && !t->isOutputTrack()) { 1762 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1763 if (sessionId == t->sessionId() && strategy != actual) { 1764 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1765 strategy, actual); 1766 lStatus = BAD_VALUE; 1767 goto Exit; 1768 } 1769 } 1770 } 1771 1772 if (!isTimed) { 1773 track = new Track(this, client, streamType, sampleRate, format, 1774 channelMask, frameCount, sharedBuffer, sessionId, flags); 1775 } else { 1776 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1777 channelMask, frameCount, sharedBuffer, sessionId); 1778 } 1779 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1780 lStatus = NO_MEMORY; 1781 goto Exit; 1782 } 1783 mTracks.add(track); 1784 1785 sp<EffectChain> chain = getEffectChain_l(sessionId); 1786 if (chain != 0) { 1787 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1788 track->setMainBuffer(chain->inBuffer()); 1789 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1790 chain->incTrackCnt(); 1791 } 1792 } 1793 1794 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1795 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1796 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1797 // so ask activity manager to do this on our behalf 1798 int err = requestPriority(callingPid, tid, 1); 1799 if (err != 0) { 1800 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1801 1, callingPid, tid, err); 1802 } 1803 } 1804 1805 lStatus = NO_ERROR; 1806 1807Exit: 1808 if (status) { 1809 *status = lStatus; 1810 } 1811 return track; 1812} 1813 1814uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1815{ 1816 if (mFastMixer != NULL) { 1817 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1818 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1819 } 1820 return latency; 1821} 1822 1823uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1824{ 1825 return latency; 1826} 1827 1828uint32_t AudioFlinger::PlaybackThread::latency() const 1829{ 1830 Mutex::Autolock _l(mLock); 1831 return latency_l(); 1832} 1833uint32_t AudioFlinger::PlaybackThread::latency_l() const 1834{ 1835 if (initCheck() == NO_ERROR) { 1836 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1837 } else { 1838 return 0; 1839 } 1840} 1841 1842void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1843{ 1844 Mutex::Autolock _l(mLock); 1845 mMasterVolume = value; 1846} 1847 1848void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1849{ 1850 Mutex::Autolock _l(mLock); 1851 setMasterMute_l(muted); 1852} 1853 1854void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1855{ 1856 Mutex::Autolock _l(mLock); 1857 mStreamTypes[stream].volume = value; 1858} 1859 1860void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1861{ 1862 Mutex::Autolock _l(mLock); 1863 mStreamTypes[stream].mute = muted; 1864} 1865 1866float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1867{ 1868 Mutex::Autolock _l(mLock); 1869 return mStreamTypes[stream].volume; 1870} 1871 1872// addTrack_l() must be called with ThreadBase::mLock held 1873status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1874{ 1875 status_t status = ALREADY_EXISTS; 1876 1877 // set retry count for buffer fill 1878 track->mRetryCount = kMaxTrackStartupRetries; 1879 if (mActiveTracks.indexOf(track) < 0) { 1880 // the track is newly added, make sure it fills up all its 1881 // buffers before playing. This is to ensure the client will 1882 // effectively get the latency it requested. 1883 track->mFillingUpStatus = Track::FS_FILLING; 1884 track->mResetDone = false; 1885 track->mPresentationCompleteFrames = 0; 1886 mActiveTracks.add(track); 1887 if (track->mainBuffer() != mMixBuffer) { 1888 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1889 if (chain != 0) { 1890 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1891 chain->incActiveTrackCnt(); 1892 } 1893 } 1894 1895 status = NO_ERROR; 1896 } 1897 1898 ALOGV("mWaitWorkCV.broadcast"); 1899 mWaitWorkCV.broadcast(); 1900 1901 return status; 1902} 1903 1904// destroyTrack_l() must be called with ThreadBase::mLock held 1905void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1906{ 1907 track->mState = TrackBase::TERMINATED; 1908 // active tracks are removed by threadLoop() 1909 if (mActiveTracks.indexOf(track) < 0) { 1910 removeTrack_l(track); 1911 } 1912} 1913 1914void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1915{ 1916 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1917 mTracks.remove(track); 1918 deleteTrackName_l(track->name()); 1919 // redundant as track is about to be destroyed, for dumpsys only 1920 track->mName = -1; 1921 if (track->isFastTrack()) { 1922 int index = track->mFastIndex; 1923 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1924 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1925 mFastTrackAvailMask |= 1 << index; 1926 // redundant as track is about to be destroyed, for dumpsys only 1927 track->mFastIndex = -1; 1928 } 1929 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1930 if (chain != 0) { 1931 chain->decTrackCnt(); 1932 } 1933} 1934 1935String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1936{ 1937 String8 out_s8 = String8(""); 1938 char *s; 1939 1940 Mutex::Autolock _l(mLock); 1941 if (initCheck() != NO_ERROR) { 1942 return out_s8; 1943 } 1944 1945 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1946 out_s8 = String8(s); 1947 free(s); 1948 return out_s8; 1949} 1950 1951// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1952void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1953 AudioSystem::OutputDescriptor desc; 1954 void *param2 = NULL; 1955 1956 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1957 1958 switch (event) { 1959 case AudioSystem::OUTPUT_OPENED: 1960 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1961 desc.channels = mChannelMask; 1962 desc.samplingRate = mSampleRate; 1963 desc.format = mFormat; 1964 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1965 desc.latency = latency(); 1966 param2 = &desc; 1967 break; 1968 1969 case AudioSystem::STREAM_CONFIG_CHANGED: 1970 param2 = ¶m; 1971 case AudioSystem::OUTPUT_CLOSED: 1972 default: 1973 break; 1974 } 1975 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1976} 1977 1978void AudioFlinger::PlaybackThread::readOutputParameters() 1979{ 1980 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1981 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1982 mChannelCount = (uint16_t)popcount(mChannelMask); 1983 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1984 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1985 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1986 if (mFrameCount & 15) { 1987 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1988 mFrameCount); 1989 } 1990 1991 // Calculate size of normal mix buffer relative to the HAL output buffer size 1992 double multiplier = 1.0; 1993 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1994 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1995 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1996 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1997 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1998 maxNormalFrameCount = maxNormalFrameCount & ~15; 1999 if (maxNormalFrameCount < minNormalFrameCount) { 2000 maxNormalFrameCount = minNormalFrameCount; 2001 } 2002 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2003 if (multiplier <= 1.0) { 2004 multiplier = 1.0; 2005 } else if (multiplier <= 2.0) { 2006 if (2 * mFrameCount <= maxNormalFrameCount) { 2007 multiplier = 2.0; 2008 } else { 2009 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2010 } 2011 } else { 2012 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2013 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2014 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2015 // FIXME this rounding up should not be done if no HAL SRC 2016 uint32_t truncMult = (uint32_t) multiplier; 2017 if ((truncMult & 1)) { 2018 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2019 ++truncMult; 2020 } 2021 } 2022 multiplier = (double) truncMult; 2023 } 2024 } 2025 mNormalFrameCount = multiplier * mFrameCount; 2026 // round up to nearest 16 frames to satisfy AudioMixer 2027 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2028 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2029 2030 delete[] mMixBuffer; 2031 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2032 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2033 2034 // force reconfiguration of effect chains and engines to take new buffer size and audio 2035 // parameters into account 2036 // Note that mLock is not held when readOutputParameters() is called from the constructor 2037 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2038 // matter. 2039 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2040 Vector< sp<EffectChain> > effectChains = mEffectChains; 2041 for (size_t i = 0; i < effectChains.size(); i ++) { 2042 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2043 } 2044} 2045 2046 2047status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2048{ 2049 if (halFrames == NULL || dspFrames == NULL) { 2050 return BAD_VALUE; 2051 } 2052 Mutex::Autolock _l(mLock); 2053 if (initCheck() != NO_ERROR) { 2054 return INVALID_OPERATION; 2055 } 2056 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2057 2058 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2059} 2060 2061uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2062{ 2063 Mutex::Autolock _l(mLock); 2064 uint32_t result = 0; 2065 if (getEffectChain_l(sessionId) != 0) { 2066 result = EFFECT_SESSION; 2067 } 2068 2069 for (size_t i = 0; i < mTracks.size(); ++i) { 2070 sp<Track> track = mTracks[i]; 2071 if (sessionId == track->sessionId() && 2072 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2073 result |= TRACK_SESSION; 2074 break; 2075 } 2076 } 2077 2078 return result; 2079} 2080 2081uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2082{ 2083 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2084 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2085 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2086 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2087 } 2088 for (size_t i = 0; i < mTracks.size(); i++) { 2089 sp<Track> track = mTracks[i]; 2090 if (sessionId == track->sessionId() && 2091 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2092 return AudioSystem::getStrategyForStream(track->streamType()); 2093 } 2094 } 2095 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2096} 2097 2098 2099AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2100{ 2101 Mutex::Autolock _l(mLock); 2102 return mOutput; 2103} 2104 2105AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2106{ 2107 Mutex::Autolock _l(mLock); 2108 AudioStreamOut *output = mOutput; 2109 mOutput = NULL; 2110 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2111 // must push a NULL and wait for ack 2112 mOutputSink.clear(); 2113 mPipeSink.clear(); 2114 mNormalSink.clear(); 2115 return output; 2116} 2117 2118// this method must always be called either with ThreadBase mLock held or inside the thread loop 2119audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2120{ 2121 if (mOutput == NULL) { 2122 return NULL; 2123 } 2124 return &mOutput->stream->common; 2125} 2126 2127uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2128{ 2129 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2130} 2131 2132status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2133{ 2134 if (!isValidSyncEvent(event)) { 2135 return BAD_VALUE; 2136 } 2137 2138 Mutex::Autolock _l(mLock); 2139 2140 for (size_t i = 0; i < mTracks.size(); ++i) { 2141 sp<Track> track = mTracks[i]; 2142 if (event->triggerSession() == track->sessionId()) { 2143 track->setSyncEvent(event); 2144 return NO_ERROR; 2145 } 2146 } 2147 2148 return NAME_NOT_FOUND; 2149} 2150 2151bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2152{ 2153 switch (event->type()) { 2154 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2155 return true; 2156 default: 2157 break; 2158 } 2159 return false; 2160} 2161 2162void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2163{ 2164 size_t count = tracksToRemove.size(); 2165 if (CC_UNLIKELY(count)) { 2166 for (size_t i = 0 ; i < count ; i++) { 2167 const sp<Track>& track = tracksToRemove.itemAt(i); 2168 if ((track->sharedBuffer() != 0) && 2169 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2170 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2171 } 2172 } 2173 } 2174 2175} 2176 2177// ---------------------------------------------------------------------------- 2178 2179AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2180 audio_io_handle_t id, uint32_t device, type_t type) 2181 : PlaybackThread(audioFlinger, output, id, device, type), 2182 // mAudioMixer below 2183 // mFastMixer below 2184 mFastMixerFutex(0) 2185 // mOutputSink below 2186 // mPipeSink below 2187 // mNormalSink below 2188{ 2189 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2190 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2191 "mFrameCount=%d, mNormalFrameCount=%d", 2192 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2193 mNormalFrameCount); 2194 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2195 2196 // FIXME - Current mixer implementation only supports stereo output 2197 if (mChannelCount != FCC_2) { 2198 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2199 } 2200 2201 // create an NBAIO sink for the HAL output stream, and negotiate 2202 mOutputSink = new AudioStreamOutSink(output->stream); 2203 size_t numCounterOffers = 0; 2204 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2205 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2206 ALOG_ASSERT(index == 0); 2207 2208 // initialize fast mixer depending on configuration 2209 bool initFastMixer; 2210 switch (kUseFastMixer) { 2211 case FastMixer_Never: 2212 initFastMixer = false; 2213 break; 2214 case FastMixer_Always: 2215 initFastMixer = true; 2216 break; 2217 case FastMixer_Static: 2218 case FastMixer_Dynamic: 2219 initFastMixer = mFrameCount < mNormalFrameCount; 2220 break; 2221 } 2222 if (initFastMixer) { 2223 2224 // create a MonoPipe to connect our submix to FastMixer 2225 NBAIO_Format format = mOutputSink->format(); 2226 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2227 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2228 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2229 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2230 const NBAIO_Format offers[1] = {format}; 2231 size_t numCounterOffers = 0; 2232 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2233 ALOG_ASSERT(index == 0); 2234 monoPipe->setAvgFrames((mScreenState & 1) ? 2235 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2236 mPipeSink = monoPipe; 2237 2238#ifdef TEE_SINK_FRAMES 2239 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2240 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2241 numCounterOffers = 0; 2242 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2243 ALOG_ASSERT(index == 0); 2244 mTeeSink = teeSink; 2245 PipeReader *teeSource = new PipeReader(*teeSink); 2246 numCounterOffers = 0; 2247 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2248 ALOG_ASSERT(index == 0); 2249 mTeeSource = teeSource; 2250#endif 2251 2252 // create fast mixer and configure it initially with just one fast track for our submix 2253 mFastMixer = new FastMixer(); 2254 FastMixerStateQueue *sq = mFastMixer->sq(); 2255#ifdef STATE_QUEUE_DUMP 2256 sq->setObserverDump(&mStateQueueObserverDump); 2257 sq->setMutatorDump(&mStateQueueMutatorDump); 2258#endif 2259 FastMixerState *state = sq->begin(); 2260 FastTrack *fastTrack = &state->mFastTracks[0]; 2261 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2262 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2263 fastTrack->mVolumeProvider = NULL; 2264 fastTrack->mGeneration++; 2265 state->mFastTracksGen++; 2266 state->mTrackMask = 1; 2267 // fast mixer will use the HAL output sink 2268 state->mOutputSink = mOutputSink.get(); 2269 state->mOutputSinkGen++; 2270 state->mFrameCount = mFrameCount; 2271 state->mCommand = FastMixerState::COLD_IDLE; 2272 // already done in constructor initialization list 2273 //mFastMixerFutex = 0; 2274 state->mColdFutexAddr = &mFastMixerFutex; 2275 state->mColdGen++; 2276 state->mDumpState = &mFastMixerDumpState; 2277 state->mTeeSink = mTeeSink.get(); 2278 sq->end(); 2279 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2280 2281 // start the fast mixer 2282 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2283 pid_t tid = mFastMixer->getTid(); 2284 int err = requestPriority(getpid_cached, tid, 2); 2285 if (err != 0) { 2286 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2287 2, getpid_cached, tid, err); 2288 } 2289 2290#ifdef AUDIO_WATCHDOG 2291 // create and start the watchdog 2292 mAudioWatchdog = new AudioWatchdog(); 2293 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2294 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2295 tid = mAudioWatchdog->getTid(); 2296 err = requestPriority(getpid_cached, tid, 1); 2297 if (err != 0) { 2298 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2299 1, getpid_cached, tid, err); 2300 } 2301#endif 2302 2303 } else { 2304 mFastMixer = NULL; 2305 } 2306 2307 switch (kUseFastMixer) { 2308 case FastMixer_Never: 2309 case FastMixer_Dynamic: 2310 mNormalSink = mOutputSink; 2311 break; 2312 case FastMixer_Always: 2313 mNormalSink = mPipeSink; 2314 break; 2315 case FastMixer_Static: 2316 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2317 break; 2318 } 2319} 2320 2321AudioFlinger::MixerThread::~MixerThread() 2322{ 2323 if (mFastMixer != NULL) { 2324 FastMixerStateQueue *sq = mFastMixer->sq(); 2325 FastMixerState *state = sq->begin(); 2326 if (state->mCommand == FastMixerState::COLD_IDLE) { 2327 int32_t old = android_atomic_inc(&mFastMixerFutex); 2328 if (old == -1) { 2329 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2330 } 2331 } 2332 state->mCommand = FastMixerState::EXIT; 2333 sq->end(); 2334 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2335 mFastMixer->join(); 2336 // Though the fast mixer thread has exited, it's state queue is still valid. 2337 // We'll use that extract the final state which contains one remaining fast track 2338 // corresponding to our sub-mix. 2339 state = sq->begin(); 2340 ALOG_ASSERT(state->mTrackMask == 1); 2341 FastTrack *fastTrack = &state->mFastTracks[0]; 2342 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2343 delete fastTrack->mBufferProvider; 2344 sq->end(false /*didModify*/); 2345 delete mFastMixer; 2346 if (mAudioWatchdog != 0) { 2347 mAudioWatchdog->requestExit(); 2348 mAudioWatchdog->requestExitAndWait(); 2349 mAudioWatchdog.clear(); 2350 } 2351 } 2352 delete mAudioMixer; 2353} 2354 2355class CpuStats { 2356public: 2357 CpuStats(); 2358 void sample(const String8 &title); 2359#ifdef DEBUG_CPU_USAGE 2360private: 2361 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2362 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2363 2364 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2365 2366 int mCpuNum; // thread's current CPU number 2367 int mCpukHz; // frequency of thread's current CPU in kHz 2368#endif 2369}; 2370 2371CpuStats::CpuStats() 2372#ifdef DEBUG_CPU_USAGE 2373 : mCpuNum(-1), mCpukHz(-1) 2374#endif 2375{ 2376} 2377 2378void CpuStats::sample(const String8 &title) { 2379#ifdef DEBUG_CPU_USAGE 2380 // get current thread's delta CPU time in wall clock ns 2381 double wcNs; 2382 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2383 2384 // record sample for wall clock statistics 2385 if (valid) { 2386 mWcStats.sample(wcNs); 2387 } 2388 2389 // get the current CPU number 2390 int cpuNum = sched_getcpu(); 2391 2392 // get the current CPU frequency in kHz 2393 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2394 2395 // check if either CPU number or frequency changed 2396 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2397 mCpuNum = cpuNum; 2398 mCpukHz = cpukHz; 2399 // ignore sample for purposes of cycles 2400 valid = false; 2401 } 2402 2403 // if no change in CPU number or frequency, then record sample for cycle statistics 2404 if (valid && mCpukHz > 0) { 2405 double cycles = wcNs * cpukHz * 0.000001; 2406 mHzStats.sample(cycles); 2407 } 2408 2409 unsigned n = mWcStats.n(); 2410 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2411 if ((n & 127) == 1) { 2412 long long elapsed = mCpuUsage.elapsed(); 2413 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2414 double perLoop = elapsed / (double) n; 2415 double perLoop100 = perLoop * 0.01; 2416 double perLoop1k = perLoop * 0.001; 2417 double mean = mWcStats.mean(); 2418 double stddev = mWcStats.stddev(); 2419 double minimum = mWcStats.minimum(); 2420 double maximum = mWcStats.maximum(); 2421 double meanCycles = mHzStats.mean(); 2422 double stddevCycles = mHzStats.stddev(); 2423 double minCycles = mHzStats.minimum(); 2424 double maxCycles = mHzStats.maximum(); 2425 mCpuUsage.resetElapsed(); 2426 mWcStats.reset(); 2427 mHzStats.reset(); 2428 ALOGD("CPU usage for %s over past %.1f secs\n" 2429 " (%u mixer loops at %.1f mean ms per loop):\n" 2430 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2431 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2432 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2433 title.string(), 2434 elapsed * .000000001, n, perLoop * .000001, 2435 mean * .001, 2436 stddev * .001, 2437 minimum * .001, 2438 maximum * .001, 2439 mean / perLoop100, 2440 stddev / perLoop100, 2441 minimum / perLoop100, 2442 maximum / perLoop100, 2443 meanCycles / perLoop1k, 2444 stddevCycles / perLoop1k, 2445 minCycles / perLoop1k, 2446 maxCycles / perLoop1k); 2447 2448 } 2449 } 2450#endif 2451}; 2452 2453void AudioFlinger::PlaybackThread::checkSilentMode_l() 2454{ 2455 if (!mMasterMute) { 2456 char value[PROPERTY_VALUE_MAX]; 2457 if (property_get("ro.audio.silent", value, "0") > 0) { 2458 char *endptr; 2459 unsigned long ul = strtoul(value, &endptr, 0); 2460 if (*endptr == '\0' && ul != 0) { 2461 ALOGD("Silence is golden"); 2462 // The setprop command will not allow a property to be changed after 2463 // the first time it is set, so we don't have to worry about un-muting. 2464 setMasterMute_l(true); 2465 } 2466 } 2467 } 2468} 2469 2470bool AudioFlinger::PlaybackThread::threadLoop() 2471{ 2472 Vector< sp<Track> > tracksToRemove; 2473 2474 standbyTime = systemTime(); 2475 2476 // MIXER 2477 nsecs_t lastWarning = 0; 2478 2479 // DUPLICATING 2480 // FIXME could this be made local to while loop? 2481 writeFrames = 0; 2482 2483 cacheParameters_l(); 2484 sleepTime = idleSleepTime; 2485 2486 if (mType == MIXER) { 2487 sleepTimeShift = 0; 2488 } 2489 2490 CpuStats cpuStats; 2491 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2492 2493 acquireWakeLock(); 2494 2495 while (!exitPending()) 2496 { 2497 cpuStats.sample(myName); 2498 2499 Vector< sp<EffectChain> > effectChains; 2500 2501 processConfigEvents(); 2502 2503 { // scope for mLock 2504 2505 Mutex::Autolock _l(mLock); 2506 2507 if (checkForNewParameters_l()) { 2508 cacheParameters_l(); 2509 } 2510 2511 saveOutputTracks(); 2512 2513 // put audio hardware into standby after short delay 2514 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2515 isSuspended())) { 2516 if (!mStandby) { 2517 2518 threadLoop_standby(); 2519 2520 mStandby = true; 2521 mBytesWritten = 0; 2522 } 2523 2524 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2525 // we're about to wait, flush the binder command buffer 2526 IPCThreadState::self()->flushCommands(); 2527 2528 clearOutputTracks(); 2529 2530 if (exitPending()) break; 2531 2532 releaseWakeLock_l(); 2533 // wait until we have something to do... 2534 ALOGV("%s going to sleep", myName.string()); 2535 mWaitWorkCV.wait(mLock); 2536 ALOGV("%s waking up", myName.string()); 2537 acquireWakeLock_l(); 2538 2539 mMixerStatus = MIXER_IDLE; 2540 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2541 2542 checkSilentMode_l(); 2543 2544 standbyTime = systemTime() + standbyDelay; 2545 sleepTime = idleSleepTime; 2546 if (mType == MIXER) { 2547 sleepTimeShift = 0; 2548 } 2549 2550 continue; 2551 } 2552 } 2553 2554 // mMixerStatusIgnoringFastTracks is also updated internally 2555 mMixerStatus = prepareTracks_l(&tracksToRemove); 2556 2557 // prevent any changes in effect chain list and in each effect chain 2558 // during mixing and effect process as the audio buffers could be deleted 2559 // or modified if an effect is created or deleted 2560 lockEffectChains_l(effectChains); 2561 } 2562 2563 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2564 threadLoop_mix(); 2565 } else { 2566 threadLoop_sleepTime(); 2567 } 2568 2569 if (isSuspended()) { 2570 sleepTime = suspendSleepTimeUs(); 2571 } 2572 2573 // only process effects if we're going to write 2574 if (sleepTime == 0) { 2575 for (size_t i = 0; i < effectChains.size(); i ++) { 2576 effectChains[i]->process_l(); 2577 } 2578 } 2579 2580 // enable changes in effect chain 2581 unlockEffectChains(effectChains); 2582 2583 // sleepTime == 0 means we must write to audio hardware 2584 if (sleepTime == 0) { 2585 2586 threadLoop_write(); 2587 2588if (mType == MIXER) { 2589 // write blocked detection 2590 nsecs_t now = systemTime(); 2591 nsecs_t delta = now - mLastWriteTime; 2592 if (!mStandby && delta > maxPeriod) { 2593 mNumDelayedWrites++; 2594 if ((now - lastWarning) > kWarningThrottleNs) { 2595#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2596 ScopedTrace st(ATRACE_TAG, "underrun"); 2597#endif 2598 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2599 ns2ms(delta), mNumDelayedWrites, this); 2600 lastWarning = now; 2601 } 2602 } 2603} 2604 2605 mStandby = false; 2606 } else { 2607 usleep(sleepTime); 2608 } 2609 2610 // Finally let go of removed track(s), without the lock held 2611 // since we can't guarantee the destructors won't acquire that 2612 // same lock. This will also mutate and push a new fast mixer state. 2613 threadLoop_removeTracks(tracksToRemove); 2614 tracksToRemove.clear(); 2615 2616 // FIXME I don't understand the need for this here; 2617 // it was in the original code but maybe the 2618 // assignment in saveOutputTracks() makes this unnecessary? 2619 clearOutputTracks(); 2620 2621 // Effect chains will be actually deleted here if they were removed from 2622 // mEffectChains list during mixing or effects processing 2623 effectChains.clear(); 2624 2625 // FIXME Note that the above .clear() is no longer necessary since effectChains 2626 // is now local to this block, but will keep it for now (at least until merge done). 2627 } 2628 2629 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2630 if (mType == MIXER || mType == DIRECT) { 2631 // put output stream into standby mode 2632 if (!mStandby) { 2633 mOutput->stream->common.standby(&mOutput->stream->common); 2634 } 2635 } 2636 2637 releaseWakeLock(); 2638 2639 ALOGV("Thread %p type %d exiting", this, mType); 2640 return false; 2641} 2642 2643void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2644{ 2645 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2646} 2647 2648void AudioFlinger::MixerThread::threadLoop_write() 2649{ 2650 // FIXME we should only do one push per cycle; confirm this is true 2651 // Start the fast mixer if it's not already running 2652 if (mFastMixer != NULL) { 2653 FastMixerStateQueue *sq = mFastMixer->sq(); 2654 FastMixerState *state = sq->begin(); 2655 if (state->mCommand != FastMixerState::MIX_WRITE && 2656 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2657 if (state->mCommand == FastMixerState::COLD_IDLE) { 2658 int32_t old = android_atomic_inc(&mFastMixerFutex); 2659 if (old == -1) { 2660 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2661 } 2662 if (mAudioWatchdog != 0) { 2663 mAudioWatchdog->resume(); 2664 } 2665 } 2666 state->mCommand = FastMixerState::MIX_WRITE; 2667 sq->end(); 2668 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2669 if (kUseFastMixer == FastMixer_Dynamic) { 2670 mNormalSink = mPipeSink; 2671 } 2672 } else { 2673 sq->end(false /*didModify*/); 2674 } 2675 } 2676 PlaybackThread::threadLoop_write(); 2677} 2678 2679// shared by MIXER and DIRECT, overridden by DUPLICATING 2680void AudioFlinger::PlaybackThread::threadLoop_write() 2681{ 2682 // FIXME rewrite to reduce number of system calls 2683 mLastWriteTime = systemTime(); 2684 mInWrite = true; 2685 int bytesWritten; 2686 2687 // If an NBAIO sink is present, use it to write the normal mixer's submix 2688 if (mNormalSink != 0) { 2689#define mBitShift 2 // FIXME 2690 size_t count = mixBufferSize >> mBitShift; 2691#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2692 Tracer::traceBegin(ATRACE_TAG, "write"); 2693#endif 2694 // update the setpoint when gScreenState changes 2695 uint32_t screenState = gScreenState; 2696 if (screenState != mScreenState) { 2697 mScreenState = screenState; 2698 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2699 if (pipe != NULL) { 2700 pipe->setAvgFrames((mScreenState & 1) ? 2701 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2702 } 2703 } 2704 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2705#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2706 Tracer::traceEnd(ATRACE_TAG); 2707#endif 2708 if (framesWritten > 0) { 2709 bytesWritten = framesWritten << mBitShift; 2710 } else { 2711 bytesWritten = framesWritten; 2712 } 2713 // otherwise use the HAL / AudioStreamOut directly 2714 } else { 2715 // Direct output thread. 2716 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2717 } 2718 2719 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2720 mNumWrites++; 2721 mInWrite = false; 2722} 2723 2724void AudioFlinger::MixerThread::threadLoop_standby() 2725{ 2726 // Idle the fast mixer if it's currently running 2727 if (mFastMixer != NULL) { 2728 FastMixerStateQueue *sq = mFastMixer->sq(); 2729 FastMixerState *state = sq->begin(); 2730 if (!(state->mCommand & FastMixerState::IDLE)) { 2731 state->mCommand = FastMixerState::COLD_IDLE; 2732 state->mColdFutexAddr = &mFastMixerFutex; 2733 state->mColdGen++; 2734 mFastMixerFutex = 0; 2735 sq->end(); 2736 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2737 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2738 if (kUseFastMixer == FastMixer_Dynamic) { 2739 mNormalSink = mOutputSink; 2740 } 2741 if (mAudioWatchdog != 0) { 2742 mAudioWatchdog->pause(); 2743 } 2744 } else { 2745 sq->end(false /*didModify*/); 2746 } 2747 } 2748 PlaybackThread::threadLoop_standby(); 2749} 2750 2751// shared by MIXER and DIRECT, overridden by DUPLICATING 2752void AudioFlinger::PlaybackThread::threadLoop_standby() 2753{ 2754 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2755 mOutput->stream->common.standby(&mOutput->stream->common); 2756} 2757 2758void AudioFlinger::MixerThread::threadLoop_mix() 2759{ 2760 // obtain the presentation timestamp of the next output buffer 2761 int64_t pts; 2762 status_t status = INVALID_OPERATION; 2763 2764 if (NULL != mOutput->stream->get_next_write_timestamp) { 2765 status = mOutput->stream->get_next_write_timestamp( 2766 mOutput->stream, &pts); 2767 } 2768 2769 if (status != NO_ERROR) { 2770 pts = AudioBufferProvider::kInvalidPTS; 2771 } 2772 2773 // mix buffers... 2774 mAudioMixer->process(pts); 2775 // increase sleep time progressively when application underrun condition clears. 2776 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2777 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2778 // such that we would underrun the audio HAL. 2779 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2780 sleepTimeShift--; 2781 } 2782 sleepTime = 0; 2783 standbyTime = systemTime() + standbyDelay; 2784 //TODO: delay standby when effects have a tail 2785} 2786 2787void AudioFlinger::MixerThread::threadLoop_sleepTime() 2788{ 2789 // If no tracks are ready, sleep once for the duration of an output 2790 // buffer size, then write 0s to the output 2791 if (sleepTime == 0) { 2792 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2793 sleepTime = activeSleepTime >> sleepTimeShift; 2794 if (sleepTime < kMinThreadSleepTimeUs) { 2795 sleepTime = kMinThreadSleepTimeUs; 2796 } 2797 // reduce sleep time in case of consecutive application underruns to avoid 2798 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2799 // duration we would end up writing less data than needed by the audio HAL if 2800 // the condition persists. 2801 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2802 sleepTimeShift++; 2803 } 2804 } else { 2805 sleepTime = idleSleepTime; 2806 } 2807 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2808 memset (mMixBuffer, 0, mixBufferSize); 2809 sleepTime = 0; 2810 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2811 } 2812 // TODO add standby time extension fct of effect tail 2813} 2814 2815// prepareTracks_l() must be called with ThreadBase::mLock held 2816AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2817 Vector< sp<Track> > *tracksToRemove) 2818{ 2819 2820 mixer_state mixerStatus = MIXER_IDLE; 2821 // find out which tracks need to be processed 2822 size_t count = mActiveTracks.size(); 2823 size_t mixedTracks = 0; 2824 size_t tracksWithEffect = 0; 2825 // counts only _active_ fast tracks 2826 size_t fastTracks = 0; 2827 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2828 2829 float masterVolume = mMasterVolume; 2830 bool masterMute = mMasterMute; 2831 2832 if (masterMute) { 2833 masterVolume = 0; 2834 } 2835 // Delegate master volume control to effect in output mix effect chain if needed 2836 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2837 if (chain != 0) { 2838 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2839 chain->setVolume_l(&v, &v); 2840 masterVolume = (float)((v + (1 << 23)) >> 24); 2841 chain.clear(); 2842 } 2843 2844 // prepare a new state to push 2845 FastMixerStateQueue *sq = NULL; 2846 FastMixerState *state = NULL; 2847 bool didModify = false; 2848 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2849 if (mFastMixer != NULL) { 2850 sq = mFastMixer->sq(); 2851 state = sq->begin(); 2852 } 2853 2854 for (size_t i=0 ; i<count ; i++) { 2855 sp<Track> t = mActiveTracks[i].promote(); 2856 if (t == 0) continue; 2857 2858 // this const just means the local variable doesn't change 2859 Track* const track = t.get(); 2860 2861 // process fast tracks 2862 if (track->isFastTrack()) { 2863 2864 // It's theoretically possible (though unlikely) for a fast track to be created 2865 // and then removed within the same normal mix cycle. This is not a problem, as 2866 // the track never becomes active so it's fast mixer slot is never touched. 2867 // The converse, of removing an (active) track and then creating a new track 2868 // at the identical fast mixer slot within the same normal mix cycle, 2869 // is impossible because the slot isn't marked available until the end of each cycle. 2870 int j = track->mFastIndex; 2871 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2872 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2873 FastTrack *fastTrack = &state->mFastTracks[j]; 2874 2875 // Determine whether the track is currently in underrun condition, 2876 // and whether it had a recent underrun. 2877 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2878 FastTrackUnderruns underruns = ftDump->mUnderruns; 2879 uint32_t recentFull = (underruns.mBitFields.mFull - 2880 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2881 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2882 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2883 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2884 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2885 uint32_t recentUnderruns = recentPartial + recentEmpty; 2886 track->mObservedUnderruns = underruns; 2887 // don't count underruns that occur while stopping or pausing 2888 // or stopped which can occur when flush() is called while active 2889 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2890 track->mUnderrunCount += recentUnderruns; 2891 } 2892 2893 // This is similar to the state machine for normal tracks, 2894 // with a few modifications for fast tracks. 2895 bool isActive = true; 2896 switch (track->mState) { 2897 case TrackBase::STOPPING_1: 2898 // track stays active in STOPPING_1 state until first underrun 2899 if (recentUnderruns > 0) { 2900 track->mState = TrackBase::STOPPING_2; 2901 } 2902 break; 2903 case TrackBase::PAUSING: 2904 // ramp down is not yet implemented 2905 track->setPaused(); 2906 break; 2907 case TrackBase::RESUMING: 2908 // ramp up is not yet implemented 2909 track->mState = TrackBase::ACTIVE; 2910 break; 2911 case TrackBase::ACTIVE: 2912 if (recentFull > 0 || recentPartial > 0) { 2913 // track has provided at least some frames recently: reset retry count 2914 track->mRetryCount = kMaxTrackRetries; 2915 } 2916 if (recentUnderruns == 0) { 2917 // no recent underruns: stay active 2918 break; 2919 } 2920 // there has recently been an underrun of some kind 2921 if (track->sharedBuffer() == 0) { 2922 // were any of the recent underruns "empty" (no frames available)? 2923 if (recentEmpty == 0) { 2924 // no, then ignore the partial underruns as they are allowed indefinitely 2925 break; 2926 } 2927 // there has recently been an "empty" underrun: decrement the retry counter 2928 if (--(track->mRetryCount) > 0) { 2929 break; 2930 } 2931 // indicate to client process that the track was disabled because of underrun; 2932 // it will then automatically call start() when data is available 2933 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2934 // remove from active list, but state remains ACTIVE [confusing but true] 2935 isActive = false; 2936 break; 2937 } 2938 // fall through 2939 case TrackBase::STOPPING_2: 2940 case TrackBase::PAUSED: 2941 case TrackBase::TERMINATED: 2942 case TrackBase::STOPPED: 2943 case TrackBase::FLUSHED: // flush() while active 2944 // Check for presentation complete if track is inactive 2945 // We have consumed all the buffers of this track. 2946 // This would be incomplete if we auto-paused on underrun 2947 { 2948 size_t audioHALFrames = 2949 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2950 size_t framesWritten = 2951 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2952 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2953 // track stays in active list until presentation is complete 2954 break; 2955 } 2956 } 2957 if (track->isStopping_2()) { 2958 track->mState = TrackBase::STOPPED; 2959 } 2960 if (track->isStopped()) { 2961 // Can't reset directly, as fast mixer is still polling this track 2962 // track->reset(); 2963 // So instead mark this track as needing to be reset after push with ack 2964 resetMask |= 1 << i; 2965 } 2966 isActive = false; 2967 break; 2968 case TrackBase::IDLE: 2969 default: 2970 LOG_FATAL("unexpected track state %d", track->mState); 2971 } 2972 2973 if (isActive) { 2974 // was it previously inactive? 2975 if (!(state->mTrackMask & (1 << j))) { 2976 ExtendedAudioBufferProvider *eabp = track; 2977 VolumeProvider *vp = track; 2978 fastTrack->mBufferProvider = eabp; 2979 fastTrack->mVolumeProvider = vp; 2980 fastTrack->mSampleRate = track->mSampleRate; 2981 fastTrack->mChannelMask = track->mChannelMask; 2982 fastTrack->mGeneration++; 2983 state->mTrackMask |= 1 << j; 2984 didModify = true; 2985 // no acknowledgement required for newly active tracks 2986 } 2987 // cache the combined master volume and stream type volume for fast mixer; this 2988 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2989 track->mCachedVolume = track->isMuted() ? 2990 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2991 ++fastTracks; 2992 } else { 2993 // was it previously active? 2994 if (state->mTrackMask & (1 << j)) { 2995 fastTrack->mBufferProvider = NULL; 2996 fastTrack->mGeneration++; 2997 state->mTrackMask &= ~(1 << j); 2998 didModify = true; 2999 // If any fast tracks were removed, we must wait for acknowledgement 3000 // because we're about to decrement the last sp<> on those tracks. 3001 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3002 } else { 3003 LOG_FATAL("fast track %d should have been active", j); 3004 } 3005 tracksToRemove->add(track); 3006 // Avoids a misleading display in dumpsys 3007 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3008 } 3009 continue; 3010 } 3011 3012 { // local variable scope to avoid goto warning 3013 3014 audio_track_cblk_t* cblk = track->cblk(); 3015 3016 // The first time a track is added we wait 3017 // for all its buffers to be filled before processing it 3018 int name = track->name(); 3019 // make sure that we have enough frames to mix one full buffer. 3020 // enforce this condition only once to enable draining the buffer in case the client 3021 // app does not call stop() and relies on underrun to stop: 3022 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3023 // during last round 3024 uint32_t minFrames = 1; 3025 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3026 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3027 if (t->sampleRate() == (int)mSampleRate) { 3028 minFrames = mNormalFrameCount; 3029 } else { 3030 // +1 for rounding and +1 for additional sample needed for interpolation 3031 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3032 // add frames already consumed but not yet released by the resampler 3033 // because cblk->framesReady() will include these frames 3034 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3035 // the minimum track buffer size is normally twice the number of frames necessary 3036 // to fill one buffer and the resampler should not leave more than one buffer worth 3037 // of unreleased frames after each pass, but just in case... 3038 ALOG_ASSERT(minFrames <= cblk->frameCount); 3039 } 3040 } 3041 if ((track->framesReady() >= minFrames) && track->isReady() && 3042 !track->isPaused() && !track->isTerminated()) 3043 { 3044 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3045 3046 mixedTracks++; 3047 3048 // track->mainBuffer() != mMixBuffer means there is an effect chain 3049 // connected to the track 3050 chain.clear(); 3051 if (track->mainBuffer() != mMixBuffer) { 3052 chain = getEffectChain_l(track->sessionId()); 3053 // Delegate volume control to effect in track effect chain if needed 3054 if (chain != 0) { 3055 tracksWithEffect++; 3056 } else { 3057 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3058 name, track->sessionId()); 3059 } 3060 } 3061 3062 3063 int param = AudioMixer::VOLUME; 3064 if (track->mFillingUpStatus == Track::FS_FILLED) { 3065 // no ramp for the first volume setting 3066 track->mFillingUpStatus = Track::FS_ACTIVE; 3067 if (track->mState == TrackBase::RESUMING) { 3068 track->mState = TrackBase::ACTIVE; 3069 param = AudioMixer::RAMP_VOLUME; 3070 } 3071 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3072 } else if (cblk->server != 0) { 3073 // If the track is stopped before the first frame was mixed, 3074 // do not apply ramp 3075 param = AudioMixer::RAMP_VOLUME; 3076 } 3077 3078 // compute volume for this track 3079 uint32_t vl, vr, va; 3080 if (track->isMuted() || track->isPausing() || 3081 mStreamTypes[track->streamType()].mute) { 3082 vl = vr = va = 0; 3083 if (track->isPausing()) { 3084 track->setPaused(); 3085 } 3086 } else { 3087 3088 // read original volumes with volume control 3089 float typeVolume = mStreamTypes[track->streamType()].volume; 3090 float v = masterVolume * typeVolume; 3091 uint32_t vlr = cblk->getVolumeLR(); 3092 vl = vlr & 0xFFFF; 3093 vr = vlr >> 16; 3094 // track volumes come from shared memory, so can't be trusted and must be clamped 3095 if (vl > MAX_GAIN_INT) { 3096 ALOGV("Track left volume out of range: %04X", vl); 3097 vl = MAX_GAIN_INT; 3098 } 3099 if (vr > MAX_GAIN_INT) { 3100 ALOGV("Track right volume out of range: %04X", vr); 3101 vr = MAX_GAIN_INT; 3102 } 3103 // now apply the master volume and stream type volume 3104 vl = (uint32_t)(v * vl) << 12; 3105 vr = (uint32_t)(v * vr) << 12; 3106 // assuming master volume and stream type volume each go up to 1.0, 3107 // vl and vr are now in 8.24 format 3108 3109 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3110 // send level comes from shared memory and so may be corrupt 3111 if (sendLevel > MAX_GAIN_INT) { 3112 ALOGV("Track send level out of range: %04X", sendLevel); 3113 sendLevel = MAX_GAIN_INT; 3114 } 3115 va = (uint32_t)(v * sendLevel); 3116 } 3117 // Delegate volume control to effect in track effect chain if needed 3118 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3119 // Do not ramp volume if volume is controlled by effect 3120 param = AudioMixer::VOLUME; 3121 track->mHasVolumeController = true; 3122 } else { 3123 // force no volume ramp when volume controller was just disabled or removed 3124 // from effect chain to avoid volume spike 3125 if (track->mHasVolumeController) { 3126 param = AudioMixer::VOLUME; 3127 } 3128 track->mHasVolumeController = false; 3129 } 3130 3131 // Convert volumes from 8.24 to 4.12 format 3132 // This additional clamping is needed in case chain->setVolume_l() overshot 3133 vl = (vl + (1 << 11)) >> 12; 3134 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3135 vr = (vr + (1 << 11)) >> 12; 3136 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3137 3138 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3139 3140 // XXX: these things DON'T need to be done each time 3141 mAudioMixer->setBufferProvider(name, track); 3142 mAudioMixer->enable(name); 3143 3144 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3145 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3146 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3147 mAudioMixer->setParameter( 3148 name, 3149 AudioMixer::TRACK, 3150 AudioMixer::FORMAT, (void *)track->format()); 3151 mAudioMixer->setParameter( 3152 name, 3153 AudioMixer::TRACK, 3154 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3155 mAudioMixer->setParameter( 3156 name, 3157 AudioMixer::RESAMPLE, 3158 AudioMixer::SAMPLE_RATE, 3159 (void *)(cblk->sampleRate)); 3160 mAudioMixer->setParameter( 3161 name, 3162 AudioMixer::TRACK, 3163 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3164 mAudioMixer->setParameter( 3165 name, 3166 AudioMixer::TRACK, 3167 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3168 3169 // reset retry count 3170 track->mRetryCount = kMaxTrackRetries; 3171 3172 // If one track is ready, set the mixer ready if: 3173 // - the mixer was not ready during previous round OR 3174 // - no other track is not ready 3175 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3176 mixerStatus != MIXER_TRACKS_ENABLED) { 3177 mixerStatus = MIXER_TRACKS_READY; 3178 } 3179 } else { 3180 // clear effect chain input buffer if an active track underruns to avoid sending 3181 // previous audio buffer again to effects 3182 chain = getEffectChain_l(track->sessionId()); 3183 if (chain != 0) { 3184 chain->clearInputBuffer(); 3185 } 3186 3187 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3188 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3189 track->isStopped() || track->isPaused()) { 3190 // We have consumed all the buffers of this track. 3191 // Remove it from the list of active tracks. 3192 // TODO: use actual buffer filling status instead of latency when available from 3193 // audio HAL 3194 size_t audioHALFrames = 3195 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3196 size_t framesWritten = 3197 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3198 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3199 if (track->isStopped()) { 3200 track->reset(); 3201 } 3202 tracksToRemove->add(track); 3203 } 3204 } else { 3205 track->mUnderrunCount++; 3206 // No buffers for this track. Give it a few chances to 3207 // fill a buffer, then remove it from active list. 3208 if (--(track->mRetryCount) <= 0) { 3209 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3210 tracksToRemove->add(track); 3211 // indicate to client process that the track was disabled because of underrun; 3212 // it will then automatically call start() when data is available 3213 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3214 // If one track is not ready, mark the mixer also not ready if: 3215 // - the mixer was ready during previous round OR 3216 // - no other track is ready 3217 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3218 mixerStatus != MIXER_TRACKS_READY) { 3219 mixerStatus = MIXER_TRACKS_ENABLED; 3220 } 3221 } 3222 mAudioMixer->disable(name); 3223 } 3224 3225 } // local variable scope to avoid goto warning 3226track_is_ready: ; 3227 3228 } 3229 3230 // Push the new FastMixer state if necessary 3231 bool pauseAudioWatchdog = false; 3232 if (didModify) { 3233 state->mFastTracksGen++; 3234 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3235 if (kUseFastMixer == FastMixer_Dynamic && 3236 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3237 state->mCommand = FastMixerState::COLD_IDLE; 3238 state->mColdFutexAddr = &mFastMixerFutex; 3239 state->mColdGen++; 3240 mFastMixerFutex = 0; 3241 if (kUseFastMixer == FastMixer_Dynamic) { 3242 mNormalSink = mOutputSink; 3243 } 3244 // If we go into cold idle, need to wait for acknowledgement 3245 // so that fast mixer stops doing I/O. 3246 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3247 pauseAudioWatchdog = true; 3248 } 3249 sq->end(); 3250 } 3251 if (sq != NULL) { 3252 sq->end(didModify); 3253 sq->push(block); 3254 } 3255 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3256 mAudioWatchdog->pause(); 3257 } 3258 3259 // Now perform the deferred reset on fast tracks that have stopped 3260 while (resetMask != 0) { 3261 size_t i = __builtin_ctz(resetMask); 3262 ALOG_ASSERT(i < count); 3263 resetMask &= ~(1 << i); 3264 sp<Track> t = mActiveTracks[i].promote(); 3265 if (t == 0) continue; 3266 Track* track = t.get(); 3267 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3268 track->reset(); 3269 } 3270 3271 // remove all the tracks that need to be... 3272 count = tracksToRemove->size(); 3273 if (CC_UNLIKELY(count)) { 3274 for (size_t i=0 ; i<count ; i++) { 3275 const sp<Track>& track = tracksToRemove->itemAt(i); 3276 mActiveTracks.remove(track); 3277 if (track->mainBuffer() != mMixBuffer) { 3278 chain = getEffectChain_l(track->sessionId()); 3279 if (chain != 0) { 3280 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3281 chain->decActiveTrackCnt(); 3282 } 3283 } 3284 if (track->isTerminated()) { 3285 removeTrack_l(track); 3286 } 3287 } 3288 } 3289 3290 // mix buffer must be cleared if all tracks are connected to an 3291 // effect chain as in this case the mixer will not write to 3292 // mix buffer and track effects will accumulate into it 3293 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3294 // FIXME as a performance optimization, should remember previous zero status 3295 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3296 } 3297 3298 // if any fast tracks, then status is ready 3299 mMixerStatusIgnoringFastTracks = mixerStatus; 3300 if (fastTracks > 0) { 3301 mixerStatus = MIXER_TRACKS_READY; 3302 } 3303 return mixerStatus; 3304} 3305 3306/* 3307The derived values that are cached: 3308 - mixBufferSize from frame count * frame size 3309 - activeSleepTime from activeSleepTimeUs() 3310 - idleSleepTime from idleSleepTimeUs() 3311 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3312 - maxPeriod from frame count and sample rate (MIXER only) 3313 3314The parameters that affect these derived values are: 3315 - frame count 3316 - frame size 3317 - sample rate 3318 - device type: A2DP or not 3319 - device latency 3320 - format: PCM or not 3321 - active sleep time 3322 - idle sleep time 3323*/ 3324 3325void AudioFlinger::PlaybackThread::cacheParameters_l() 3326{ 3327 mixBufferSize = mNormalFrameCount * mFrameSize; 3328 activeSleepTime = activeSleepTimeUs(); 3329 idleSleepTime = idleSleepTimeUs(); 3330} 3331 3332void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3333{ 3334 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3335 this, streamType, mTracks.size()); 3336 Mutex::Autolock _l(mLock); 3337 3338 size_t size = mTracks.size(); 3339 for (size_t i = 0; i < size; i++) { 3340 sp<Track> t = mTracks[i]; 3341 if (t->streamType() == streamType) { 3342 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3343 t->mCblk->cv.signal(); 3344 } 3345 } 3346} 3347 3348// getTrackName_l() must be called with ThreadBase::mLock held 3349int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3350{ 3351 return mAudioMixer->getTrackName(channelMask); 3352} 3353 3354// deleteTrackName_l() must be called with ThreadBase::mLock held 3355void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3356{ 3357 ALOGV("remove track (%d) and delete from mixer", name); 3358 mAudioMixer->deleteTrackName(name); 3359} 3360 3361// checkForNewParameters_l() must be called with ThreadBase::mLock held 3362bool AudioFlinger::MixerThread::checkForNewParameters_l() 3363{ 3364 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3365 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3366 bool reconfig = false; 3367 3368 while (!mNewParameters.isEmpty()) { 3369 3370 if (mFastMixer != NULL) { 3371 FastMixerStateQueue *sq = mFastMixer->sq(); 3372 FastMixerState *state = sq->begin(); 3373 if (!(state->mCommand & FastMixerState::IDLE)) { 3374 previousCommand = state->mCommand; 3375 state->mCommand = FastMixerState::HOT_IDLE; 3376 sq->end(); 3377 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3378 } else { 3379 sq->end(false /*didModify*/); 3380 } 3381 } 3382 3383 status_t status = NO_ERROR; 3384 String8 keyValuePair = mNewParameters[0]; 3385 AudioParameter param = AudioParameter(keyValuePair); 3386 int value; 3387 3388 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3389 reconfig = true; 3390 } 3391 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3392 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3393 status = BAD_VALUE; 3394 } else { 3395 reconfig = true; 3396 } 3397 } 3398 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3399 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3400 status = BAD_VALUE; 3401 } else { 3402 reconfig = true; 3403 } 3404 } 3405 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3406 // do not accept frame count changes if tracks are open as the track buffer 3407 // size depends on frame count and correct behavior would not be guaranteed 3408 // if frame count is changed after track creation 3409 if (!mTracks.isEmpty()) { 3410 status = INVALID_OPERATION; 3411 } else { 3412 reconfig = true; 3413 } 3414 } 3415 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3416#ifdef ADD_BATTERY_DATA 3417 // when changing the audio output device, call addBatteryData to notify 3418 // the change 3419 if ((int)mDevice != value) { 3420 uint32_t params = 0; 3421 // check whether speaker is on 3422 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3423 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3424 } 3425 3426 int deviceWithoutSpeaker 3427 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3428 // check if any other device (except speaker) is on 3429 if (value & deviceWithoutSpeaker ) { 3430 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3431 } 3432 3433 if (params != 0) { 3434 addBatteryData(params); 3435 } 3436 } 3437#endif 3438 3439 // forward device change to effects that have requested to be 3440 // aware of attached audio device. 3441 mDevice = (audio_devices_t) value; 3442 for (size_t i = 0; i < mEffectChains.size(); i++) { 3443 mEffectChains[i]->setDevice_l(mDevice); 3444 } 3445 } 3446 3447 if (status == NO_ERROR) { 3448 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3449 keyValuePair.string()); 3450 if (!mStandby && status == INVALID_OPERATION) { 3451 mOutput->stream->common.standby(&mOutput->stream->common); 3452 mStandby = true; 3453 mBytesWritten = 0; 3454 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3455 keyValuePair.string()); 3456 } 3457 if (status == NO_ERROR && reconfig) { 3458 delete mAudioMixer; 3459 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3460 mAudioMixer = NULL; 3461 readOutputParameters(); 3462 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3463 for (size_t i = 0; i < mTracks.size() ; i++) { 3464 int name = getTrackName_l(mTracks[i]->mChannelMask); 3465 if (name < 0) break; 3466 mTracks[i]->mName = name; 3467 // limit track sample rate to 2 x new output sample rate 3468 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3469 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3470 } 3471 } 3472 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3473 } 3474 } 3475 3476 mNewParameters.removeAt(0); 3477 3478 mParamStatus = status; 3479 mParamCond.signal(); 3480 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3481 // already timed out waiting for the status and will never signal the condition. 3482 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3483 } 3484 3485 if (!(previousCommand & FastMixerState::IDLE)) { 3486 ALOG_ASSERT(mFastMixer != NULL); 3487 FastMixerStateQueue *sq = mFastMixer->sq(); 3488 FastMixerState *state = sq->begin(); 3489 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3490 state->mCommand = previousCommand; 3491 sq->end(); 3492 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3493 } 3494 3495 return reconfig; 3496} 3497 3498status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3499{ 3500 const size_t SIZE = 256; 3501 char buffer[SIZE]; 3502 String8 result; 3503 3504 PlaybackThread::dumpInternals(fd, args); 3505 3506 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3507 result.append(buffer); 3508 write(fd, result.string(), result.size()); 3509 3510 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3511 FastMixerDumpState copy = mFastMixerDumpState; 3512 copy.dump(fd); 3513 3514#ifdef STATE_QUEUE_DUMP 3515 // Similar for state queue 3516 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3517 observerCopy.dump(fd); 3518 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3519 mutatorCopy.dump(fd); 3520#endif 3521 3522 // Write the tee output to a .wav file 3523 NBAIO_Source *teeSource = mTeeSource.get(); 3524 if (teeSource != NULL) { 3525 char teePath[64]; 3526 struct timeval tv; 3527 gettimeofday(&tv, NULL); 3528 struct tm tm; 3529 localtime_r(&tv.tv_sec, &tm); 3530 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3531 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3532 if (teeFd >= 0) { 3533 char wavHeader[44]; 3534 memcpy(wavHeader, 3535 "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", 3536 sizeof(wavHeader)); 3537 NBAIO_Format format = teeSource->format(); 3538 unsigned channelCount = Format_channelCount(format); 3539 ALOG_ASSERT(channelCount <= FCC_2); 3540 unsigned sampleRate = Format_sampleRate(format); 3541 wavHeader[22] = channelCount; // number of channels 3542 wavHeader[24] = sampleRate; // sample rate 3543 wavHeader[25] = sampleRate >> 8; 3544 wavHeader[32] = channelCount * 2; // block alignment 3545 write(teeFd, wavHeader, sizeof(wavHeader)); 3546 size_t total = 0; 3547 bool firstRead = true; 3548 for (;;) { 3549#define TEE_SINK_READ 1024 3550 short buffer[TEE_SINK_READ * FCC_2]; 3551 size_t count = TEE_SINK_READ; 3552 ssize_t actual = teeSource->read(buffer, count); 3553 bool wasFirstRead = firstRead; 3554 firstRead = false; 3555 if (actual <= 0) { 3556 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3557 continue; 3558 } 3559 break; 3560 } 3561 ALOG_ASSERT(actual <= (ssize_t)count); 3562 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3563 total += actual; 3564 } 3565 lseek(teeFd, (off_t) 4, SEEK_SET); 3566 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3567 write(teeFd, &temp, sizeof(temp)); 3568 lseek(teeFd, (off_t) 40, SEEK_SET); 3569 temp = total * channelCount * sizeof(short); 3570 write(teeFd, &temp, sizeof(temp)); 3571 close(teeFd); 3572 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3573 } else { 3574 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3575 } 3576 } 3577 3578 if (mAudioWatchdog != 0) { 3579 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3580 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3581 wdCopy.dump(fd); 3582 } 3583 3584 return NO_ERROR; 3585} 3586 3587uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3588{ 3589 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3590} 3591 3592uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3593{ 3594 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3595} 3596 3597void AudioFlinger::MixerThread::cacheParameters_l() 3598{ 3599 PlaybackThread::cacheParameters_l(); 3600 3601 // FIXME: Relaxed timing because of a certain device that can't meet latency 3602 // Should be reduced to 2x after the vendor fixes the driver issue 3603 // increase threshold again due to low power audio mode. The way this warning 3604 // threshold is calculated and its usefulness should be reconsidered anyway. 3605 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3606} 3607 3608// ---------------------------------------------------------------------------- 3609AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3610 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3611 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3612 // mLeftVolFloat, mRightVolFloat 3613{ 3614} 3615 3616AudioFlinger::DirectOutputThread::~DirectOutputThread() 3617{ 3618} 3619 3620AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3621 Vector< sp<Track> > *tracksToRemove 3622) 3623{ 3624 sp<Track> trackToRemove; 3625 3626 mixer_state mixerStatus = MIXER_IDLE; 3627 3628 // find out which tracks need to be processed 3629 if (mActiveTracks.size() != 0) { 3630 sp<Track> t = mActiveTracks[0].promote(); 3631 // The track died recently 3632 if (t == 0) return MIXER_IDLE; 3633 3634 Track* const track = t.get(); 3635 audio_track_cblk_t* cblk = track->cblk(); 3636 3637 // The first time a track is added we wait 3638 // for all its buffers to be filled before processing it 3639 uint32_t minFrames; 3640 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3641 minFrames = mNormalFrameCount; 3642 } else { 3643 minFrames = 1; 3644 } 3645 if ((track->framesReady() >= minFrames) && track->isReady() && 3646 !track->isPaused() && !track->isTerminated()) 3647 { 3648 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3649 3650 if (track->mFillingUpStatus == Track::FS_FILLED) { 3651 track->mFillingUpStatus = Track::FS_ACTIVE; 3652 mLeftVolFloat = mRightVolFloat = 0; 3653 if (track->mState == TrackBase::RESUMING) { 3654 track->mState = TrackBase::ACTIVE; 3655 } 3656 } 3657 3658 // compute volume for this track 3659 float left, right; 3660 if (track->isMuted() || mMasterMute || track->isPausing() || 3661 mStreamTypes[track->streamType()].mute) { 3662 left = right = 0; 3663 if (track->isPausing()) { 3664 track->setPaused(); 3665 } 3666 } else { 3667 float typeVolume = mStreamTypes[track->streamType()].volume; 3668 float v = mMasterVolume * typeVolume; 3669 uint32_t vlr = cblk->getVolumeLR(); 3670 float v_clamped = v * (vlr & 0xFFFF); 3671 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3672 left = v_clamped/MAX_GAIN; 3673 v_clamped = v * (vlr >> 16); 3674 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3675 right = v_clamped/MAX_GAIN; 3676 } 3677 3678 if (left != mLeftVolFloat || right != mRightVolFloat) { 3679 mLeftVolFloat = left; 3680 mRightVolFloat = right; 3681 3682 // Convert volumes from float to 8.24 3683 uint32_t vl = (uint32_t)(left * (1 << 24)); 3684 uint32_t vr = (uint32_t)(right * (1 << 24)); 3685 3686 // Delegate volume control to effect in track effect chain if needed 3687 // only one effect chain can be present on DirectOutputThread, so if 3688 // there is one, the track is connected to it 3689 if (!mEffectChains.isEmpty()) { 3690 // Do not ramp volume if volume is controlled by effect 3691 mEffectChains[0]->setVolume_l(&vl, &vr); 3692 left = (float)vl / (1 << 24); 3693 right = (float)vr / (1 << 24); 3694 } 3695 mOutput->stream->set_volume(mOutput->stream, left, right); 3696 } 3697 3698 // reset retry count 3699 track->mRetryCount = kMaxTrackRetriesDirect; 3700 mActiveTrack = t; 3701 mixerStatus = MIXER_TRACKS_READY; 3702 } else { 3703 // clear effect chain input buffer if an active track underruns to avoid sending 3704 // previous audio buffer again to effects 3705 if (!mEffectChains.isEmpty()) { 3706 mEffectChains[0]->clearInputBuffer(); 3707 } 3708 3709 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3710 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3711 track->isStopped() || track->isPaused()) { 3712 // We have consumed all the buffers of this track. 3713 // Remove it from the list of active tracks. 3714 // TODO: implement behavior for compressed audio 3715 size_t audioHALFrames = 3716 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3717 size_t framesWritten = 3718 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3719 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3720 if (track->isStopped()) { 3721 track->reset(); 3722 } 3723 trackToRemove = track; 3724 } 3725 } else { 3726 // No buffers for this track. Give it a few chances to 3727 // fill a buffer, then remove it from active list. 3728 if (--(track->mRetryCount) <= 0) { 3729 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3730 trackToRemove = track; 3731 } else { 3732 mixerStatus = MIXER_TRACKS_ENABLED; 3733 } 3734 } 3735 } 3736 } 3737 3738 // FIXME merge this with similar code for removing multiple tracks 3739 // remove all the tracks that need to be... 3740 if (CC_UNLIKELY(trackToRemove != 0)) { 3741 tracksToRemove->add(trackToRemove); 3742 mActiveTracks.remove(trackToRemove); 3743 if (!mEffectChains.isEmpty()) { 3744 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3745 trackToRemove->sessionId()); 3746 mEffectChains[0]->decActiveTrackCnt(); 3747 } 3748 if (trackToRemove->isTerminated()) { 3749 removeTrack_l(trackToRemove); 3750 } 3751 } 3752 3753 return mixerStatus; 3754} 3755 3756void AudioFlinger::DirectOutputThread::threadLoop_mix() 3757{ 3758 AudioBufferProvider::Buffer buffer; 3759 size_t frameCount = mFrameCount; 3760 int8_t *curBuf = (int8_t *)mMixBuffer; 3761 // output audio to hardware 3762 while (frameCount) { 3763 buffer.frameCount = frameCount; 3764 mActiveTrack->getNextBuffer(&buffer); 3765 if (CC_UNLIKELY(buffer.raw == NULL)) { 3766 memset(curBuf, 0, frameCount * mFrameSize); 3767 break; 3768 } 3769 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3770 frameCount -= buffer.frameCount; 3771 curBuf += buffer.frameCount * mFrameSize; 3772 mActiveTrack->releaseBuffer(&buffer); 3773 } 3774 sleepTime = 0; 3775 standbyTime = systemTime() + standbyDelay; 3776 mActiveTrack.clear(); 3777 3778} 3779 3780void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3781{ 3782 if (sleepTime == 0) { 3783 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3784 sleepTime = activeSleepTime; 3785 } else { 3786 sleepTime = idleSleepTime; 3787 } 3788 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3789 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3790 sleepTime = 0; 3791 } 3792} 3793 3794// getTrackName_l() must be called with ThreadBase::mLock held 3795int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3796{ 3797 return 0; 3798} 3799 3800// deleteTrackName_l() must be called with ThreadBase::mLock held 3801void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3802{ 3803} 3804 3805// checkForNewParameters_l() must be called with ThreadBase::mLock held 3806bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3807{ 3808 bool reconfig = false; 3809 3810 while (!mNewParameters.isEmpty()) { 3811 status_t status = NO_ERROR; 3812 String8 keyValuePair = mNewParameters[0]; 3813 AudioParameter param = AudioParameter(keyValuePair); 3814 int value; 3815 3816 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3817 // do not accept frame count changes if tracks are open as the track buffer 3818 // size depends on frame count and correct behavior would not be garantied 3819 // if frame count is changed after track creation 3820 if (!mTracks.isEmpty()) { 3821 status = INVALID_OPERATION; 3822 } else { 3823 reconfig = true; 3824 } 3825 } 3826 if (status == NO_ERROR) { 3827 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3828 keyValuePair.string()); 3829 if (!mStandby && status == INVALID_OPERATION) { 3830 mOutput->stream->common.standby(&mOutput->stream->common); 3831 mStandby = true; 3832 mBytesWritten = 0; 3833 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3834 keyValuePair.string()); 3835 } 3836 if (status == NO_ERROR && reconfig) { 3837 readOutputParameters(); 3838 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3839 } 3840 } 3841 3842 mNewParameters.removeAt(0); 3843 3844 mParamStatus = status; 3845 mParamCond.signal(); 3846 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3847 // already timed out waiting for the status and will never signal the condition. 3848 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3849 } 3850 return reconfig; 3851} 3852 3853uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3854{ 3855 uint32_t time; 3856 if (audio_is_linear_pcm(mFormat)) { 3857 time = PlaybackThread::activeSleepTimeUs(); 3858 } else { 3859 time = 10000; 3860 } 3861 return time; 3862} 3863 3864uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3865{ 3866 uint32_t time; 3867 if (audio_is_linear_pcm(mFormat)) { 3868 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3869 } else { 3870 time = 10000; 3871 } 3872 return time; 3873} 3874 3875uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3876{ 3877 uint32_t time; 3878 if (audio_is_linear_pcm(mFormat)) { 3879 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3880 } else { 3881 time = 10000; 3882 } 3883 return time; 3884} 3885 3886void AudioFlinger::DirectOutputThread::cacheParameters_l() 3887{ 3888 PlaybackThread::cacheParameters_l(); 3889 3890 // use shorter standby delay as on normal output to release 3891 // hardware resources as soon as possible 3892 standbyDelay = microseconds(activeSleepTime*2); 3893} 3894 3895// ---------------------------------------------------------------------------- 3896 3897AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3898 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3899 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3900 mWaitTimeMs(UINT_MAX) 3901{ 3902 addOutputTrack(mainThread); 3903} 3904 3905AudioFlinger::DuplicatingThread::~DuplicatingThread() 3906{ 3907 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3908 mOutputTracks[i]->destroy(); 3909 } 3910} 3911 3912void AudioFlinger::DuplicatingThread::threadLoop_mix() 3913{ 3914 // mix buffers... 3915 if (outputsReady(outputTracks)) { 3916 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3917 } else { 3918 memset(mMixBuffer, 0, mixBufferSize); 3919 } 3920 sleepTime = 0; 3921 writeFrames = mNormalFrameCount; 3922 standbyTime = systemTime() + standbyDelay; 3923} 3924 3925void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3926{ 3927 if (sleepTime == 0) { 3928 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3929 sleepTime = activeSleepTime; 3930 } else { 3931 sleepTime = idleSleepTime; 3932 } 3933 } else if (mBytesWritten != 0) { 3934 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3935 writeFrames = mNormalFrameCount; 3936 memset(mMixBuffer, 0, mixBufferSize); 3937 } else { 3938 // flush remaining overflow buffers in output tracks 3939 writeFrames = 0; 3940 } 3941 sleepTime = 0; 3942 } 3943} 3944 3945void AudioFlinger::DuplicatingThread::threadLoop_write() 3946{ 3947 for (size_t i = 0; i < outputTracks.size(); i++) { 3948 outputTracks[i]->write(mMixBuffer, writeFrames); 3949 } 3950 mBytesWritten += mixBufferSize; 3951} 3952 3953void AudioFlinger::DuplicatingThread::threadLoop_standby() 3954{ 3955 // DuplicatingThread implements standby by stopping all tracks 3956 for (size_t i = 0; i < outputTracks.size(); i++) { 3957 outputTracks[i]->stop(); 3958 } 3959} 3960 3961void AudioFlinger::DuplicatingThread::saveOutputTracks() 3962{ 3963 outputTracks = mOutputTracks; 3964} 3965 3966void AudioFlinger::DuplicatingThread::clearOutputTracks() 3967{ 3968 outputTracks.clear(); 3969} 3970 3971void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3972{ 3973 Mutex::Autolock _l(mLock); 3974 // FIXME explain this formula 3975 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3976 OutputTrack *outputTrack = new OutputTrack(thread, 3977 this, 3978 mSampleRate, 3979 mFormat, 3980 mChannelMask, 3981 frameCount); 3982 if (outputTrack->cblk() != NULL) { 3983 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3984 mOutputTracks.add(outputTrack); 3985 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3986 updateWaitTime_l(); 3987 } 3988} 3989 3990void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3991{ 3992 Mutex::Autolock _l(mLock); 3993 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3994 if (mOutputTracks[i]->thread() == thread) { 3995 mOutputTracks[i]->destroy(); 3996 mOutputTracks.removeAt(i); 3997 updateWaitTime_l(); 3998 return; 3999 } 4000 } 4001 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4002} 4003 4004// caller must hold mLock 4005void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4006{ 4007 mWaitTimeMs = UINT_MAX; 4008 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4009 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4010 if (strong != 0) { 4011 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4012 if (waitTimeMs < mWaitTimeMs) { 4013 mWaitTimeMs = waitTimeMs; 4014 } 4015 } 4016 } 4017} 4018 4019 4020bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4021{ 4022 for (size_t i = 0; i < outputTracks.size(); i++) { 4023 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4024 if (thread == 0) { 4025 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4026 return false; 4027 } 4028 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4029 // see note at standby() declaration 4030 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4031 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4032 return false; 4033 } 4034 } 4035 return true; 4036} 4037 4038uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4039{ 4040 return (mWaitTimeMs * 1000) / 2; 4041} 4042 4043void AudioFlinger::DuplicatingThread::cacheParameters_l() 4044{ 4045 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4046 updateWaitTime_l(); 4047 4048 MixerThread::cacheParameters_l(); 4049} 4050 4051// ---------------------------------------------------------------------------- 4052 4053// TrackBase constructor must be called with AudioFlinger::mLock held 4054AudioFlinger::ThreadBase::TrackBase::TrackBase( 4055 ThreadBase *thread, 4056 const sp<Client>& client, 4057 uint32_t sampleRate, 4058 audio_format_t format, 4059 audio_channel_mask_t channelMask, 4060 int frameCount, 4061 const sp<IMemory>& sharedBuffer, 4062 int sessionId) 4063 : RefBase(), 4064 mThread(thread), 4065 mClient(client), 4066 mCblk(NULL), 4067 // mBuffer 4068 // mBufferEnd 4069 mFrameCount(0), 4070 mState(IDLE), 4071 mSampleRate(sampleRate), 4072 mFormat(format), 4073 mStepServerFailed(false), 4074 mSessionId(sessionId) 4075 // mChannelCount 4076 // mChannelMask 4077{ 4078 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4079 4080 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4081 size_t size = sizeof(audio_track_cblk_t); 4082 uint8_t channelCount = popcount(channelMask); 4083 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4084 if (sharedBuffer == 0) { 4085 size += bufferSize; 4086 } 4087 4088 if (client != NULL) { 4089 mCblkMemory = client->heap()->allocate(size); 4090 if (mCblkMemory != 0) { 4091 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4092 if (mCblk != NULL) { // construct the shared structure in-place. 4093 new(mCblk) audio_track_cblk_t(); 4094 // clear all buffers 4095 mCblk->frameCount = frameCount; 4096 mCblk->sampleRate = sampleRate; 4097// uncomment the following lines to quickly test 32-bit wraparound 4098// mCblk->user = 0xffff0000; 4099// mCblk->server = 0xffff0000; 4100// mCblk->userBase = 0xffff0000; 4101// mCblk->serverBase = 0xffff0000; 4102 mChannelCount = channelCount; 4103 mChannelMask = channelMask; 4104 if (sharedBuffer == 0) { 4105 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4106 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4107 // Force underrun condition to avoid false underrun callback until first data is 4108 // written to buffer (other flags are cleared) 4109 mCblk->flags = CBLK_UNDERRUN_ON; 4110 } else { 4111 mBuffer = sharedBuffer->pointer(); 4112 } 4113 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4114 } 4115 } else { 4116 ALOGE("not enough memory for AudioTrack size=%u", size); 4117 client->heap()->dump("AudioTrack"); 4118 return; 4119 } 4120 } else { 4121 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4122 // construct the shared structure in-place. 4123 new(mCblk) audio_track_cblk_t(); 4124 // clear all buffers 4125 mCblk->frameCount = frameCount; 4126 mCblk->sampleRate = sampleRate; 4127// uncomment the following lines to quickly test 32-bit wraparound 4128// mCblk->user = 0xffff0000; 4129// mCblk->server = 0xffff0000; 4130// mCblk->userBase = 0xffff0000; 4131// mCblk->serverBase = 0xffff0000; 4132 mChannelCount = channelCount; 4133 mChannelMask = channelMask; 4134 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4135 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4136 // Force underrun condition to avoid false underrun callback until first data is 4137 // written to buffer (other flags are cleared) 4138 mCblk->flags = CBLK_UNDERRUN_ON; 4139 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4140 } 4141} 4142 4143AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4144{ 4145 if (mCblk != NULL) { 4146 if (mClient == 0) { 4147 delete mCblk; 4148 } else { 4149 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4150 } 4151 } 4152 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4153 if (mClient != 0) { 4154 // Client destructor must run with AudioFlinger mutex locked 4155 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4156 // If the client's reference count drops to zero, the associated destructor 4157 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4158 // relying on the automatic clear() at end of scope. 4159 mClient.clear(); 4160 } 4161} 4162 4163// AudioBufferProvider interface 4164// getNextBuffer() = 0; 4165// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4166void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4167{ 4168 buffer->raw = NULL; 4169 mFrameCount = buffer->frameCount; 4170 // FIXME See note at getNextBuffer() 4171 (void) step(); // ignore return value of step() 4172 buffer->frameCount = 0; 4173} 4174 4175bool AudioFlinger::ThreadBase::TrackBase::step() { 4176 bool result; 4177 audio_track_cblk_t* cblk = this->cblk(); 4178 4179 result = cblk->stepServer(mFrameCount); 4180 if (!result) { 4181 ALOGV("stepServer failed acquiring cblk mutex"); 4182 mStepServerFailed = true; 4183 } 4184 return result; 4185} 4186 4187void AudioFlinger::ThreadBase::TrackBase::reset() { 4188 audio_track_cblk_t* cblk = this->cblk(); 4189 4190 cblk->user = 0; 4191 cblk->server = 0; 4192 cblk->userBase = 0; 4193 cblk->serverBase = 0; 4194 mStepServerFailed = false; 4195 ALOGV("TrackBase::reset"); 4196} 4197 4198int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4199 return (int)mCblk->sampleRate; 4200} 4201 4202void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4203 audio_track_cblk_t* cblk = this->cblk(); 4204 size_t frameSize = cblk->frameSize; 4205 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4206 int8_t *bufferEnd = bufferStart + frames * frameSize; 4207 4208 // Check validity of returned pointer in case the track control block would have been corrupted. 4209 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4210 "TrackBase::getBuffer buffer out of range:\n" 4211 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4212 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4213 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4214 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4215 4216 return bufferStart; 4217} 4218 4219status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4220{ 4221 mSyncEvents.add(event); 4222 return NO_ERROR; 4223} 4224 4225// ---------------------------------------------------------------------------- 4226 4227// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4228AudioFlinger::PlaybackThread::Track::Track( 4229 PlaybackThread *thread, 4230 const sp<Client>& client, 4231 audio_stream_type_t streamType, 4232 uint32_t sampleRate, 4233 audio_format_t format, 4234 audio_channel_mask_t channelMask, 4235 int frameCount, 4236 const sp<IMemory>& sharedBuffer, 4237 int sessionId, 4238 IAudioFlinger::track_flags_t flags) 4239 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4240 mMute(false), 4241 mFillingUpStatus(FS_INVALID), 4242 // mRetryCount initialized later when needed 4243 mSharedBuffer(sharedBuffer), 4244 mStreamType(streamType), 4245 mName(-1), // see note below 4246 mMainBuffer(thread->mixBuffer()), 4247 mAuxBuffer(NULL), 4248 mAuxEffectId(0), mHasVolumeController(false), 4249 mPresentationCompleteFrames(0), 4250 mFlags(flags), 4251 mFastIndex(-1), 4252 mUnderrunCount(0), 4253 mCachedVolume(1.0) 4254{ 4255 if (mCblk != NULL) { 4256 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4257 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4258 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4259 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4260 mName = thread->getTrackName_l(channelMask); 4261 mCblk->mName = mName; 4262 if (mName < 0) { 4263 ALOGE("no more track names available"); 4264 return; 4265 } 4266 // only allocate a fast track index if we were able to allocate a normal track name 4267 if (flags & IAudioFlinger::TRACK_FAST) { 4268 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4269 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4270 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4271 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4272 // FIXME This is too eager. We allocate a fast track index before the 4273 // fast track becomes active. Since fast tracks are a scarce resource, 4274 // this means we are potentially denying other more important fast tracks from 4275 // being created. It would be better to allocate the index dynamically. 4276 mFastIndex = i; 4277 mCblk->mName = i; 4278 // Read the initial underruns because this field is never cleared by the fast mixer 4279 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4280 thread->mFastTrackAvailMask &= ~(1 << i); 4281 } 4282 } 4283 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4284} 4285 4286AudioFlinger::PlaybackThread::Track::~Track() 4287{ 4288 ALOGV("PlaybackThread::Track destructor"); 4289 sp<ThreadBase> thread = mThread.promote(); 4290 if (thread != 0) { 4291 Mutex::Autolock _l(thread->mLock); 4292 mState = TERMINATED; 4293 } 4294} 4295 4296void AudioFlinger::PlaybackThread::Track::destroy() 4297{ 4298 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4299 // by removing it from mTracks vector, so there is a risk that this Tracks's 4300 // destructor is called. As the destructor needs to lock mLock, 4301 // we must acquire a strong reference on this Track before locking mLock 4302 // here so that the destructor is called only when exiting this function. 4303 // On the other hand, as long as Track::destroy() is only called by 4304 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4305 // this Track with its member mTrack. 4306 sp<Track> keep(this); 4307 { // scope for mLock 4308 sp<ThreadBase> thread = mThread.promote(); 4309 if (thread != 0) { 4310 if (!isOutputTrack()) { 4311 if (mState == ACTIVE || mState == RESUMING) { 4312 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4313 4314#ifdef ADD_BATTERY_DATA 4315 // to track the speaker usage 4316 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4317#endif 4318 } 4319 AudioSystem::releaseOutput(thread->id()); 4320 } 4321 Mutex::Autolock _l(thread->mLock); 4322 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4323 playbackThread->destroyTrack_l(this); 4324 } 4325 } 4326} 4327 4328/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4329{ 4330 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4331 " Server User Main buf Aux Buf Flags Underruns\n"); 4332} 4333 4334void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4335{ 4336 uint32_t vlr = mCblk->getVolumeLR(); 4337 if (isFastTrack()) { 4338 sprintf(buffer, " F %2d", mFastIndex); 4339 } else { 4340 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4341 } 4342 track_state state = mState; 4343 char stateChar; 4344 switch (state) { 4345 case IDLE: 4346 stateChar = 'I'; 4347 break; 4348 case TERMINATED: 4349 stateChar = 'T'; 4350 break; 4351 case STOPPING_1: 4352 stateChar = 's'; 4353 break; 4354 case STOPPING_2: 4355 stateChar = '5'; 4356 break; 4357 case STOPPED: 4358 stateChar = 'S'; 4359 break; 4360 case RESUMING: 4361 stateChar = 'R'; 4362 break; 4363 case ACTIVE: 4364 stateChar = 'A'; 4365 break; 4366 case PAUSING: 4367 stateChar = 'p'; 4368 break; 4369 case PAUSED: 4370 stateChar = 'P'; 4371 break; 4372 case FLUSHED: 4373 stateChar = 'F'; 4374 break; 4375 default: 4376 stateChar = '?'; 4377 break; 4378 } 4379 char nowInUnderrun; 4380 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4381 case UNDERRUN_FULL: 4382 nowInUnderrun = ' '; 4383 break; 4384 case UNDERRUN_PARTIAL: 4385 nowInUnderrun = '<'; 4386 break; 4387 case UNDERRUN_EMPTY: 4388 nowInUnderrun = '*'; 4389 break; 4390 default: 4391 nowInUnderrun = '?'; 4392 break; 4393 } 4394 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4395 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4396 (mClient == 0) ? getpid_cached : mClient->pid(), 4397 mStreamType, 4398 mFormat, 4399 mChannelMask, 4400 mSessionId, 4401 mFrameCount, 4402 mCblk->frameCount, 4403 stateChar, 4404 mMute, 4405 mFillingUpStatus, 4406 mCblk->sampleRate, 4407 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4408 20.0 * log10((vlr >> 16) / 4096.0), 4409 mCblk->server, 4410 mCblk->user, 4411 (int)mMainBuffer, 4412 (int)mAuxBuffer, 4413 mCblk->flags, 4414 mUnderrunCount, 4415 nowInUnderrun); 4416} 4417 4418// AudioBufferProvider interface 4419status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4420 AudioBufferProvider::Buffer* buffer, int64_t pts) 4421{ 4422 audio_track_cblk_t* cblk = this->cblk(); 4423 uint32_t framesReady; 4424 uint32_t framesReq = buffer->frameCount; 4425 4426 // Check if last stepServer failed, try to step now 4427 if (mStepServerFailed) { 4428 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4429 // Since the fast mixer is higher priority than client callback thread, 4430 // it does not result in priority inversion for client. 4431 // But a non-blocking solution would be preferable to avoid 4432 // fast mixer being unable to tryLock(), and 4433 // to avoid the extra context switches if the client wakes up, 4434 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4435 if (!step()) goto getNextBuffer_exit; 4436 ALOGV("stepServer recovered"); 4437 mStepServerFailed = false; 4438 } 4439 4440 // FIXME Same as above 4441 framesReady = cblk->framesReady(); 4442 4443 if (CC_LIKELY(framesReady)) { 4444 uint32_t s = cblk->server; 4445 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4446 4447 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4448 if (framesReq > framesReady) { 4449 framesReq = framesReady; 4450 } 4451 if (framesReq > bufferEnd - s) { 4452 framesReq = bufferEnd - s; 4453 } 4454 4455 buffer->raw = getBuffer(s, framesReq); 4456 buffer->frameCount = framesReq; 4457 return NO_ERROR; 4458 } 4459 4460getNextBuffer_exit: 4461 buffer->raw = NULL; 4462 buffer->frameCount = 0; 4463 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4464 return NOT_ENOUGH_DATA; 4465} 4466 4467// Note that framesReady() takes a mutex on the control block using tryLock(). 4468// This could result in priority inversion if framesReady() is called by the normal mixer, 4469// as the normal mixer thread runs at lower 4470// priority than the client's callback thread: there is a short window within framesReady() 4471// during which the normal mixer could be preempted, and the client callback would block. 4472// Another problem can occur if framesReady() is called by the fast mixer: 4473// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4474// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4475size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4476 return mCblk->framesReady(); 4477} 4478 4479// Don't call for fast tracks; the framesReady() could result in priority inversion 4480bool AudioFlinger::PlaybackThread::Track::isReady() const { 4481 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4482 4483 if (framesReady() >= mCblk->frameCount || 4484 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4485 mFillingUpStatus = FS_FILLED; 4486 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4487 return true; 4488 } 4489 return false; 4490} 4491 4492status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4493 int triggerSession) 4494{ 4495 status_t status = NO_ERROR; 4496 ALOGV("start(%d), calling pid %d session %d", 4497 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4498 4499 sp<ThreadBase> thread = mThread.promote(); 4500 if (thread != 0) { 4501 Mutex::Autolock _l(thread->mLock); 4502 track_state state = mState; 4503 // here the track could be either new, or restarted 4504 // in both cases "unstop" the track 4505 if (mState == PAUSED) { 4506 mState = TrackBase::RESUMING; 4507 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4508 } else { 4509 mState = TrackBase::ACTIVE; 4510 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4511 } 4512 4513 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4514 thread->mLock.unlock(); 4515 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4516 thread->mLock.lock(); 4517 4518#ifdef ADD_BATTERY_DATA 4519 // to track the speaker usage 4520 if (status == NO_ERROR) { 4521 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4522 } 4523#endif 4524 } 4525 if (status == NO_ERROR) { 4526 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4527 playbackThread->addTrack_l(this); 4528 } else { 4529 mState = state; 4530 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4531 } 4532 } else { 4533 status = BAD_VALUE; 4534 } 4535 return status; 4536} 4537 4538void AudioFlinger::PlaybackThread::Track::stop() 4539{ 4540 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4541 sp<ThreadBase> thread = mThread.promote(); 4542 if (thread != 0) { 4543 Mutex::Autolock _l(thread->mLock); 4544 track_state state = mState; 4545 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4546 // If the track is not active (PAUSED and buffers full), flush buffers 4547 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4548 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4549 reset(); 4550 mState = STOPPED; 4551 } else if (!isFastTrack()) { 4552 mState = STOPPED; 4553 } else { 4554 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4555 // and then to STOPPED and reset() when presentation is complete 4556 mState = STOPPING_1; 4557 } 4558 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4559 } 4560 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4561 thread->mLock.unlock(); 4562 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4563 thread->mLock.lock(); 4564 4565#ifdef ADD_BATTERY_DATA 4566 // to track the speaker usage 4567 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4568#endif 4569 } 4570 } 4571} 4572 4573void AudioFlinger::PlaybackThread::Track::pause() 4574{ 4575 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4576 sp<ThreadBase> thread = mThread.promote(); 4577 if (thread != 0) { 4578 Mutex::Autolock _l(thread->mLock); 4579 if (mState == ACTIVE || mState == RESUMING) { 4580 mState = PAUSING; 4581 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4582 if (!isOutputTrack()) { 4583 thread->mLock.unlock(); 4584 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4585 thread->mLock.lock(); 4586 4587#ifdef ADD_BATTERY_DATA 4588 // to track the speaker usage 4589 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4590#endif 4591 } 4592 } 4593 } 4594} 4595 4596void AudioFlinger::PlaybackThread::Track::flush() 4597{ 4598 ALOGV("flush(%d)", mName); 4599 sp<ThreadBase> thread = mThread.promote(); 4600 if (thread != 0) { 4601 Mutex::Autolock _l(thread->mLock); 4602 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4603 mState != PAUSING) { 4604 return; 4605 } 4606 // No point remaining in PAUSED state after a flush => go to 4607 // FLUSHED state 4608 mState = FLUSHED; 4609 // do not reset the track if it is still in the process of being stopped or paused. 4610 // this will be done by prepareTracks_l() when the track is stopped. 4611 // prepareTracks_l() will see mState == FLUSHED, then 4612 // remove from active track list, reset(), and trigger presentation complete 4613 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4614 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4615 reset(); 4616 } 4617 } 4618} 4619 4620void AudioFlinger::PlaybackThread::Track::reset() 4621{ 4622 // Do not reset twice to avoid discarding data written just after a flush and before 4623 // the audioflinger thread detects the track is stopped. 4624 if (!mResetDone) { 4625 TrackBase::reset(); 4626 // Force underrun condition to avoid false underrun callback until first data is 4627 // written to buffer 4628 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4629 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4630 mFillingUpStatus = FS_FILLING; 4631 mResetDone = true; 4632 if (mState == FLUSHED) { 4633 mState = IDLE; 4634 } 4635 } 4636} 4637 4638void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4639{ 4640 mMute = muted; 4641} 4642 4643status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4644{ 4645 status_t status = DEAD_OBJECT; 4646 sp<ThreadBase> thread = mThread.promote(); 4647 if (thread != 0) { 4648 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4649 sp<AudioFlinger> af = mClient->audioFlinger(); 4650 4651 Mutex::Autolock _l(af->mLock); 4652 4653 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4654 4655 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4656 Mutex::Autolock _dl(playbackThread->mLock); 4657 Mutex::Autolock _sl(srcThread->mLock); 4658 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4659 if (chain == 0) { 4660 return INVALID_OPERATION; 4661 } 4662 4663 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4664 if (effect == 0) { 4665 return INVALID_OPERATION; 4666 } 4667 srcThread->removeEffect_l(effect); 4668 playbackThread->addEffect_l(effect); 4669 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4670 if (effect->state() == EffectModule::ACTIVE || 4671 effect->state() == EffectModule::STOPPING) { 4672 effect->start(); 4673 } 4674 4675 sp<EffectChain> dstChain = effect->chain().promote(); 4676 if (dstChain == 0) { 4677 srcThread->addEffect_l(effect); 4678 return INVALID_OPERATION; 4679 } 4680 AudioSystem::unregisterEffect(effect->id()); 4681 AudioSystem::registerEffect(&effect->desc(), 4682 srcThread->id(), 4683 dstChain->strategy(), 4684 AUDIO_SESSION_OUTPUT_MIX, 4685 effect->id()); 4686 } 4687 status = playbackThread->attachAuxEffect(this, EffectId); 4688 } 4689 return status; 4690} 4691 4692void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4693{ 4694 mAuxEffectId = EffectId; 4695 mAuxBuffer = buffer; 4696} 4697 4698bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4699 size_t audioHalFrames) 4700{ 4701 // a track is considered presented when the total number of frames written to audio HAL 4702 // corresponds to the number of frames written when presentationComplete() is called for the 4703 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4704 if (mPresentationCompleteFrames == 0) { 4705 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4706 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4707 mPresentationCompleteFrames, audioHalFrames); 4708 } 4709 if (framesWritten >= mPresentationCompleteFrames) { 4710 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4711 mSessionId, framesWritten); 4712 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4713 return true; 4714 } 4715 return false; 4716} 4717 4718void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4719{ 4720 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4721 if (mSyncEvents[i]->type() == type) { 4722 mSyncEvents[i]->trigger(); 4723 mSyncEvents.removeAt(i); 4724 i--; 4725 } 4726 } 4727} 4728 4729// implement VolumeBufferProvider interface 4730 4731uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4732{ 4733 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4734 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4735 uint32_t vlr = mCblk->getVolumeLR(); 4736 uint32_t vl = vlr & 0xFFFF; 4737 uint32_t vr = vlr >> 16; 4738 // track volumes come from shared memory, so can't be trusted and must be clamped 4739 if (vl > MAX_GAIN_INT) { 4740 vl = MAX_GAIN_INT; 4741 } 4742 if (vr > MAX_GAIN_INT) { 4743 vr = MAX_GAIN_INT; 4744 } 4745 // now apply the cached master volume and stream type volume; 4746 // this is trusted but lacks any synchronization or barrier so may be stale 4747 float v = mCachedVolume; 4748 vl *= v; 4749 vr *= v; 4750 // re-combine into U4.16 4751 vlr = (vr << 16) | (vl & 0xFFFF); 4752 // FIXME look at mute, pause, and stop flags 4753 return vlr; 4754} 4755 4756status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4757{ 4758 if (mState == TERMINATED || mState == PAUSED || 4759 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4760 (mState == STOPPED)))) { 4761 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4762 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4763 event->cancel(); 4764 return INVALID_OPERATION; 4765 } 4766 TrackBase::setSyncEvent(event); 4767 return NO_ERROR; 4768} 4769 4770// timed audio tracks 4771 4772sp<AudioFlinger::PlaybackThread::TimedTrack> 4773AudioFlinger::PlaybackThread::TimedTrack::create( 4774 PlaybackThread *thread, 4775 const sp<Client>& client, 4776 audio_stream_type_t streamType, 4777 uint32_t sampleRate, 4778 audio_format_t format, 4779 audio_channel_mask_t channelMask, 4780 int frameCount, 4781 const sp<IMemory>& sharedBuffer, 4782 int sessionId) { 4783 if (!client->reserveTimedTrack()) 4784 return 0; 4785 4786 return new TimedTrack( 4787 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4788 sharedBuffer, sessionId); 4789} 4790 4791AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4792 PlaybackThread *thread, 4793 const sp<Client>& client, 4794 audio_stream_type_t streamType, 4795 uint32_t sampleRate, 4796 audio_format_t format, 4797 audio_channel_mask_t channelMask, 4798 int frameCount, 4799 const sp<IMemory>& sharedBuffer, 4800 int sessionId) 4801 : Track(thread, client, streamType, sampleRate, format, channelMask, 4802 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4803 mQueueHeadInFlight(false), 4804 mTrimQueueHeadOnRelease(false), 4805 mFramesPendingInQueue(0), 4806 mTimedSilenceBuffer(NULL), 4807 mTimedSilenceBufferSize(0), 4808 mTimedAudioOutputOnTime(false), 4809 mMediaTimeTransformValid(false) 4810{ 4811 LocalClock lc; 4812 mLocalTimeFreq = lc.getLocalFreq(); 4813 4814 mLocalTimeToSampleTransform.a_zero = 0; 4815 mLocalTimeToSampleTransform.b_zero = 0; 4816 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4817 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4818 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4819 &mLocalTimeToSampleTransform.a_to_b_denom); 4820 4821 mMediaTimeToSampleTransform.a_zero = 0; 4822 mMediaTimeToSampleTransform.b_zero = 0; 4823 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4824 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4825 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4826 &mMediaTimeToSampleTransform.a_to_b_denom); 4827} 4828 4829AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4830 mClient->releaseTimedTrack(); 4831 delete [] mTimedSilenceBuffer; 4832} 4833 4834status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4835 size_t size, sp<IMemory>* buffer) { 4836 4837 Mutex::Autolock _l(mTimedBufferQueueLock); 4838 4839 trimTimedBufferQueue_l(); 4840 4841 // lazily initialize the shared memory heap for timed buffers 4842 if (mTimedMemoryDealer == NULL) { 4843 const int kTimedBufferHeapSize = 512 << 10; 4844 4845 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4846 "AudioFlingerTimed"); 4847 if (mTimedMemoryDealer == NULL) 4848 return NO_MEMORY; 4849 } 4850 4851 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4852 if (newBuffer == NULL) { 4853 newBuffer = mTimedMemoryDealer->allocate(size); 4854 if (newBuffer == NULL) 4855 return NO_MEMORY; 4856 } 4857 4858 *buffer = newBuffer; 4859 return NO_ERROR; 4860} 4861 4862// caller must hold mTimedBufferQueueLock 4863void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4864 int64_t mediaTimeNow; 4865 { 4866 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4867 if (!mMediaTimeTransformValid) 4868 return; 4869 4870 int64_t targetTimeNow; 4871 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4872 ? mCCHelper.getCommonTime(&targetTimeNow) 4873 : mCCHelper.getLocalTime(&targetTimeNow); 4874 4875 if (OK != res) 4876 return; 4877 4878 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4879 &mediaTimeNow)) { 4880 return; 4881 } 4882 } 4883 4884 size_t trimEnd; 4885 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4886 int64_t bufEnd; 4887 4888 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4889 // We have a next buffer. Just use its PTS as the PTS of the frame 4890 // following the last frame in this buffer. If the stream is sparse 4891 // (ie, there are deliberate gaps left in the stream which should be 4892 // filled with silence by the TimedAudioTrack), then this can result 4893 // in one extra buffer being left un-trimmed when it could have 4894 // been. In general, this is not typical, and we would rather 4895 // optimized away the TS calculation below for the more common case 4896 // where PTSes are contiguous. 4897 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4898 } else { 4899 // We have no next buffer. Compute the PTS of the frame following 4900 // the last frame in this buffer by computing the duration of of 4901 // this frame in media time units and adding it to the PTS of the 4902 // buffer. 4903 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4904 / mCblk->frameSize; 4905 4906 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4907 &bufEnd)) { 4908 ALOGE("Failed to convert frame count of %lld to media time" 4909 " duration" " (scale factor %d/%u) in %s", 4910 frameCount, 4911 mMediaTimeToSampleTransform.a_to_b_numer, 4912 mMediaTimeToSampleTransform.a_to_b_denom, 4913 __PRETTY_FUNCTION__); 4914 break; 4915 } 4916 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4917 } 4918 4919 if (bufEnd > mediaTimeNow) 4920 break; 4921 4922 // Is the buffer we want to use in the middle of a mix operation right 4923 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4924 // from the mixer which should be coming back shortly. 4925 if (!trimEnd && mQueueHeadInFlight) { 4926 mTrimQueueHeadOnRelease = true; 4927 } 4928 } 4929 4930 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4931 if (trimStart < trimEnd) { 4932 // Update the bookkeeping for framesReady() 4933 for (size_t i = trimStart; i < trimEnd; ++i) { 4934 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4935 } 4936 4937 // Now actually remove the buffers from the queue. 4938 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4939 } 4940} 4941 4942void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4943 const char* logTag) { 4944 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4945 "%s called (reason \"%s\"), but timed buffer queue has no" 4946 " elements to trim.", __FUNCTION__, logTag); 4947 4948 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4949 mTimedBufferQueue.removeAt(0); 4950} 4951 4952void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4953 const TimedBuffer& buf, 4954 const char* logTag) { 4955 uint32_t bufBytes = buf.buffer()->size(); 4956 uint32_t consumedAlready = buf.position(); 4957 4958 ALOG_ASSERT(consumedAlready <= bufBytes, 4959 "Bad bookkeeping while updating frames pending. Timed buffer is" 4960 " only %u bytes long, but claims to have consumed %u" 4961 " bytes. (update reason: \"%s\")", 4962 bufBytes, consumedAlready, logTag); 4963 4964 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4965 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4966 "Bad bookkeeping while updating frames pending. Should have at" 4967 " least %u queued frames, but we think we have only %u. (update" 4968 " reason: \"%s\")", 4969 bufFrames, mFramesPendingInQueue, logTag); 4970 4971 mFramesPendingInQueue -= bufFrames; 4972} 4973 4974status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4975 const sp<IMemory>& buffer, int64_t pts) { 4976 4977 { 4978 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4979 if (!mMediaTimeTransformValid) 4980 return INVALID_OPERATION; 4981 } 4982 4983 Mutex::Autolock _l(mTimedBufferQueueLock); 4984 4985 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4986 mFramesPendingInQueue += bufFrames; 4987 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4988 4989 return NO_ERROR; 4990} 4991 4992status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4993 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4994 4995 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4996 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4997 target); 4998 4999 if (!(target == TimedAudioTrack::LOCAL_TIME || 5000 target == TimedAudioTrack::COMMON_TIME)) { 5001 return BAD_VALUE; 5002 } 5003 5004 Mutex::Autolock lock(mMediaTimeTransformLock); 5005 mMediaTimeTransform = xform; 5006 mMediaTimeTransformTarget = target; 5007 mMediaTimeTransformValid = true; 5008 5009 return NO_ERROR; 5010} 5011 5012#define min(a, b) ((a) < (b) ? (a) : (b)) 5013 5014// implementation of getNextBuffer for tracks whose buffers have timestamps 5015status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5016 AudioBufferProvider::Buffer* buffer, int64_t pts) 5017{ 5018 if (pts == AudioBufferProvider::kInvalidPTS) { 5019 buffer->raw = NULL; 5020 buffer->frameCount = 0; 5021 mTimedAudioOutputOnTime = false; 5022 return INVALID_OPERATION; 5023 } 5024 5025 Mutex::Autolock _l(mTimedBufferQueueLock); 5026 5027 ALOG_ASSERT(!mQueueHeadInFlight, 5028 "getNextBuffer called without releaseBuffer!"); 5029 5030 while (true) { 5031 5032 // if we have no timed buffers, then fail 5033 if (mTimedBufferQueue.isEmpty()) { 5034 buffer->raw = NULL; 5035 buffer->frameCount = 0; 5036 return NOT_ENOUGH_DATA; 5037 } 5038 5039 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5040 5041 // calculate the PTS of the head of the timed buffer queue expressed in 5042 // local time 5043 int64_t headLocalPTS; 5044 { 5045 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5046 5047 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5048 5049 if (mMediaTimeTransform.a_to_b_denom == 0) { 5050 // the transform represents a pause, so yield silence 5051 timedYieldSilence_l(buffer->frameCount, buffer); 5052 return NO_ERROR; 5053 } 5054 5055 int64_t transformedPTS; 5056 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5057 &transformedPTS)) { 5058 // the transform failed. this shouldn't happen, but if it does 5059 // then just drop this buffer 5060 ALOGW("timedGetNextBuffer transform failed"); 5061 buffer->raw = NULL; 5062 buffer->frameCount = 0; 5063 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5064 return NO_ERROR; 5065 } 5066 5067 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5068 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5069 &headLocalPTS)) { 5070 buffer->raw = NULL; 5071 buffer->frameCount = 0; 5072 return INVALID_OPERATION; 5073 } 5074 } else { 5075 headLocalPTS = transformedPTS; 5076 } 5077 } 5078 5079 // adjust the head buffer's PTS to reflect the portion of the head buffer 5080 // that has already been consumed 5081 int64_t effectivePTS = headLocalPTS + 5082 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5083 5084 // Calculate the delta in samples between the head of the input buffer 5085 // queue and the start of the next output buffer that will be written. 5086 // If the transformation fails because of over or underflow, it means 5087 // that the sample's position in the output stream is so far out of 5088 // whack that it should just be dropped. 5089 int64_t sampleDelta; 5090 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5091 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5092 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5093 " mix"); 5094 continue; 5095 } 5096 if (!mLocalTimeToSampleTransform.doForwardTransform( 5097 (effectivePTS - pts) << 32, &sampleDelta)) { 5098 ALOGV("*** too late during sample rate transform: dropped buffer"); 5099 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5100 continue; 5101 } 5102 5103 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5104 " sampleDelta=[%d.%08x]", 5105 head.pts(), head.position(), pts, 5106 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5107 + (sampleDelta >> 32)), 5108 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5109 5110 // if the delta between the ideal placement for the next input sample and 5111 // the current output position is within this threshold, then we will 5112 // concatenate the next input samples to the previous output 5113 const int64_t kSampleContinuityThreshold = 5114 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5115 5116 // if this is the first buffer of audio that we're emitting from this track 5117 // then it should be almost exactly on time. 5118 const int64_t kSampleStartupThreshold = 1LL << 32; 5119 5120 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5121 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5122 // the next input is close enough to being on time, so concatenate it 5123 // with the last output 5124 timedYieldSamples_l(buffer); 5125 5126 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5127 head.position(), buffer->frameCount); 5128 return NO_ERROR; 5129 } 5130 5131 // Looks like our output is not on time. Reset our on timed status. 5132 // Next time we mix samples from our input queue, then should be within 5133 // the StartupThreshold. 5134 mTimedAudioOutputOnTime = false; 5135 if (sampleDelta > 0) { 5136 // the gap between the current output position and the proper start of 5137 // the next input sample is too big, so fill it with silence 5138 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5139 5140 timedYieldSilence_l(framesUntilNextInput, buffer); 5141 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5142 return NO_ERROR; 5143 } else { 5144 // the next input sample is late 5145 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5146 size_t onTimeSamplePosition = 5147 head.position() + lateFrames * mCblk->frameSize; 5148 5149 if (onTimeSamplePosition > head.buffer()->size()) { 5150 // all the remaining samples in the head are too late, so 5151 // drop it and move on 5152 ALOGV("*** too late: dropped buffer"); 5153 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5154 continue; 5155 } else { 5156 // skip over the late samples 5157 head.setPosition(onTimeSamplePosition); 5158 5159 // yield the available samples 5160 timedYieldSamples_l(buffer); 5161 5162 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5163 return NO_ERROR; 5164 } 5165 } 5166 } 5167} 5168 5169// Yield samples from the timed buffer queue head up to the given output 5170// buffer's capacity. 5171// 5172// Caller must hold mTimedBufferQueueLock 5173void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5174 AudioBufferProvider::Buffer* buffer) { 5175 5176 const TimedBuffer& head = mTimedBufferQueue[0]; 5177 5178 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5179 head.position()); 5180 5181 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5182 mCblk->frameSize); 5183 size_t framesRequested = buffer->frameCount; 5184 buffer->frameCount = min(framesLeftInHead, framesRequested); 5185 5186 mQueueHeadInFlight = true; 5187 mTimedAudioOutputOnTime = true; 5188} 5189 5190// Yield samples of silence up to the given output buffer's capacity 5191// 5192// Caller must hold mTimedBufferQueueLock 5193void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5194 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5195 5196 // lazily allocate a buffer filled with silence 5197 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5198 delete [] mTimedSilenceBuffer; 5199 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5200 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5201 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5202 } 5203 5204 buffer->raw = mTimedSilenceBuffer; 5205 size_t framesRequested = buffer->frameCount; 5206 buffer->frameCount = min(numFrames, framesRequested); 5207 5208 mTimedAudioOutputOnTime = false; 5209} 5210 5211// AudioBufferProvider interface 5212void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5213 AudioBufferProvider::Buffer* buffer) { 5214 5215 Mutex::Autolock _l(mTimedBufferQueueLock); 5216 5217 // If the buffer which was just released is part of the buffer at the head 5218 // of the queue, be sure to update the amt of the buffer which has been 5219 // consumed. If the buffer being returned is not part of the head of the 5220 // queue, its either because the buffer is part of the silence buffer, or 5221 // because the head of the timed queue was trimmed after the mixer called 5222 // getNextBuffer but before the mixer called releaseBuffer. 5223 if (buffer->raw == mTimedSilenceBuffer) { 5224 ALOG_ASSERT(!mQueueHeadInFlight, 5225 "Queue head in flight during release of silence buffer!"); 5226 goto done; 5227 } 5228 5229 ALOG_ASSERT(mQueueHeadInFlight, 5230 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5231 " head in flight."); 5232 5233 if (mTimedBufferQueue.size()) { 5234 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5235 5236 void* start = head.buffer()->pointer(); 5237 void* end = reinterpret_cast<void*>( 5238 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5239 + head.buffer()->size()); 5240 5241 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5242 "released buffer not within the head of the timed buffer" 5243 " queue; qHead = [%p, %p], released buffer = %p", 5244 start, end, buffer->raw); 5245 5246 head.setPosition(head.position() + 5247 (buffer->frameCount * mCblk->frameSize)); 5248 mQueueHeadInFlight = false; 5249 5250 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5251 "Bad bookkeeping during releaseBuffer! Should have at" 5252 " least %u queued frames, but we think we have only %u", 5253 buffer->frameCount, mFramesPendingInQueue); 5254 5255 mFramesPendingInQueue -= buffer->frameCount; 5256 5257 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5258 || mTrimQueueHeadOnRelease) { 5259 trimTimedBufferQueueHead_l("releaseBuffer"); 5260 mTrimQueueHeadOnRelease = false; 5261 } 5262 } else { 5263 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5264 " buffers in the timed buffer queue"); 5265 } 5266 5267done: 5268 buffer->raw = 0; 5269 buffer->frameCount = 0; 5270} 5271 5272size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5273 Mutex::Autolock _l(mTimedBufferQueueLock); 5274 return mFramesPendingInQueue; 5275} 5276 5277AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5278 : mPTS(0), mPosition(0) {} 5279 5280AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5281 const sp<IMemory>& buffer, int64_t pts) 5282 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5283 5284// ---------------------------------------------------------------------------- 5285 5286// RecordTrack constructor must be called with AudioFlinger::mLock held 5287AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5288 RecordThread *thread, 5289 const sp<Client>& client, 5290 uint32_t sampleRate, 5291 audio_format_t format, 5292 audio_channel_mask_t channelMask, 5293 int frameCount, 5294 int sessionId) 5295 : TrackBase(thread, client, sampleRate, format, 5296 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5297 mOverflow(false) 5298{ 5299 if (mCblk != NULL) { 5300 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5301 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5302 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5303 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5304 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5305 } else { 5306 mCblk->frameSize = sizeof(int8_t); 5307 } 5308 } 5309} 5310 5311AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5312{ 5313 sp<ThreadBase> thread = mThread.promote(); 5314 if (thread != 0) { 5315 AudioSystem::releaseInput(thread->id()); 5316 } 5317} 5318 5319// AudioBufferProvider interface 5320status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5321{ 5322 audio_track_cblk_t* cblk = this->cblk(); 5323 uint32_t framesAvail; 5324 uint32_t framesReq = buffer->frameCount; 5325 5326 // Check if last stepServer failed, try to step now 5327 if (mStepServerFailed) { 5328 if (!step()) goto getNextBuffer_exit; 5329 ALOGV("stepServer recovered"); 5330 mStepServerFailed = false; 5331 } 5332 5333 framesAvail = cblk->framesAvailable_l(); 5334 5335 if (CC_LIKELY(framesAvail)) { 5336 uint32_t s = cblk->server; 5337 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5338 5339 if (framesReq > framesAvail) { 5340 framesReq = framesAvail; 5341 } 5342 if (framesReq > bufferEnd - s) { 5343 framesReq = bufferEnd - s; 5344 } 5345 5346 buffer->raw = getBuffer(s, framesReq); 5347 buffer->frameCount = framesReq; 5348 return NO_ERROR; 5349 } 5350 5351getNextBuffer_exit: 5352 buffer->raw = NULL; 5353 buffer->frameCount = 0; 5354 return NOT_ENOUGH_DATA; 5355} 5356 5357status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5358 int triggerSession) 5359{ 5360 sp<ThreadBase> thread = mThread.promote(); 5361 if (thread != 0) { 5362 RecordThread *recordThread = (RecordThread *)thread.get(); 5363 return recordThread->start(this, event, triggerSession); 5364 } else { 5365 return BAD_VALUE; 5366 } 5367} 5368 5369void AudioFlinger::RecordThread::RecordTrack::stop() 5370{ 5371 sp<ThreadBase> thread = mThread.promote(); 5372 if (thread != 0) { 5373 RecordThread *recordThread = (RecordThread *)thread.get(); 5374 recordThread->stop(this); 5375 TrackBase::reset(); 5376 // Force overrun condition to avoid false overrun callback until first data is 5377 // read from buffer 5378 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5379 } 5380} 5381 5382void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5383{ 5384 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5385 (mClient == 0) ? getpid_cached : mClient->pid(), 5386 mFormat, 5387 mChannelMask, 5388 mSessionId, 5389 mFrameCount, 5390 mState, 5391 mCblk->sampleRate, 5392 mCblk->server, 5393 mCblk->user); 5394} 5395 5396 5397// ---------------------------------------------------------------------------- 5398 5399AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5400 PlaybackThread *playbackThread, 5401 DuplicatingThread *sourceThread, 5402 uint32_t sampleRate, 5403 audio_format_t format, 5404 audio_channel_mask_t channelMask, 5405 int frameCount) 5406 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5407 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5408 mActive(false), mSourceThread(sourceThread) 5409{ 5410 5411 if (mCblk != NULL) { 5412 mCblk->flags |= CBLK_DIRECTION_OUT; 5413 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5414 mOutBuffer.frameCount = 0; 5415 playbackThread->mTracks.add(this); 5416 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5417 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5418 mCblk, mBuffer, mCblk->buffers, 5419 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5420 } else { 5421 ALOGW("Error creating output track on thread %p", playbackThread); 5422 } 5423} 5424 5425AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5426{ 5427 clearBufferQueue(); 5428} 5429 5430status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5431 int triggerSession) 5432{ 5433 status_t status = Track::start(event, triggerSession); 5434 if (status != NO_ERROR) { 5435 return status; 5436 } 5437 5438 mActive = true; 5439 mRetryCount = 127; 5440 return status; 5441} 5442 5443void AudioFlinger::PlaybackThread::OutputTrack::stop() 5444{ 5445 Track::stop(); 5446 clearBufferQueue(); 5447 mOutBuffer.frameCount = 0; 5448 mActive = false; 5449} 5450 5451bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5452{ 5453 Buffer *pInBuffer; 5454 Buffer inBuffer; 5455 uint32_t channelCount = mChannelCount; 5456 bool outputBufferFull = false; 5457 inBuffer.frameCount = frames; 5458 inBuffer.i16 = data; 5459 5460 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5461 5462 if (!mActive && frames != 0) { 5463 start(); 5464 sp<ThreadBase> thread = mThread.promote(); 5465 if (thread != 0) { 5466 MixerThread *mixerThread = (MixerThread *)thread.get(); 5467 if (mCblk->frameCount > frames){ 5468 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5469 uint32_t startFrames = (mCblk->frameCount - frames); 5470 pInBuffer = new Buffer; 5471 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5472 pInBuffer->frameCount = startFrames; 5473 pInBuffer->i16 = pInBuffer->mBuffer; 5474 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5475 mBufferQueue.add(pInBuffer); 5476 } else { 5477 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5478 } 5479 } 5480 } 5481 } 5482 5483 while (waitTimeLeftMs) { 5484 // First write pending buffers, then new data 5485 if (mBufferQueue.size()) { 5486 pInBuffer = mBufferQueue.itemAt(0); 5487 } else { 5488 pInBuffer = &inBuffer; 5489 } 5490 5491 if (pInBuffer->frameCount == 0) { 5492 break; 5493 } 5494 5495 if (mOutBuffer.frameCount == 0) { 5496 mOutBuffer.frameCount = pInBuffer->frameCount; 5497 nsecs_t startTime = systemTime(); 5498 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5499 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5500 outputBufferFull = true; 5501 break; 5502 } 5503 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5504 if (waitTimeLeftMs >= waitTimeMs) { 5505 waitTimeLeftMs -= waitTimeMs; 5506 } else { 5507 waitTimeLeftMs = 0; 5508 } 5509 } 5510 5511 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5512 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5513 mCblk->stepUser(outFrames); 5514 pInBuffer->frameCount -= outFrames; 5515 pInBuffer->i16 += outFrames * channelCount; 5516 mOutBuffer.frameCount -= outFrames; 5517 mOutBuffer.i16 += outFrames * channelCount; 5518 5519 if (pInBuffer->frameCount == 0) { 5520 if (mBufferQueue.size()) { 5521 mBufferQueue.removeAt(0); 5522 delete [] pInBuffer->mBuffer; 5523 delete pInBuffer; 5524 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5525 } else { 5526 break; 5527 } 5528 } 5529 } 5530 5531 // If we could not write all frames, allocate a buffer and queue it for next time. 5532 if (inBuffer.frameCount) { 5533 sp<ThreadBase> thread = mThread.promote(); 5534 if (thread != 0 && !thread->standby()) { 5535 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5536 pInBuffer = new Buffer; 5537 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5538 pInBuffer->frameCount = inBuffer.frameCount; 5539 pInBuffer->i16 = pInBuffer->mBuffer; 5540 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5541 mBufferQueue.add(pInBuffer); 5542 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5543 } else { 5544 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5545 } 5546 } 5547 } 5548 5549 // Calling write() with a 0 length buffer, means that no more data will be written: 5550 // If no more buffers are pending, fill output track buffer to make sure it is started 5551 // by output mixer. 5552 if (frames == 0 && mBufferQueue.size() == 0) { 5553 if (mCblk->user < mCblk->frameCount) { 5554 frames = mCblk->frameCount - mCblk->user; 5555 pInBuffer = new Buffer; 5556 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5557 pInBuffer->frameCount = frames; 5558 pInBuffer->i16 = pInBuffer->mBuffer; 5559 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5560 mBufferQueue.add(pInBuffer); 5561 } else if (mActive) { 5562 stop(); 5563 } 5564 } 5565 5566 return outputBufferFull; 5567} 5568 5569status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5570{ 5571 int active; 5572 status_t result; 5573 audio_track_cblk_t* cblk = mCblk; 5574 uint32_t framesReq = buffer->frameCount; 5575 5576// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5577 buffer->frameCount = 0; 5578 5579 uint32_t framesAvail = cblk->framesAvailable(); 5580 5581 5582 if (framesAvail == 0) { 5583 Mutex::Autolock _l(cblk->lock); 5584 goto start_loop_here; 5585 while (framesAvail == 0) { 5586 active = mActive; 5587 if (CC_UNLIKELY(!active)) { 5588 ALOGV("Not active and NO_MORE_BUFFERS"); 5589 return NO_MORE_BUFFERS; 5590 } 5591 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5592 if (result != NO_ERROR) { 5593 return NO_MORE_BUFFERS; 5594 } 5595 // read the server count again 5596 start_loop_here: 5597 framesAvail = cblk->framesAvailable_l(); 5598 } 5599 } 5600 5601// if (framesAvail < framesReq) { 5602// return NO_MORE_BUFFERS; 5603// } 5604 5605 if (framesReq > framesAvail) { 5606 framesReq = framesAvail; 5607 } 5608 5609 uint32_t u = cblk->user; 5610 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5611 5612 if (framesReq > bufferEnd - u) { 5613 framesReq = bufferEnd - u; 5614 } 5615 5616 buffer->frameCount = framesReq; 5617 buffer->raw = (void *)cblk->buffer(u); 5618 return NO_ERROR; 5619} 5620 5621 5622void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5623{ 5624 size_t size = mBufferQueue.size(); 5625 5626 for (size_t i = 0; i < size; i++) { 5627 Buffer *pBuffer = mBufferQueue.itemAt(i); 5628 delete [] pBuffer->mBuffer; 5629 delete pBuffer; 5630 } 5631 mBufferQueue.clear(); 5632} 5633 5634// ---------------------------------------------------------------------------- 5635 5636AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5637 : RefBase(), 5638 mAudioFlinger(audioFlinger), 5639 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5640 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5641 mPid(pid), 5642 mTimedTrackCount(0) 5643{ 5644 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5645} 5646 5647// Client destructor must be called with AudioFlinger::mLock held 5648AudioFlinger::Client::~Client() 5649{ 5650 mAudioFlinger->removeClient_l(mPid); 5651} 5652 5653sp<MemoryDealer> AudioFlinger::Client::heap() const 5654{ 5655 return mMemoryDealer; 5656} 5657 5658// Reserve one of the limited slots for a timed audio track associated 5659// with this client 5660bool AudioFlinger::Client::reserveTimedTrack() 5661{ 5662 const int kMaxTimedTracksPerClient = 4; 5663 5664 Mutex::Autolock _l(mTimedTrackLock); 5665 5666 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5667 ALOGW("can not create timed track - pid %d has exceeded the limit", 5668 mPid); 5669 return false; 5670 } 5671 5672 mTimedTrackCount++; 5673 return true; 5674} 5675 5676// Release a slot for a timed audio track 5677void AudioFlinger::Client::releaseTimedTrack() 5678{ 5679 Mutex::Autolock _l(mTimedTrackLock); 5680 mTimedTrackCount--; 5681} 5682 5683// ---------------------------------------------------------------------------- 5684 5685AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5686 const sp<IAudioFlingerClient>& client, 5687 pid_t pid) 5688 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5689{ 5690} 5691 5692AudioFlinger::NotificationClient::~NotificationClient() 5693{ 5694} 5695 5696void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5697{ 5698 sp<NotificationClient> keep(this); 5699 mAudioFlinger->removeNotificationClient(mPid); 5700} 5701 5702// ---------------------------------------------------------------------------- 5703 5704AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5705 : BnAudioTrack(), 5706 mTrack(track) 5707{ 5708} 5709 5710AudioFlinger::TrackHandle::~TrackHandle() { 5711 // just stop the track on deletion, associated resources 5712 // will be freed from the main thread once all pending buffers have 5713 // been played. Unless it's not in the active track list, in which 5714 // case we free everything now... 5715 mTrack->destroy(); 5716} 5717 5718sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5719 return mTrack->getCblk(); 5720} 5721 5722status_t AudioFlinger::TrackHandle::start() { 5723 return mTrack->start(); 5724} 5725 5726void AudioFlinger::TrackHandle::stop() { 5727 mTrack->stop(); 5728} 5729 5730void AudioFlinger::TrackHandle::flush() { 5731 mTrack->flush(); 5732} 5733 5734void AudioFlinger::TrackHandle::mute(bool e) { 5735 mTrack->mute(e); 5736} 5737 5738void AudioFlinger::TrackHandle::pause() { 5739 mTrack->pause(); 5740} 5741 5742status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5743{ 5744 return mTrack->attachAuxEffect(EffectId); 5745} 5746 5747status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5748 sp<IMemory>* buffer) { 5749 if (!mTrack->isTimedTrack()) 5750 return INVALID_OPERATION; 5751 5752 PlaybackThread::TimedTrack* tt = 5753 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5754 return tt->allocateTimedBuffer(size, buffer); 5755} 5756 5757status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5758 int64_t pts) { 5759 if (!mTrack->isTimedTrack()) 5760 return INVALID_OPERATION; 5761 5762 PlaybackThread::TimedTrack* tt = 5763 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5764 return tt->queueTimedBuffer(buffer, pts); 5765} 5766 5767status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5768 const LinearTransform& xform, int target) { 5769 5770 if (!mTrack->isTimedTrack()) 5771 return INVALID_OPERATION; 5772 5773 PlaybackThread::TimedTrack* tt = 5774 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5775 return tt->setMediaTimeTransform( 5776 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5777} 5778 5779status_t AudioFlinger::TrackHandle::onTransact( 5780 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5781{ 5782 return BnAudioTrack::onTransact(code, data, reply, flags); 5783} 5784 5785// ---------------------------------------------------------------------------- 5786 5787sp<IAudioRecord> AudioFlinger::openRecord( 5788 pid_t pid, 5789 audio_io_handle_t input, 5790 uint32_t sampleRate, 5791 audio_format_t format, 5792 audio_channel_mask_t channelMask, 5793 int frameCount, 5794 IAudioFlinger::track_flags_t flags, 5795 pid_t tid, 5796 int *sessionId, 5797 status_t *status) 5798{ 5799 sp<RecordThread::RecordTrack> recordTrack; 5800 sp<RecordHandle> recordHandle; 5801 sp<Client> client; 5802 status_t lStatus; 5803 RecordThread *thread; 5804 size_t inFrameCount; 5805 int lSessionId; 5806 5807 // check calling permissions 5808 if (!recordingAllowed()) { 5809 lStatus = PERMISSION_DENIED; 5810 goto Exit; 5811 } 5812 5813 // add client to list 5814 { // scope for mLock 5815 Mutex::Autolock _l(mLock); 5816 thread = checkRecordThread_l(input); 5817 if (thread == NULL) { 5818 lStatus = BAD_VALUE; 5819 goto Exit; 5820 } 5821 5822 client = registerPid_l(pid); 5823 5824 // If no audio session id is provided, create one here 5825 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5826 lSessionId = *sessionId; 5827 } else { 5828 lSessionId = nextUniqueId(); 5829 if (sessionId != NULL) { 5830 *sessionId = lSessionId; 5831 } 5832 } 5833 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5834 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5835 frameCount, lSessionId, flags, tid, &lStatus); 5836 } 5837 if (lStatus != NO_ERROR) { 5838 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5839 // destructor is called by the TrackBase destructor with mLock held 5840 client.clear(); 5841 recordTrack.clear(); 5842 goto Exit; 5843 } 5844 5845 // return to handle to client 5846 recordHandle = new RecordHandle(recordTrack); 5847 lStatus = NO_ERROR; 5848 5849Exit: 5850 if (status) { 5851 *status = lStatus; 5852 } 5853 return recordHandle; 5854} 5855 5856// ---------------------------------------------------------------------------- 5857 5858AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5859 : BnAudioRecord(), 5860 mRecordTrack(recordTrack) 5861{ 5862} 5863 5864AudioFlinger::RecordHandle::~RecordHandle() { 5865 stop_nonvirtual(); 5866} 5867 5868sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5869 return mRecordTrack->getCblk(); 5870} 5871 5872status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5873 ALOGV("RecordHandle::start()"); 5874 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5875} 5876 5877void AudioFlinger::RecordHandle::stop() { 5878 stop_nonvirtual(); 5879} 5880 5881void AudioFlinger::RecordHandle::stop_nonvirtual() { 5882 ALOGV("RecordHandle::stop()"); 5883 mRecordTrack->stop(); 5884} 5885 5886status_t AudioFlinger::RecordHandle::onTransact( 5887 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5888{ 5889 return BnAudioRecord::onTransact(code, data, reply, flags); 5890} 5891 5892// ---------------------------------------------------------------------------- 5893 5894AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5895 AudioStreamIn *input, 5896 uint32_t sampleRate, 5897 audio_channel_mask_t channelMask, 5898 audio_io_handle_t id, 5899 uint32_t device) : 5900 ThreadBase(audioFlinger, id, device, RECORD), 5901 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5902 // mRsmpInIndex and mInputBytes set by readInputParameters() 5903 mReqChannelCount(popcount(channelMask)), 5904 mReqSampleRate(sampleRate) 5905 // mBytesRead is only meaningful while active, and so is cleared in start() 5906 // (but might be better to also clear here for dump?) 5907{ 5908 snprintf(mName, kNameLength, "AudioIn_%X", id); 5909 5910 readInputParameters(); 5911} 5912 5913 5914AudioFlinger::RecordThread::~RecordThread() 5915{ 5916 delete[] mRsmpInBuffer; 5917 delete mResampler; 5918 delete[] mRsmpOutBuffer; 5919} 5920 5921void AudioFlinger::RecordThread::onFirstRef() 5922{ 5923 run(mName, PRIORITY_URGENT_AUDIO); 5924} 5925 5926status_t AudioFlinger::RecordThread::readyToRun() 5927{ 5928 status_t status = initCheck(); 5929 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5930 return status; 5931} 5932 5933bool AudioFlinger::RecordThread::threadLoop() 5934{ 5935 AudioBufferProvider::Buffer buffer; 5936 sp<RecordTrack> activeTrack; 5937 Vector< sp<EffectChain> > effectChains; 5938 5939 nsecs_t lastWarning = 0; 5940 5941 acquireWakeLock(); 5942 5943 // start recording 5944 while (!exitPending()) { 5945 5946 processConfigEvents(); 5947 5948 { // scope for mLock 5949 Mutex::Autolock _l(mLock); 5950 checkForNewParameters_l(); 5951 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5952 if (!mStandby) { 5953 mInput->stream->common.standby(&mInput->stream->common); 5954 mStandby = true; 5955 } 5956 5957 if (exitPending()) break; 5958 5959 releaseWakeLock_l(); 5960 ALOGV("RecordThread: loop stopping"); 5961 // go to sleep 5962 mWaitWorkCV.wait(mLock); 5963 ALOGV("RecordThread: loop starting"); 5964 acquireWakeLock_l(); 5965 continue; 5966 } 5967 if (mActiveTrack != 0) { 5968 if (mActiveTrack->mState == TrackBase::PAUSING) { 5969 if (!mStandby) { 5970 mInput->stream->common.standby(&mInput->stream->common); 5971 mStandby = true; 5972 } 5973 mActiveTrack.clear(); 5974 mStartStopCond.broadcast(); 5975 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5976 if (mReqChannelCount != mActiveTrack->channelCount()) { 5977 mActiveTrack.clear(); 5978 mStartStopCond.broadcast(); 5979 } else if (mBytesRead != 0) { 5980 // record start succeeds only if first read from audio input 5981 // succeeds 5982 if (mBytesRead > 0) { 5983 mActiveTrack->mState = TrackBase::ACTIVE; 5984 } else { 5985 mActiveTrack.clear(); 5986 } 5987 mStartStopCond.broadcast(); 5988 } 5989 mStandby = false; 5990 } 5991 } 5992 lockEffectChains_l(effectChains); 5993 } 5994 5995 if (mActiveTrack != 0) { 5996 if (mActiveTrack->mState != TrackBase::ACTIVE && 5997 mActiveTrack->mState != TrackBase::RESUMING) { 5998 unlockEffectChains(effectChains); 5999 usleep(kRecordThreadSleepUs); 6000 continue; 6001 } 6002 for (size_t i = 0; i < effectChains.size(); i ++) { 6003 effectChains[i]->process_l(); 6004 } 6005 6006 buffer.frameCount = mFrameCount; 6007 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6008 size_t framesOut = buffer.frameCount; 6009 if (mResampler == NULL) { 6010 // no resampling 6011 while (framesOut) { 6012 size_t framesIn = mFrameCount - mRsmpInIndex; 6013 if (framesIn) { 6014 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6015 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6016 if (framesIn > framesOut) 6017 framesIn = framesOut; 6018 mRsmpInIndex += framesIn; 6019 framesOut -= framesIn; 6020 if ((int)mChannelCount == mReqChannelCount || 6021 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6022 memcpy(dst, src, framesIn * mFrameSize); 6023 } else { 6024 int16_t *src16 = (int16_t *)src; 6025 int16_t *dst16 = (int16_t *)dst; 6026 if (mChannelCount == 1) { 6027 while (framesIn--) { 6028 *dst16++ = *src16; 6029 *dst16++ = *src16++; 6030 } 6031 } else { 6032 while (framesIn--) { 6033 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6034 src16 += 2; 6035 } 6036 } 6037 } 6038 } 6039 if (framesOut && mFrameCount == mRsmpInIndex) { 6040 if (framesOut == mFrameCount && 6041 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6042 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6043 framesOut = 0; 6044 } else { 6045 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6046 mRsmpInIndex = 0; 6047 } 6048 if (mBytesRead < 0) { 6049 ALOGE("Error reading audio input"); 6050 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6051 // Force input into standby so that it tries to 6052 // recover at next read attempt 6053 mInput->stream->common.standby(&mInput->stream->common); 6054 usleep(kRecordThreadSleepUs); 6055 } 6056 mRsmpInIndex = mFrameCount; 6057 framesOut = 0; 6058 buffer.frameCount = 0; 6059 } 6060 } 6061 } 6062 } else { 6063 // resampling 6064 6065 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6066 // alter output frame count as if we were expecting stereo samples 6067 if (mChannelCount == 1 && mReqChannelCount == 1) { 6068 framesOut >>= 1; 6069 } 6070 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6071 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6072 // are 32 bit aligned which should be always true. 6073 if (mChannelCount == 2 && mReqChannelCount == 1) { 6074 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6075 // the resampler always outputs stereo samples: do post stereo to mono conversion 6076 int16_t *src = (int16_t *)mRsmpOutBuffer; 6077 int16_t *dst = buffer.i16; 6078 while (framesOut--) { 6079 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6080 src += 2; 6081 } 6082 } else { 6083 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6084 } 6085 6086 } 6087 if (mFramestoDrop == 0) { 6088 mActiveTrack->releaseBuffer(&buffer); 6089 } else { 6090 if (mFramestoDrop > 0) { 6091 mFramestoDrop -= buffer.frameCount; 6092 if (mFramestoDrop <= 0) { 6093 clearSyncStartEvent(); 6094 } 6095 } else { 6096 mFramestoDrop += buffer.frameCount; 6097 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6098 mSyncStartEvent->isCancelled()) { 6099 ALOGW("Synced record %s, session %d, trigger session %d", 6100 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6101 mActiveTrack->sessionId(), 6102 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6103 clearSyncStartEvent(); 6104 } 6105 } 6106 } 6107 mActiveTrack->clearOverflow(); 6108 } 6109 // client isn't retrieving buffers fast enough 6110 else { 6111 if (!mActiveTrack->setOverflow()) { 6112 nsecs_t now = systemTime(); 6113 if ((now - lastWarning) > kWarningThrottleNs) { 6114 ALOGW("RecordThread: buffer overflow"); 6115 lastWarning = now; 6116 } 6117 } 6118 // Release the processor for a while before asking for a new buffer. 6119 // This will give the application more chance to read from the buffer and 6120 // clear the overflow. 6121 usleep(kRecordThreadSleepUs); 6122 } 6123 } 6124 // enable changes in effect chain 6125 unlockEffectChains(effectChains); 6126 effectChains.clear(); 6127 } 6128 6129 if (!mStandby) { 6130 mInput->stream->common.standby(&mInput->stream->common); 6131 } 6132 mActiveTrack.clear(); 6133 6134 mStartStopCond.broadcast(); 6135 6136 releaseWakeLock(); 6137 6138 ALOGV("RecordThread %p exiting", this); 6139 return false; 6140} 6141 6142 6143sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6144 const sp<AudioFlinger::Client>& client, 6145 uint32_t sampleRate, 6146 audio_format_t format, 6147 audio_channel_mask_t channelMask, 6148 int frameCount, 6149 int sessionId, 6150 IAudioFlinger::track_flags_t flags, 6151 pid_t tid, 6152 status_t *status) 6153{ 6154 sp<RecordTrack> track; 6155 status_t lStatus; 6156 6157 lStatus = initCheck(); 6158 if (lStatus != NO_ERROR) { 6159 ALOGE("Audio driver not initialized."); 6160 goto Exit; 6161 } 6162 6163 // FIXME use flags and tid similar to createTrack_l() 6164 6165 { // scope for mLock 6166 Mutex::Autolock _l(mLock); 6167 6168 track = new RecordTrack(this, client, sampleRate, 6169 format, channelMask, frameCount, sessionId); 6170 6171 if (track->getCblk() == 0) { 6172 lStatus = NO_MEMORY; 6173 goto Exit; 6174 } 6175 6176 mTrack = track.get(); 6177 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6178 bool suspend = audio_is_bluetooth_sco_device( 6179 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6180 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6181 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6182 } 6183 lStatus = NO_ERROR; 6184 6185Exit: 6186 if (status) { 6187 *status = lStatus; 6188 } 6189 return track; 6190} 6191 6192status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6193 AudioSystem::sync_event_t event, 6194 int triggerSession) 6195{ 6196 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6197 sp<ThreadBase> strongMe = this; 6198 status_t status = NO_ERROR; 6199 6200 if (event == AudioSystem::SYNC_EVENT_NONE) { 6201 clearSyncStartEvent(); 6202 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6203 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6204 triggerSession, 6205 recordTrack->sessionId(), 6206 syncStartEventCallback, 6207 this); 6208 // Sync event can be cancelled by the trigger session if the track is not in a 6209 // compatible state in which case we start record immediately 6210 if (mSyncStartEvent->isCancelled()) { 6211 clearSyncStartEvent(); 6212 } else { 6213 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6214 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6215 } 6216 } 6217 6218 { 6219 AutoMutex lock(mLock); 6220 if (mActiveTrack != 0) { 6221 if (recordTrack != mActiveTrack.get()) { 6222 status = -EBUSY; 6223 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6224 mActiveTrack->mState = TrackBase::ACTIVE; 6225 } 6226 return status; 6227 } 6228 6229 recordTrack->mState = TrackBase::IDLE; 6230 mActiveTrack = recordTrack; 6231 mLock.unlock(); 6232 status_t status = AudioSystem::startInput(mId); 6233 mLock.lock(); 6234 if (status != NO_ERROR) { 6235 mActiveTrack.clear(); 6236 clearSyncStartEvent(); 6237 return status; 6238 } 6239 mRsmpInIndex = mFrameCount; 6240 mBytesRead = 0; 6241 if (mResampler != NULL) { 6242 mResampler->reset(); 6243 } 6244 mActiveTrack->mState = TrackBase::RESUMING; 6245 // signal thread to start 6246 ALOGV("Signal record thread"); 6247 mWaitWorkCV.signal(); 6248 // do not wait for mStartStopCond if exiting 6249 if (exitPending()) { 6250 mActiveTrack.clear(); 6251 status = INVALID_OPERATION; 6252 goto startError; 6253 } 6254 mStartStopCond.wait(mLock); 6255 if (mActiveTrack == 0) { 6256 ALOGV("Record failed to start"); 6257 status = BAD_VALUE; 6258 goto startError; 6259 } 6260 ALOGV("Record started OK"); 6261 return status; 6262 } 6263startError: 6264 AudioSystem::stopInput(mId); 6265 clearSyncStartEvent(); 6266 return status; 6267} 6268 6269void AudioFlinger::RecordThread::clearSyncStartEvent() 6270{ 6271 if (mSyncStartEvent != 0) { 6272 mSyncStartEvent->cancel(); 6273 } 6274 mSyncStartEvent.clear(); 6275 mFramestoDrop = 0; 6276} 6277 6278void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6279{ 6280 sp<SyncEvent> strongEvent = event.promote(); 6281 6282 if (strongEvent != 0) { 6283 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6284 me->handleSyncStartEvent(strongEvent); 6285 } 6286} 6287 6288void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6289{ 6290 if (event == mSyncStartEvent) { 6291 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6292 // from audio HAL 6293 mFramestoDrop = mFrameCount * 2; 6294 } 6295} 6296 6297void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6298 ALOGV("RecordThread::stop"); 6299 sp<ThreadBase> strongMe = this; 6300 { 6301 AutoMutex lock(mLock); 6302 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6303 mActiveTrack->mState = TrackBase::PAUSING; 6304 // do not wait for mStartStopCond if exiting 6305 if (exitPending()) { 6306 return; 6307 } 6308 mStartStopCond.wait(mLock); 6309 // if we have been restarted, recordTrack == mActiveTrack.get() here 6310 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6311 mLock.unlock(); 6312 AudioSystem::stopInput(mId); 6313 mLock.lock(); 6314 ALOGV("Record stopped OK"); 6315 } 6316 } 6317 } 6318} 6319 6320bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6321{ 6322 return false; 6323} 6324 6325status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6326{ 6327 if (!isValidSyncEvent(event)) { 6328 return BAD_VALUE; 6329 } 6330 6331 Mutex::Autolock _l(mLock); 6332 6333 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6334 mTrack->setSyncEvent(event); 6335 return NO_ERROR; 6336 } 6337 return NAME_NOT_FOUND; 6338} 6339 6340status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6341{ 6342 const size_t SIZE = 256; 6343 char buffer[SIZE]; 6344 String8 result; 6345 6346 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6347 result.append(buffer); 6348 6349 if (mActiveTrack != 0) { 6350 result.append("Active Track:\n"); 6351 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6352 mActiveTrack->dump(buffer, SIZE); 6353 result.append(buffer); 6354 6355 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6356 result.append(buffer); 6357 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6358 result.append(buffer); 6359 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6360 result.append(buffer); 6361 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6362 result.append(buffer); 6363 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6364 result.append(buffer); 6365 6366 6367 } else { 6368 result.append("No record client\n"); 6369 } 6370 write(fd, result.string(), result.size()); 6371 6372 dumpBase(fd, args); 6373 dumpEffectChains(fd, args); 6374 6375 return NO_ERROR; 6376} 6377 6378// AudioBufferProvider interface 6379status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6380{ 6381 size_t framesReq = buffer->frameCount; 6382 size_t framesReady = mFrameCount - mRsmpInIndex; 6383 int channelCount; 6384 6385 if (framesReady == 0) { 6386 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6387 if (mBytesRead < 0) { 6388 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6389 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6390 // Force input into standby so that it tries to 6391 // recover at next read attempt 6392 mInput->stream->common.standby(&mInput->stream->common); 6393 usleep(kRecordThreadSleepUs); 6394 } 6395 buffer->raw = NULL; 6396 buffer->frameCount = 0; 6397 return NOT_ENOUGH_DATA; 6398 } 6399 mRsmpInIndex = 0; 6400 framesReady = mFrameCount; 6401 } 6402 6403 if (framesReq > framesReady) { 6404 framesReq = framesReady; 6405 } 6406 6407 if (mChannelCount == 1 && mReqChannelCount == 2) { 6408 channelCount = 1; 6409 } else { 6410 channelCount = 2; 6411 } 6412 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6413 buffer->frameCount = framesReq; 6414 return NO_ERROR; 6415} 6416 6417// AudioBufferProvider interface 6418void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6419{ 6420 mRsmpInIndex += buffer->frameCount; 6421 buffer->frameCount = 0; 6422} 6423 6424bool AudioFlinger::RecordThread::checkForNewParameters_l() 6425{ 6426 bool reconfig = false; 6427 6428 while (!mNewParameters.isEmpty()) { 6429 status_t status = NO_ERROR; 6430 String8 keyValuePair = mNewParameters[0]; 6431 AudioParameter param = AudioParameter(keyValuePair); 6432 int value; 6433 audio_format_t reqFormat = mFormat; 6434 int reqSamplingRate = mReqSampleRate; 6435 int reqChannelCount = mReqChannelCount; 6436 6437 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6438 reqSamplingRate = value; 6439 reconfig = true; 6440 } 6441 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6442 reqFormat = (audio_format_t) value; 6443 reconfig = true; 6444 } 6445 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6446 reqChannelCount = popcount(value); 6447 reconfig = true; 6448 } 6449 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6450 // do not accept frame count changes if tracks are open as the track buffer 6451 // size depends on frame count and correct behavior would not be guaranteed 6452 // if frame count is changed after track creation 6453 if (mActiveTrack != 0) { 6454 status = INVALID_OPERATION; 6455 } else { 6456 reconfig = true; 6457 } 6458 } 6459 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6460 // forward device change to effects that have requested to be 6461 // aware of attached audio device. 6462 for (size_t i = 0; i < mEffectChains.size(); i++) { 6463 mEffectChains[i]->setDevice_l(value); 6464 } 6465 // store input device and output device but do not forward output device to audio HAL. 6466 // Note that status is ignored by the caller for output device 6467 // (see AudioFlinger::setParameters() 6468 uint32_t /*audio_devices_t*/ newDevice = mDevice; 6469 if (value & AUDIO_DEVICE_OUT_ALL) { 6470 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6471 status = BAD_VALUE; 6472 } else { 6473 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6474 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6475 if (mTrack != NULL) { 6476 bool suspend = audio_is_bluetooth_sco_device( 6477 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6478 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6479 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6480 } 6481 } 6482 newDevice |= value; 6483 mDevice = (audio_devices_t) newDevice; // since mDevice is read by other threads, only write to it once 6484 } 6485 if (status == NO_ERROR) { 6486 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6487 if (status == INVALID_OPERATION) { 6488 mInput->stream->common.standby(&mInput->stream->common); 6489 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6490 keyValuePair.string()); 6491 } 6492 if (reconfig) { 6493 if (status == BAD_VALUE && 6494 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6495 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6496 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6497 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6498 (reqChannelCount <= FCC_2)) { 6499 status = NO_ERROR; 6500 } 6501 if (status == NO_ERROR) { 6502 readInputParameters(); 6503 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6504 } 6505 } 6506 } 6507 6508 mNewParameters.removeAt(0); 6509 6510 mParamStatus = status; 6511 mParamCond.signal(); 6512 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6513 // already timed out waiting for the status and will never signal the condition. 6514 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6515 } 6516 return reconfig; 6517} 6518 6519String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6520{ 6521 char *s; 6522 String8 out_s8 = String8(); 6523 6524 Mutex::Autolock _l(mLock); 6525 if (initCheck() != NO_ERROR) { 6526 return out_s8; 6527 } 6528 6529 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6530 out_s8 = String8(s); 6531 free(s); 6532 return out_s8; 6533} 6534 6535void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6536 AudioSystem::OutputDescriptor desc; 6537 void *param2 = NULL; 6538 6539 switch (event) { 6540 case AudioSystem::INPUT_OPENED: 6541 case AudioSystem::INPUT_CONFIG_CHANGED: 6542 desc.channels = mChannelMask; 6543 desc.samplingRate = mSampleRate; 6544 desc.format = mFormat; 6545 desc.frameCount = mFrameCount; 6546 desc.latency = 0; 6547 param2 = &desc; 6548 break; 6549 6550 case AudioSystem::INPUT_CLOSED: 6551 default: 6552 break; 6553 } 6554 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6555} 6556 6557void AudioFlinger::RecordThread::readInputParameters() 6558{ 6559 delete mRsmpInBuffer; 6560 // mRsmpInBuffer is always assigned a new[] below 6561 delete mRsmpOutBuffer; 6562 mRsmpOutBuffer = NULL; 6563 delete mResampler; 6564 mResampler = NULL; 6565 6566 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6567 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6568 mChannelCount = (uint16_t)popcount(mChannelMask); 6569 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6570 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6571 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6572 mFrameCount = mInputBytes / mFrameSize; 6573 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6574 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6575 6576 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6577 { 6578 int channelCount; 6579 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6580 // stereo to mono post process as the resampler always outputs stereo. 6581 if (mChannelCount == 1 && mReqChannelCount == 2) { 6582 channelCount = 1; 6583 } else { 6584 channelCount = 2; 6585 } 6586 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6587 mResampler->setSampleRate(mSampleRate); 6588 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6589 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6590 6591 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6592 if (mChannelCount == 1 && mReqChannelCount == 1) { 6593 mFrameCount >>= 1; 6594 } 6595 6596 } 6597 mRsmpInIndex = mFrameCount; 6598} 6599 6600unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6601{ 6602 Mutex::Autolock _l(mLock); 6603 if (initCheck() != NO_ERROR) { 6604 return 0; 6605 } 6606 6607 return mInput->stream->get_input_frames_lost(mInput->stream); 6608} 6609 6610uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6611{ 6612 Mutex::Autolock _l(mLock); 6613 uint32_t result = 0; 6614 if (getEffectChain_l(sessionId) != 0) { 6615 result = EFFECT_SESSION; 6616 } 6617 6618 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6619 result |= TRACK_SESSION; 6620 } 6621 6622 return result; 6623} 6624 6625AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6626{ 6627 Mutex::Autolock _l(mLock); 6628 return mTrack; 6629} 6630 6631AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6632{ 6633 Mutex::Autolock _l(mLock); 6634 AudioStreamIn *input = mInput; 6635 mInput = NULL; 6636 return input; 6637} 6638 6639// this method must always be called either with ThreadBase mLock held or inside the thread loop 6640audio_stream_t* AudioFlinger::RecordThread::stream() const 6641{ 6642 if (mInput == NULL) { 6643 return NULL; 6644 } 6645 return &mInput->stream->common; 6646} 6647 6648 6649// ---------------------------------------------------------------------------- 6650 6651audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6652{ 6653 if (!settingsAllowed()) { 6654 return 0; 6655 } 6656 Mutex::Autolock _l(mLock); 6657 return loadHwModule_l(name); 6658} 6659 6660// loadHwModule_l() must be called with AudioFlinger::mLock held 6661audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6662{ 6663 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6664 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6665 ALOGW("loadHwModule() module %s already loaded", name); 6666 return mAudioHwDevs.keyAt(i); 6667 } 6668 } 6669 6670 audio_hw_device_t *dev; 6671 6672 int rc = load_audio_interface(name, &dev); 6673 if (rc) { 6674 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6675 return 0; 6676 } 6677 6678 mHardwareStatus = AUDIO_HW_INIT; 6679 rc = dev->init_check(dev); 6680 mHardwareStatus = AUDIO_HW_IDLE; 6681 if (rc) { 6682 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6683 return 0; 6684 } 6685 6686 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6687 (NULL != dev->set_master_volume)) { 6688 AutoMutex lock(mHardwareLock); 6689 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6690 dev->set_master_volume(dev, mMasterVolume); 6691 mHardwareStatus = AUDIO_HW_IDLE; 6692 } 6693 6694 audio_module_handle_t handle = nextUniqueId(); 6695 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6696 6697 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6698 name, dev->common.module->name, dev->common.module->id, handle); 6699 6700 return handle; 6701 6702} 6703 6704audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6705 audio_devices_t *pDevices, 6706 uint32_t *pSamplingRate, 6707 audio_format_t *pFormat, 6708 audio_channel_mask_t *pChannelMask, 6709 uint32_t *pLatencyMs, 6710 audio_output_flags_t flags) 6711{ 6712 status_t status; 6713 PlaybackThread *thread = NULL; 6714 struct audio_config config = { 6715 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6716 channel_mask: pChannelMask ? *pChannelMask : 0, 6717 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6718 }; 6719 audio_stream_out_t *outStream = NULL; 6720 audio_hw_device_t *outHwDev; 6721 6722 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6723 module, 6724 (pDevices != NULL) ? (int)*pDevices : 0, 6725 config.sample_rate, 6726 config.format, 6727 config.channel_mask, 6728 flags); 6729 6730 if (pDevices == NULL || *pDevices == 0) { 6731 return 0; 6732 } 6733 6734 Mutex::Autolock _l(mLock); 6735 6736 outHwDev = findSuitableHwDev_l(module, *pDevices); 6737 if (outHwDev == NULL) 6738 return 0; 6739 6740 audio_io_handle_t id = nextUniqueId(); 6741 6742 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6743 6744 status = outHwDev->open_output_stream(outHwDev, 6745 id, 6746 *pDevices, 6747 (audio_output_flags_t)flags, 6748 &config, 6749 &outStream); 6750 6751 mHardwareStatus = AUDIO_HW_IDLE; 6752 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6753 outStream, 6754 config.sample_rate, 6755 config.format, 6756 config.channel_mask, 6757 status); 6758 6759 if (status == NO_ERROR && outStream != NULL) { 6760 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6761 6762 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6763 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6764 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6765 thread = new DirectOutputThread(this, output, id, *pDevices); 6766 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6767 } else { 6768 thread = new MixerThread(this, output, id, *pDevices); 6769 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6770 } 6771 mPlaybackThreads.add(id, thread); 6772 6773 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6774 if (pFormat != NULL) *pFormat = config.format; 6775 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6776 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6777 6778 // notify client processes of the new output creation 6779 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6780 6781 // the first primary output opened designates the primary hw device 6782 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6783 ALOGI("Using module %d has the primary audio interface", module); 6784 mPrimaryHardwareDev = outHwDev; 6785 6786 AutoMutex lock(mHardwareLock); 6787 mHardwareStatus = AUDIO_HW_SET_MODE; 6788 outHwDev->set_mode(outHwDev, mMode); 6789 6790 // Determine the level of master volume support the primary audio HAL has, 6791 // and set the initial master volume at the same time. 6792 float initialVolume = 1.0; 6793 mMasterVolumeSupportLvl = MVS_NONE; 6794 6795 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6796 if ((NULL != outHwDev->get_master_volume) && 6797 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6798 mMasterVolumeSupportLvl = MVS_FULL; 6799 } else { 6800 mMasterVolumeSupportLvl = MVS_SETONLY; 6801 initialVolume = 1.0; 6802 } 6803 6804 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6805 if ((NULL == outHwDev->set_master_volume) || 6806 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6807 mMasterVolumeSupportLvl = MVS_NONE; 6808 } 6809 // now that we have a primary device, initialize master volume on other devices 6810 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6811 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6812 6813 if ((dev != mPrimaryHardwareDev) && 6814 (NULL != dev->set_master_volume)) { 6815 dev->set_master_volume(dev, initialVolume); 6816 } 6817 } 6818 mHardwareStatus = AUDIO_HW_IDLE; 6819 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6820 ? initialVolume 6821 : 1.0; 6822 mMasterVolume = initialVolume; 6823 } 6824 return id; 6825 } 6826 6827 return 0; 6828} 6829 6830audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6831 audio_io_handle_t output2) 6832{ 6833 Mutex::Autolock _l(mLock); 6834 MixerThread *thread1 = checkMixerThread_l(output1); 6835 MixerThread *thread2 = checkMixerThread_l(output2); 6836 6837 if (thread1 == NULL || thread2 == NULL) { 6838 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6839 return 0; 6840 } 6841 6842 audio_io_handle_t id = nextUniqueId(); 6843 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6844 thread->addOutputTrack(thread2); 6845 mPlaybackThreads.add(id, thread); 6846 // notify client processes of the new output creation 6847 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6848 return id; 6849} 6850 6851status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6852{ 6853 return closeOutput_nonvirtual(output); 6854} 6855 6856status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 6857{ 6858 // keep strong reference on the playback thread so that 6859 // it is not destroyed while exit() is executed 6860 sp<PlaybackThread> thread; 6861 { 6862 Mutex::Autolock _l(mLock); 6863 thread = checkPlaybackThread_l(output); 6864 if (thread == NULL) { 6865 return BAD_VALUE; 6866 } 6867 6868 ALOGV("closeOutput() %d", output); 6869 6870 if (thread->type() == ThreadBase::MIXER) { 6871 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6872 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6873 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6874 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6875 } 6876 } 6877 } 6878 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6879 mPlaybackThreads.removeItem(output); 6880 } 6881 thread->exit(); 6882 // The thread entity (active unit of execution) is no longer running here, 6883 // but the ThreadBase container still exists. 6884 6885 if (thread->type() != ThreadBase::DUPLICATING) { 6886 AudioStreamOut *out = thread->clearOutput(); 6887 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6888 // from now on thread->mOutput is NULL 6889 out->hwDev->close_output_stream(out->hwDev, out->stream); 6890 delete out; 6891 } 6892 return NO_ERROR; 6893} 6894 6895status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6896{ 6897 Mutex::Autolock _l(mLock); 6898 PlaybackThread *thread = checkPlaybackThread_l(output); 6899 6900 if (thread == NULL) { 6901 return BAD_VALUE; 6902 } 6903 6904 ALOGV("suspendOutput() %d", output); 6905 thread->suspend(); 6906 6907 return NO_ERROR; 6908} 6909 6910status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6911{ 6912 Mutex::Autolock _l(mLock); 6913 PlaybackThread *thread = checkPlaybackThread_l(output); 6914 6915 if (thread == NULL) { 6916 return BAD_VALUE; 6917 } 6918 6919 ALOGV("restoreOutput() %d", output); 6920 6921 thread->restore(); 6922 6923 return NO_ERROR; 6924} 6925 6926audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6927 audio_devices_t *pDevices, 6928 uint32_t *pSamplingRate, 6929 audio_format_t *pFormat, 6930 audio_channel_mask_t *pChannelMask) 6931{ 6932 status_t status; 6933 RecordThread *thread = NULL; 6934 struct audio_config config = { 6935 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6936 channel_mask: pChannelMask ? *pChannelMask : 0, 6937 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6938 }; 6939 uint32_t reqSamplingRate = config.sample_rate; 6940 audio_format_t reqFormat = config.format; 6941 audio_channel_mask_t reqChannels = config.channel_mask; 6942 audio_stream_in_t *inStream = NULL; 6943 audio_hw_device_t *inHwDev; 6944 6945 if (pDevices == NULL || *pDevices == 0) { 6946 return 0; 6947 } 6948 6949 Mutex::Autolock _l(mLock); 6950 6951 inHwDev = findSuitableHwDev_l(module, *pDevices); 6952 if (inHwDev == NULL) 6953 return 0; 6954 6955 audio_io_handle_t id = nextUniqueId(); 6956 6957 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6958 &inStream); 6959 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6960 inStream, 6961 config.sample_rate, 6962 config.format, 6963 config.channel_mask, 6964 status); 6965 6966 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6967 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6968 // or stereo to mono conversions on 16 bit PCM inputs. 6969 if (status == BAD_VALUE && 6970 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6971 (config.sample_rate <= 2 * reqSamplingRate) && 6972 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6973 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 6974 inStream = NULL; 6975 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6976 } 6977 6978 if (status == NO_ERROR && inStream != NULL) { 6979 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6980 6981 // Start record thread 6982 // RecorThread require both input and output device indication to forward to audio 6983 // pre processing modules 6984 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6985 thread = new RecordThread(this, 6986 input, 6987 reqSamplingRate, 6988 reqChannels, 6989 id, 6990 device); 6991 mRecordThreads.add(id, thread); 6992 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6993 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6994 if (pFormat != NULL) *pFormat = config.format; 6995 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6996 6997 input->stream->common.standby(&input->stream->common); 6998 6999 // notify client processes of the new input creation 7000 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7001 return id; 7002 } 7003 7004 return 0; 7005} 7006 7007status_t AudioFlinger::closeInput(audio_io_handle_t input) 7008{ 7009 return closeInput_nonvirtual(input); 7010} 7011 7012status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7013{ 7014 // keep strong reference on the record thread so that 7015 // it is not destroyed while exit() is executed 7016 sp<RecordThread> thread; 7017 { 7018 Mutex::Autolock _l(mLock); 7019 thread = checkRecordThread_l(input); 7020 if (thread == 0) { 7021 return BAD_VALUE; 7022 } 7023 7024 ALOGV("closeInput() %d", input); 7025 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7026 mRecordThreads.removeItem(input); 7027 } 7028 thread->exit(); 7029 // The thread entity (active unit of execution) is no longer running here, 7030 // but the ThreadBase container still exists. 7031 7032 AudioStreamIn *in = thread->clearInput(); 7033 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7034 // from now on thread->mInput is NULL 7035 in->hwDev->close_input_stream(in->hwDev, in->stream); 7036 delete in; 7037 7038 return NO_ERROR; 7039} 7040 7041status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7042{ 7043 Mutex::Autolock _l(mLock); 7044 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7045 7046 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7047 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7048 thread->invalidateTracks(stream); 7049 } 7050 7051 return NO_ERROR; 7052} 7053 7054 7055int AudioFlinger::newAudioSessionId() 7056{ 7057 return nextUniqueId(); 7058} 7059 7060void AudioFlinger::acquireAudioSessionId(int audioSession) 7061{ 7062 Mutex::Autolock _l(mLock); 7063 pid_t caller = IPCThreadState::self()->getCallingPid(); 7064 ALOGV("acquiring %d from %d", audioSession, caller); 7065 size_t num = mAudioSessionRefs.size(); 7066 for (size_t i = 0; i< num; i++) { 7067 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7068 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7069 ref->mCnt++; 7070 ALOGV(" incremented refcount to %d", ref->mCnt); 7071 return; 7072 } 7073 } 7074 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7075 ALOGV(" added new entry for %d", audioSession); 7076} 7077 7078void AudioFlinger::releaseAudioSessionId(int audioSession) 7079{ 7080 Mutex::Autolock _l(mLock); 7081 pid_t caller = IPCThreadState::self()->getCallingPid(); 7082 ALOGV("releasing %d from %d", audioSession, caller); 7083 size_t num = mAudioSessionRefs.size(); 7084 for (size_t i = 0; i< num; i++) { 7085 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7086 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7087 ref->mCnt--; 7088 ALOGV(" decremented refcount to %d", ref->mCnt); 7089 if (ref->mCnt == 0) { 7090 mAudioSessionRefs.removeAt(i); 7091 delete ref; 7092 purgeStaleEffects_l(); 7093 } 7094 return; 7095 } 7096 } 7097 ALOGW("session id %d not found for pid %d", audioSession, caller); 7098} 7099 7100void AudioFlinger::purgeStaleEffects_l() { 7101 7102 ALOGV("purging stale effects"); 7103 7104 Vector< sp<EffectChain> > chains; 7105 7106 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7107 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7108 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7109 sp<EffectChain> ec = t->mEffectChains[j]; 7110 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7111 chains.push(ec); 7112 } 7113 } 7114 } 7115 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7116 sp<RecordThread> t = mRecordThreads.valueAt(i); 7117 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7118 sp<EffectChain> ec = t->mEffectChains[j]; 7119 chains.push(ec); 7120 } 7121 } 7122 7123 for (size_t i = 0; i < chains.size(); i++) { 7124 sp<EffectChain> ec = chains[i]; 7125 int sessionid = ec->sessionId(); 7126 sp<ThreadBase> t = ec->mThread.promote(); 7127 if (t == 0) { 7128 continue; 7129 } 7130 size_t numsessionrefs = mAudioSessionRefs.size(); 7131 bool found = false; 7132 for (size_t k = 0; k < numsessionrefs; k++) { 7133 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7134 if (ref->mSessionid == sessionid) { 7135 ALOGV(" session %d still exists for %d with %d refs", 7136 sessionid, ref->mPid, ref->mCnt); 7137 found = true; 7138 break; 7139 } 7140 } 7141 if (!found) { 7142 Mutex::Autolock _l (t->mLock); 7143 // remove all effects from the chain 7144 while (ec->mEffects.size()) { 7145 sp<EffectModule> effect = ec->mEffects[0]; 7146 effect->unPin(); 7147 t->removeEffect_l(effect); 7148 if (effect->purgeHandles()) { 7149 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7150 } 7151 AudioSystem::unregisterEffect(effect->id()); 7152 } 7153 } 7154 } 7155 return; 7156} 7157 7158// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7159AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7160{ 7161 return mPlaybackThreads.valueFor(output).get(); 7162} 7163 7164// checkMixerThread_l() must be called with AudioFlinger::mLock held 7165AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7166{ 7167 PlaybackThread *thread = checkPlaybackThread_l(output); 7168 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7169} 7170 7171// checkRecordThread_l() must be called with AudioFlinger::mLock held 7172AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7173{ 7174 return mRecordThreads.valueFor(input).get(); 7175} 7176 7177uint32_t AudioFlinger::nextUniqueId() 7178{ 7179 return android_atomic_inc(&mNextUniqueId); 7180} 7181 7182AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7183{ 7184 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7185 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7186 AudioStreamOut *output = thread->getOutput(); 7187 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7188 return thread; 7189 } 7190 } 7191 return NULL; 7192} 7193 7194uint32_t AudioFlinger::primaryOutputDevice_l() const 7195{ 7196 PlaybackThread *thread = primaryPlaybackThread_l(); 7197 7198 if (thread == NULL) { 7199 return 0; 7200 } 7201 7202 return thread->device(); 7203} 7204 7205sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7206 int triggerSession, 7207 int listenerSession, 7208 sync_event_callback_t callBack, 7209 void *cookie) 7210{ 7211 Mutex::Autolock _l(mLock); 7212 7213 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7214 status_t playStatus = NAME_NOT_FOUND; 7215 status_t recStatus = NAME_NOT_FOUND; 7216 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7217 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7218 if (playStatus == NO_ERROR) { 7219 return event; 7220 } 7221 } 7222 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7223 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7224 if (recStatus == NO_ERROR) { 7225 return event; 7226 } 7227 } 7228 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7229 mPendingSyncEvents.add(event); 7230 } else { 7231 ALOGV("createSyncEvent() invalid event %d", event->type()); 7232 event.clear(); 7233 } 7234 return event; 7235} 7236 7237// ---------------------------------------------------------------------------- 7238// Effect management 7239// ---------------------------------------------------------------------------- 7240 7241 7242status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7243{ 7244 Mutex::Autolock _l(mLock); 7245 return EffectQueryNumberEffects(numEffects); 7246} 7247 7248status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7249{ 7250 Mutex::Autolock _l(mLock); 7251 return EffectQueryEffect(index, descriptor); 7252} 7253 7254status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7255 effect_descriptor_t *descriptor) const 7256{ 7257 Mutex::Autolock _l(mLock); 7258 return EffectGetDescriptor(pUuid, descriptor); 7259} 7260 7261 7262sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7263 effect_descriptor_t *pDesc, 7264 const sp<IEffectClient>& effectClient, 7265 int32_t priority, 7266 audio_io_handle_t io, 7267 int sessionId, 7268 status_t *status, 7269 int *id, 7270 int *enabled) 7271{ 7272 status_t lStatus = NO_ERROR; 7273 sp<EffectHandle> handle; 7274 effect_descriptor_t desc; 7275 7276 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7277 pid, effectClient.get(), priority, sessionId, io); 7278 7279 if (pDesc == NULL) { 7280 lStatus = BAD_VALUE; 7281 goto Exit; 7282 } 7283 7284 // check audio settings permission for global effects 7285 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7286 lStatus = PERMISSION_DENIED; 7287 goto Exit; 7288 } 7289 7290 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7291 // that can only be created by audio policy manager (running in same process) 7292 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7293 lStatus = PERMISSION_DENIED; 7294 goto Exit; 7295 } 7296 7297 if (io == 0) { 7298 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7299 // output must be specified by AudioPolicyManager when using session 7300 // AUDIO_SESSION_OUTPUT_STAGE 7301 lStatus = BAD_VALUE; 7302 goto Exit; 7303 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7304 // if the output returned by getOutputForEffect() is removed before we lock the 7305 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7306 // and we will exit safely 7307 io = AudioSystem::getOutputForEffect(&desc); 7308 } 7309 } 7310 7311 { 7312 Mutex::Autolock _l(mLock); 7313 7314 7315 if (!EffectIsNullUuid(&pDesc->uuid)) { 7316 // if uuid is specified, request effect descriptor 7317 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7318 if (lStatus < 0) { 7319 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7320 goto Exit; 7321 } 7322 } else { 7323 // if uuid is not specified, look for an available implementation 7324 // of the required type in effect factory 7325 if (EffectIsNullUuid(&pDesc->type)) { 7326 ALOGW("createEffect() no effect type"); 7327 lStatus = BAD_VALUE; 7328 goto Exit; 7329 } 7330 uint32_t numEffects = 0; 7331 effect_descriptor_t d; 7332 d.flags = 0; // prevent compiler warning 7333 bool found = false; 7334 7335 lStatus = EffectQueryNumberEffects(&numEffects); 7336 if (lStatus < 0) { 7337 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7338 goto Exit; 7339 } 7340 for (uint32_t i = 0; i < numEffects; i++) { 7341 lStatus = EffectQueryEffect(i, &desc); 7342 if (lStatus < 0) { 7343 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7344 continue; 7345 } 7346 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7347 // If matching type found save effect descriptor. If the session is 7348 // 0 and the effect is not auxiliary, continue enumeration in case 7349 // an auxiliary version of this effect type is available 7350 found = true; 7351 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7352 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7353 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7354 break; 7355 } 7356 } 7357 } 7358 if (!found) { 7359 lStatus = BAD_VALUE; 7360 ALOGW("createEffect() effect not found"); 7361 goto Exit; 7362 } 7363 // For same effect type, chose auxiliary version over insert version if 7364 // connect to output mix (Compliance to OpenSL ES) 7365 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7366 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7367 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7368 } 7369 } 7370 7371 // Do not allow auxiliary effects on a session different from 0 (output mix) 7372 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7373 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7374 lStatus = INVALID_OPERATION; 7375 goto Exit; 7376 } 7377 7378 // check recording permission for visualizer 7379 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7380 !recordingAllowed()) { 7381 lStatus = PERMISSION_DENIED; 7382 goto Exit; 7383 } 7384 7385 // return effect descriptor 7386 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7387 7388 // If output is not specified try to find a matching audio session ID in one of the 7389 // output threads. 7390 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7391 // because of code checking output when entering the function. 7392 // Note: io is never 0 when creating an effect on an input 7393 if (io == 0) { 7394 // look for the thread where the specified audio session is present 7395 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7396 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7397 io = mPlaybackThreads.keyAt(i); 7398 break; 7399 } 7400 } 7401 if (io == 0) { 7402 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7403 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7404 io = mRecordThreads.keyAt(i); 7405 break; 7406 } 7407 } 7408 } 7409 // If no output thread contains the requested session ID, default to 7410 // first output. The effect chain will be moved to the correct output 7411 // thread when a track with the same session ID is created 7412 if (io == 0 && mPlaybackThreads.size()) { 7413 io = mPlaybackThreads.keyAt(0); 7414 } 7415 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7416 } 7417 ThreadBase *thread = checkRecordThread_l(io); 7418 if (thread == NULL) { 7419 thread = checkPlaybackThread_l(io); 7420 if (thread == NULL) { 7421 ALOGE("createEffect() unknown output thread"); 7422 lStatus = BAD_VALUE; 7423 goto Exit; 7424 } 7425 } 7426 7427 sp<Client> client = registerPid_l(pid); 7428 7429 // create effect on selected output thread 7430 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7431 &desc, enabled, &lStatus); 7432 if (handle != 0 && id != NULL) { 7433 *id = handle->id(); 7434 } 7435 } 7436 7437Exit: 7438 if (status != NULL) { 7439 *status = lStatus; 7440 } 7441 return handle; 7442} 7443 7444status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7445 audio_io_handle_t dstOutput) 7446{ 7447 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7448 sessionId, srcOutput, dstOutput); 7449 Mutex::Autolock _l(mLock); 7450 if (srcOutput == dstOutput) { 7451 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7452 return NO_ERROR; 7453 } 7454 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7455 if (srcThread == NULL) { 7456 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7457 return BAD_VALUE; 7458 } 7459 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7460 if (dstThread == NULL) { 7461 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7462 return BAD_VALUE; 7463 } 7464 7465 Mutex::Autolock _dl(dstThread->mLock); 7466 Mutex::Autolock _sl(srcThread->mLock); 7467 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7468 7469 return NO_ERROR; 7470} 7471 7472// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7473status_t AudioFlinger::moveEffectChain_l(int sessionId, 7474 AudioFlinger::PlaybackThread *srcThread, 7475 AudioFlinger::PlaybackThread *dstThread, 7476 bool reRegister) 7477{ 7478 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7479 sessionId, srcThread, dstThread); 7480 7481 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7482 if (chain == 0) { 7483 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7484 sessionId, srcThread); 7485 return INVALID_OPERATION; 7486 } 7487 7488 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7489 // so that a new chain is created with correct parameters when first effect is added. This is 7490 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7491 // removed. 7492 srcThread->removeEffectChain_l(chain); 7493 7494 // transfer all effects one by one so that new effect chain is created on new thread with 7495 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7496 audio_io_handle_t dstOutput = dstThread->id(); 7497 sp<EffectChain> dstChain; 7498 uint32_t strategy = 0; // prevent compiler warning 7499 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7500 while (effect != 0) { 7501 srcThread->removeEffect_l(effect); 7502 dstThread->addEffect_l(effect); 7503 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7504 if (effect->state() == EffectModule::ACTIVE || 7505 effect->state() == EffectModule::STOPPING) { 7506 effect->start(); 7507 } 7508 // if the move request is not received from audio policy manager, the effect must be 7509 // re-registered with the new strategy and output 7510 if (dstChain == 0) { 7511 dstChain = effect->chain().promote(); 7512 if (dstChain == 0) { 7513 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7514 srcThread->addEffect_l(effect); 7515 return NO_INIT; 7516 } 7517 strategy = dstChain->strategy(); 7518 } 7519 if (reRegister) { 7520 AudioSystem::unregisterEffect(effect->id()); 7521 AudioSystem::registerEffect(&effect->desc(), 7522 dstOutput, 7523 strategy, 7524 sessionId, 7525 effect->id()); 7526 } 7527 effect = chain->getEffectFromId_l(0); 7528 } 7529 7530 return NO_ERROR; 7531} 7532 7533 7534// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7535sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7536 const sp<AudioFlinger::Client>& client, 7537 const sp<IEffectClient>& effectClient, 7538 int32_t priority, 7539 int sessionId, 7540 effect_descriptor_t *desc, 7541 int *enabled, 7542 status_t *status 7543 ) 7544{ 7545 sp<EffectModule> effect; 7546 sp<EffectHandle> handle; 7547 status_t lStatus; 7548 sp<EffectChain> chain; 7549 bool chainCreated = false; 7550 bool effectCreated = false; 7551 bool effectRegistered = false; 7552 7553 lStatus = initCheck(); 7554 if (lStatus != NO_ERROR) { 7555 ALOGW("createEffect_l() Audio driver not initialized."); 7556 goto Exit; 7557 } 7558 7559 // Do not allow effects with session ID 0 on direct output or duplicating threads 7560 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7561 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7562 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7563 desc->name, sessionId); 7564 lStatus = BAD_VALUE; 7565 goto Exit; 7566 } 7567 // Only Pre processor effects are allowed on input threads and only on input threads 7568 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7569 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7570 desc->name, desc->flags, mType); 7571 lStatus = BAD_VALUE; 7572 goto Exit; 7573 } 7574 7575 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7576 7577 { // scope for mLock 7578 Mutex::Autolock _l(mLock); 7579 7580 // check for existing effect chain with the requested audio session 7581 chain = getEffectChain_l(sessionId); 7582 if (chain == 0) { 7583 // create a new chain for this session 7584 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7585 chain = new EffectChain(this, sessionId); 7586 addEffectChain_l(chain); 7587 chain->setStrategy(getStrategyForSession_l(sessionId)); 7588 chainCreated = true; 7589 } else { 7590 effect = chain->getEffectFromDesc_l(desc); 7591 } 7592 7593 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7594 7595 if (effect == 0) { 7596 int id = mAudioFlinger->nextUniqueId(); 7597 // Check CPU and memory usage 7598 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7599 if (lStatus != NO_ERROR) { 7600 goto Exit; 7601 } 7602 effectRegistered = true; 7603 // create a new effect module if none present in the chain 7604 effect = new EffectModule(this, chain, desc, id, sessionId); 7605 lStatus = effect->status(); 7606 if (lStatus != NO_ERROR) { 7607 goto Exit; 7608 } 7609 lStatus = chain->addEffect_l(effect); 7610 if (lStatus != NO_ERROR) { 7611 goto Exit; 7612 } 7613 effectCreated = true; 7614 7615 effect->setDevice(mDevice); 7616 effect->setMode(mAudioFlinger->getMode()); 7617 } 7618 // create effect handle and connect it to effect module 7619 handle = new EffectHandle(effect, client, effectClient, priority); 7620 lStatus = effect->addHandle(handle.get()); 7621 if (enabled != NULL) { 7622 *enabled = (int)effect->isEnabled(); 7623 } 7624 } 7625 7626Exit: 7627 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7628 Mutex::Autolock _l(mLock); 7629 if (effectCreated) { 7630 chain->removeEffect_l(effect); 7631 } 7632 if (effectRegistered) { 7633 AudioSystem::unregisterEffect(effect->id()); 7634 } 7635 if (chainCreated) { 7636 removeEffectChain_l(chain); 7637 } 7638 handle.clear(); 7639 } 7640 7641 if (status != NULL) { 7642 *status = lStatus; 7643 } 7644 return handle; 7645} 7646 7647sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7648{ 7649 Mutex::Autolock _l(mLock); 7650 return getEffect_l(sessionId, effectId); 7651} 7652 7653sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7654{ 7655 sp<EffectChain> chain = getEffectChain_l(sessionId); 7656 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7657} 7658 7659// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7660// PlaybackThread::mLock held 7661status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7662{ 7663 // check for existing effect chain with the requested audio session 7664 int sessionId = effect->sessionId(); 7665 sp<EffectChain> chain = getEffectChain_l(sessionId); 7666 bool chainCreated = false; 7667 7668 if (chain == 0) { 7669 // create a new chain for this session 7670 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7671 chain = new EffectChain(this, sessionId); 7672 addEffectChain_l(chain); 7673 chain->setStrategy(getStrategyForSession_l(sessionId)); 7674 chainCreated = true; 7675 } 7676 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7677 7678 if (chain->getEffectFromId_l(effect->id()) != 0) { 7679 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7680 this, effect->desc().name, chain.get()); 7681 return BAD_VALUE; 7682 } 7683 7684 status_t status = chain->addEffect_l(effect); 7685 if (status != NO_ERROR) { 7686 if (chainCreated) { 7687 removeEffectChain_l(chain); 7688 } 7689 return status; 7690 } 7691 7692 effect->setDevice(mDevice); 7693 effect->setMode(mAudioFlinger->getMode()); 7694 return NO_ERROR; 7695} 7696 7697void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7698 7699 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7700 effect_descriptor_t desc = effect->desc(); 7701 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7702 detachAuxEffect_l(effect->id()); 7703 } 7704 7705 sp<EffectChain> chain = effect->chain().promote(); 7706 if (chain != 0) { 7707 // remove effect chain if removing last effect 7708 if (chain->removeEffect_l(effect) == 0) { 7709 removeEffectChain_l(chain); 7710 } 7711 } else { 7712 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7713 } 7714} 7715 7716void AudioFlinger::ThreadBase::lockEffectChains_l( 7717 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7718{ 7719 effectChains = mEffectChains; 7720 for (size_t i = 0; i < mEffectChains.size(); i++) { 7721 mEffectChains[i]->lock(); 7722 } 7723} 7724 7725void AudioFlinger::ThreadBase::unlockEffectChains( 7726 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7727{ 7728 for (size_t i = 0; i < effectChains.size(); i++) { 7729 effectChains[i]->unlock(); 7730 } 7731} 7732 7733sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7734{ 7735 Mutex::Autolock _l(mLock); 7736 return getEffectChain_l(sessionId); 7737} 7738 7739sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7740{ 7741 size_t size = mEffectChains.size(); 7742 for (size_t i = 0; i < size; i++) { 7743 if (mEffectChains[i]->sessionId() == sessionId) { 7744 return mEffectChains[i]; 7745 } 7746 } 7747 return 0; 7748} 7749 7750void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7751{ 7752 Mutex::Autolock _l(mLock); 7753 size_t size = mEffectChains.size(); 7754 for (size_t i = 0; i < size; i++) { 7755 mEffectChains[i]->setMode_l(mode); 7756 } 7757} 7758 7759void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7760 EffectHandle *handle, 7761 bool unpinIfLast) { 7762 7763 Mutex::Autolock _l(mLock); 7764 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7765 // delete the effect module if removing last handle on it 7766 if (effect->removeHandle(handle) == 0) { 7767 if (!effect->isPinned() || unpinIfLast) { 7768 removeEffect_l(effect); 7769 AudioSystem::unregisterEffect(effect->id()); 7770 } 7771 } 7772} 7773 7774status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7775{ 7776 int session = chain->sessionId(); 7777 int16_t *buffer = mMixBuffer; 7778 bool ownsBuffer = false; 7779 7780 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7781 if (session > 0) { 7782 // Only one effect chain can be present in direct output thread and it uses 7783 // the mix buffer as input 7784 if (mType != DIRECT) { 7785 size_t numSamples = mNormalFrameCount * mChannelCount; 7786 buffer = new int16_t[numSamples]; 7787 memset(buffer, 0, numSamples * sizeof(int16_t)); 7788 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7789 ownsBuffer = true; 7790 } 7791 7792 // Attach all tracks with same session ID to this chain. 7793 for (size_t i = 0; i < mTracks.size(); ++i) { 7794 sp<Track> track = mTracks[i]; 7795 if (session == track->sessionId()) { 7796 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7797 track->setMainBuffer(buffer); 7798 chain->incTrackCnt(); 7799 } 7800 } 7801 7802 // indicate all active tracks in the chain 7803 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7804 sp<Track> track = mActiveTracks[i].promote(); 7805 if (track == 0) continue; 7806 if (session == track->sessionId()) { 7807 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7808 chain->incActiveTrackCnt(); 7809 } 7810 } 7811 } 7812 7813 chain->setInBuffer(buffer, ownsBuffer); 7814 chain->setOutBuffer(mMixBuffer); 7815 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7816 // chains list in order to be processed last as it contains output stage effects 7817 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7818 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7819 // after track specific effects and before output stage 7820 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7821 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7822 // Effect chain for other sessions are inserted at beginning of effect 7823 // chains list to be processed before output mix effects. Relative order between other 7824 // sessions is not important 7825 size_t size = mEffectChains.size(); 7826 size_t i = 0; 7827 for (i = 0; i < size; i++) { 7828 if (mEffectChains[i]->sessionId() < session) break; 7829 } 7830 mEffectChains.insertAt(chain, i); 7831 checkSuspendOnAddEffectChain_l(chain); 7832 7833 return NO_ERROR; 7834} 7835 7836size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7837{ 7838 int session = chain->sessionId(); 7839 7840 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7841 7842 for (size_t i = 0; i < mEffectChains.size(); i++) { 7843 if (chain == mEffectChains[i]) { 7844 mEffectChains.removeAt(i); 7845 // detach all active tracks from the chain 7846 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7847 sp<Track> track = mActiveTracks[i].promote(); 7848 if (track == 0) continue; 7849 if (session == track->sessionId()) { 7850 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7851 chain.get(), session); 7852 chain->decActiveTrackCnt(); 7853 } 7854 } 7855 7856 // detach all tracks with same session ID from this chain 7857 for (size_t i = 0; i < mTracks.size(); ++i) { 7858 sp<Track> track = mTracks[i]; 7859 if (session == track->sessionId()) { 7860 track->setMainBuffer(mMixBuffer); 7861 chain->decTrackCnt(); 7862 } 7863 } 7864 break; 7865 } 7866 } 7867 return mEffectChains.size(); 7868} 7869 7870status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7871 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7872{ 7873 Mutex::Autolock _l(mLock); 7874 return attachAuxEffect_l(track, EffectId); 7875} 7876 7877status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7878 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7879{ 7880 status_t status = NO_ERROR; 7881 7882 if (EffectId == 0) { 7883 track->setAuxBuffer(0, NULL); 7884 } else { 7885 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7886 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7887 if (effect != 0) { 7888 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7889 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7890 } else { 7891 status = INVALID_OPERATION; 7892 } 7893 } else { 7894 status = BAD_VALUE; 7895 } 7896 } 7897 return status; 7898} 7899 7900void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7901{ 7902 for (size_t i = 0; i < mTracks.size(); ++i) { 7903 sp<Track> track = mTracks[i]; 7904 if (track->auxEffectId() == effectId) { 7905 attachAuxEffect_l(track, 0); 7906 } 7907 } 7908} 7909 7910status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7911{ 7912 // only one chain per input thread 7913 if (mEffectChains.size() != 0) { 7914 return INVALID_OPERATION; 7915 } 7916 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7917 7918 chain->setInBuffer(NULL); 7919 chain->setOutBuffer(NULL); 7920 7921 checkSuspendOnAddEffectChain_l(chain); 7922 7923 mEffectChains.add(chain); 7924 7925 return NO_ERROR; 7926} 7927 7928size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7929{ 7930 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7931 ALOGW_IF(mEffectChains.size() != 1, 7932 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7933 chain.get(), mEffectChains.size(), this); 7934 if (mEffectChains.size() == 1) { 7935 mEffectChains.removeAt(0); 7936 } 7937 return 0; 7938} 7939 7940// ---------------------------------------------------------------------------- 7941// EffectModule implementation 7942// ---------------------------------------------------------------------------- 7943 7944#undef LOG_TAG 7945#define LOG_TAG "AudioFlinger::EffectModule" 7946 7947AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7948 const wp<AudioFlinger::EffectChain>& chain, 7949 effect_descriptor_t *desc, 7950 int id, 7951 int sessionId) 7952 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 7953 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 7954 // mDescriptor is set below 7955 // mConfig is set by configure() and not used before then 7956 mEffectInterface(NULL), 7957 mStatus(NO_INIT), mState(IDLE), 7958 // mMaxDisableWaitCnt is set by configure() and not used before then 7959 // mDisableWaitCnt is set by process() and updateState() and not used before then 7960 mSuspended(false) 7961{ 7962 ALOGV("Constructor %p", this); 7963 int lStatus; 7964 if (thread == NULL) { 7965 return; 7966 } 7967 7968 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7969 7970 // create effect engine from effect factory 7971 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7972 7973 if (mStatus != NO_ERROR) { 7974 return; 7975 } 7976 lStatus = init(); 7977 if (lStatus < 0) { 7978 mStatus = lStatus; 7979 goto Error; 7980 } 7981 7982 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7983 return; 7984Error: 7985 EffectRelease(mEffectInterface); 7986 mEffectInterface = NULL; 7987 ALOGV("Constructor Error %d", mStatus); 7988} 7989 7990AudioFlinger::EffectModule::~EffectModule() 7991{ 7992 ALOGV("Destructor %p", this); 7993 if (mEffectInterface != NULL) { 7994 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7995 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7996 sp<ThreadBase> thread = mThread.promote(); 7997 if (thread != 0) { 7998 audio_stream_t *stream = thread->stream(); 7999 if (stream != NULL) { 8000 stream->remove_audio_effect(stream, mEffectInterface); 8001 } 8002 } 8003 } 8004 // release effect engine 8005 EffectRelease(mEffectInterface); 8006 } 8007} 8008 8009status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8010{ 8011 status_t status; 8012 8013 Mutex::Autolock _l(mLock); 8014 int priority = handle->priority(); 8015 size_t size = mHandles.size(); 8016 EffectHandle *controlHandle = NULL; 8017 size_t i; 8018 for (i = 0; i < size; i++) { 8019 EffectHandle *h = mHandles[i]; 8020 if (h == NULL || h->destroyed_l()) continue; 8021 // first non destroyed handle is considered in control 8022 if (controlHandle == NULL) 8023 controlHandle = h; 8024 if (h->priority() <= priority) break; 8025 } 8026 // if inserted in first place, move effect control from previous owner to this handle 8027 if (i == 0) { 8028 bool enabled = false; 8029 if (controlHandle != NULL) { 8030 enabled = controlHandle->enabled(); 8031 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8032 } 8033 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8034 status = NO_ERROR; 8035 } else { 8036 status = ALREADY_EXISTS; 8037 } 8038 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8039 mHandles.insertAt(handle, i); 8040 return status; 8041} 8042 8043size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8044{ 8045 Mutex::Autolock _l(mLock); 8046 size_t size = mHandles.size(); 8047 size_t i; 8048 for (i = 0; i < size; i++) { 8049 if (mHandles[i] == handle) break; 8050 } 8051 if (i == size) { 8052 return size; 8053 } 8054 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8055 8056 mHandles.removeAt(i); 8057 // if removed from first place, move effect control from this handle to next in line 8058 if (i == 0) { 8059 EffectHandle *h = controlHandle_l(); 8060 if (h != NULL) { 8061 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8062 } 8063 } 8064 8065 // Prevent calls to process() and other functions on effect interface from now on. 8066 // The effect engine will be released by the destructor when the last strong reference on 8067 // this object is released which can happen after next process is called. 8068 if (mHandles.size() == 0 && !mPinned) { 8069 mState = DESTROYED; 8070 } 8071 8072 return size; 8073} 8074 8075// must be called with EffectModule::mLock held 8076AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8077{ 8078 // the first valid handle in the list has control over the module 8079 for (size_t i = 0; i < mHandles.size(); i++) { 8080 EffectHandle *h = mHandles[i]; 8081 if (h != NULL && !h->destroyed_l()) { 8082 return h; 8083 } 8084 } 8085 8086 return NULL; 8087} 8088 8089size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8090{ 8091 ALOGV("disconnect() %p handle %p", this, handle); 8092 // keep a strong reference on this EffectModule to avoid calling the 8093 // destructor before we exit 8094 sp<EffectModule> keep(this); 8095 { 8096 sp<ThreadBase> thread = mThread.promote(); 8097 if (thread != 0) { 8098 thread->disconnectEffect(keep, handle, unpinIfLast); 8099 } 8100 } 8101 return mHandles.size(); 8102} 8103 8104void AudioFlinger::EffectModule::updateState() { 8105 Mutex::Autolock _l(mLock); 8106 8107 switch (mState) { 8108 case RESTART: 8109 reset_l(); 8110 // FALL THROUGH 8111 8112 case STARTING: 8113 // clear auxiliary effect input buffer for next accumulation 8114 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8115 memset(mConfig.inputCfg.buffer.raw, 8116 0, 8117 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8118 } 8119 start_l(); 8120 mState = ACTIVE; 8121 break; 8122 case STOPPING: 8123 stop_l(); 8124 mDisableWaitCnt = mMaxDisableWaitCnt; 8125 mState = STOPPED; 8126 break; 8127 case STOPPED: 8128 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8129 // turn off sequence. 8130 if (--mDisableWaitCnt == 0) { 8131 reset_l(); 8132 mState = IDLE; 8133 } 8134 break; 8135 default: //IDLE , ACTIVE, DESTROYED 8136 break; 8137 } 8138} 8139 8140void AudioFlinger::EffectModule::process() 8141{ 8142 Mutex::Autolock _l(mLock); 8143 8144 if (mState == DESTROYED || mEffectInterface == NULL || 8145 mConfig.inputCfg.buffer.raw == NULL || 8146 mConfig.outputCfg.buffer.raw == NULL) { 8147 return; 8148 } 8149 8150 if (isProcessEnabled()) { 8151 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8152 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8153 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8154 mConfig.inputCfg.buffer.s32, 8155 mConfig.inputCfg.buffer.frameCount/2); 8156 } 8157 8158 // do the actual processing in the effect engine 8159 int ret = (*mEffectInterface)->process(mEffectInterface, 8160 &mConfig.inputCfg.buffer, 8161 &mConfig.outputCfg.buffer); 8162 8163 // force transition to IDLE state when engine is ready 8164 if (mState == STOPPED && ret == -ENODATA) { 8165 mDisableWaitCnt = 1; 8166 } 8167 8168 // clear auxiliary effect input buffer for next accumulation 8169 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8170 memset(mConfig.inputCfg.buffer.raw, 0, 8171 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8172 } 8173 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8174 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8175 // If an insert effect is idle and input buffer is different from output buffer, 8176 // accumulate input onto output 8177 sp<EffectChain> chain = mChain.promote(); 8178 if (chain != 0 && chain->activeTrackCnt() != 0) { 8179 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8180 int16_t *in = mConfig.inputCfg.buffer.s16; 8181 int16_t *out = mConfig.outputCfg.buffer.s16; 8182 for (size_t i = 0; i < frameCnt; i++) { 8183 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8184 } 8185 } 8186 } 8187} 8188 8189void AudioFlinger::EffectModule::reset_l() 8190{ 8191 if (mEffectInterface == NULL) { 8192 return; 8193 } 8194 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8195} 8196 8197status_t AudioFlinger::EffectModule::configure() 8198{ 8199 if (mEffectInterface == NULL) { 8200 return NO_INIT; 8201 } 8202 8203 sp<ThreadBase> thread = mThread.promote(); 8204 if (thread == 0) { 8205 return DEAD_OBJECT; 8206 } 8207 8208 // TODO: handle configuration of effects replacing track process 8209 audio_channel_mask_t channelMask = thread->channelMask(); 8210 8211 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8212 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8213 } else { 8214 mConfig.inputCfg.channels = channelMask; 8215 } 8216 mConfig.outputCfg.channels = channelMask; 8217 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8218 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8219 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8220 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8221 mConfig.inputCfg.bufferProvider.cookie = NULL; 8222 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8223 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8224 mConfig.outputCfg.bufferProvider.cookie = NULL; 8225 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8226 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8227 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8228 // Insert effect: 8229 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8230 // always overwrites output buffer: input buffer == output buffer 8231 // - in other sessions: 8232 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8233 // other effect: overwrites output buffer: input buffer == output buffer 8234 // Auxiliary effect: 8235 // accumulates in output buffer: input buffer != output buffer 8236 // Therefore: accumulate <=> input buffer != output buffer 8237 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8238 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8239 } else { 8240 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8241 } 8242 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8243 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8244 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8245 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8246 8247 ALOGV("configure() %p thread %p buffer %p framecount %d", 8248 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8249 8250 status_t cmdStatus; 8251 uint32_t size = sizeof(int); 8252 status_t status = (*mEffectInterface)->command(mEffectInterface, 8253 EFFECT_CMD_SET_CONFIG, 8254 sizeof(effect_config_t), 8255 &mConfig, 8256 &size, 8257 &cmdStatus); 8258 if (status == 0) { 8259 status = cmdStatus; 8260 } 8261 8262 if (status == 0 && 8263 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8264 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8265 effect_param_t *p = (effect_param_t *)buf32; 8266 8267 p->psize = sizeof(uint32_t); 8268 p->vsize = sizeof(uint32_t); 8269 size = sizeof(int); 8270 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8271 8272 uint32_t latency = 0; 8273 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8274 if (pbt != NULL) { 8275 latency = pbt->latency_l(); 8276 } 8277 8278 *((int32_t *)p->data + 1)= latency; 8279 (*mEffectInterface)->command(mEffectInterface, 8280 EFFECT_CMD_SET_PARAM, 8281 sizeof(effect_param_t) + 8, 8282 &buf32, 8283 &size, 8284 &cmdStatus); 8285 } 8286 8287 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8288 (1000 * mConfig.outputCfg.buffer.frameCount); 8289 8290 return status; 8291} 8292 8293status_t AudioFlinger::EffectModule::init() 8294{ 8295 Mutex::Autolock _l(mLock); 8296 if (mEffectInterface == NULL) { 8297 return NO_INIT; 8298 } 8299 status_t cmdStatus; 8300 uint32_t size = sizeof(status_t); 8301 status_t status = (*mEffectInterface)->command(mEffectInterface, 8302 EFFECT_CMD_INIT, 8303 0, 8304 NULL, 8305 &size, 8306 &cmdStatus); 8307 if (status == 0) { 8308 status = cmdStatus; 8309 } 8310 return status; 8311} 8312 8313status_t AudioFlinger::EffectModule::start() 8314{ 8315 Mutex::Autolock _l(mLock); 8316 return start_l(); 8317} 8318 8319status_t AudioFlinger::EffectModule::start_l() 8320{ 8321 if (mEffectInterface == NULL) { 8322 return NO_INIT; 8323 } 8324 status_t cmdStatus; 8325 uint32_t size = sizeof(status_t); 8326 status_t status = (*mEffectInterface)->command(mEffectInterface, 8327 EFFECT_CMD_ENABLE, 8328 0, 8329 NULL, 8330 &size, 8331 &cmdStatus); 8332 if (status == 0) { 8333 status = cmdStatus; 8334 } 8335 if (status == 0 && 8336 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8337 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8338 sp<ThreadBase> thread = mThread.promote(); 8339 if (thread != 0) { 8340 audio_stream_t *stream = thread->stream(); 8341 if (stream != NULL) { 8342 stream->add_audio_effect(stream, mEffectInterface); 8343 } 8344 } 8345 } 8346 return status; 8347} 8348 8349status_t AudioFlinger::EffectModule::stop() 8350{ 8351 Mutex::Autolock _l(mLock); 8352 return stop_l(); 8353} 8354 8355status_t AudioFlinger::EffectModule::stop_l() 8356{ 8357 if (mEffectInterface == NULL) { 8358 return NO_INIT; 8359 } 8360 status_t cmdStatus; 8361 uint32_t size = sizeof(status_t); 8362 status_t status = (*mEffectInterface)->command(mEffectInterface, 8363 EFFECT_CMD_DISABLE, 8364 0, 8365 NULL, 8366 &size, 8367 &cmdStatus); 8368 if (status == 0) { 8369 status = cmdStatus; 8370 } 8371 if (status == 0 && 8372 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8373 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8374 sp<ThreadBase> thread = mThread.promote(); 8375 if (thread != 0) { 8376 audio_stream_t *stream = thread->stream(); 8377 if (stream != NULL) { 8378 stream->remove_audio_effect(stream, mEffectInterface); 8379 } 8380 } 8381 } 8382 return status; 8383} 8384 8385status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8386 uint32_t cmdSize, 8387 void *pCmdData, 8388 uint32_t *replySize, 8389 void *pReplyData) 8390{ 8391 Mutex::Autolock _l(mLock); 8392// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8393 8394 if (mState == DESTROYED || mEffectInterface == NULL) { 8395 return NO_INIT; 8396 } 8397 status_t status = (*mEffectInterface)->command(mEffectInterface, 8398 cmdCode, 8399 cmdSize, 8400 pCmdData, 8401 replySize, 8402 pReplyData); 8403 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8404 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8405 for (size_t i = 1; i < mHandles.size(); i++) { 8406 EffectHandle *h = mHandles[i]; 8407 if (h != NULL && !h->destroyed_l()) { 8408 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8409 } 8410 } 8411 } 8412 return status; 8413} 8414 8415status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8416{ 8417 Mutex::Autolock _l(mLock); 8418 return setEnabled_l(enabled); 8419} 8420 8421// must be called with EffectModule::mLock held 8422status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8423{ 8424 8425 ALOGV("setEnabled %p enabled %d", this, enabled); 8426 8427 if (enabled != isEnabled()) { 8428 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8429 if (enabled && status != NO_ERROR) { 8430 return status; 8431 } 8432 8433 switch (mState) { 8434 // going from disabled to enabled 8435 case IDLE: 8436 mState = STARTING; 8437 break; 8438 case STOPPED: 8439 mState = RESTART; 8440 break; 8441 case STOPPING: 8442 mState = ACTIVE; 8443 break; 8444 8445 // going from enabled to disabled 8446 case RESTART: 8447 mState = STOPPED; 8448 break; 8449 case STARTING: 8450 mState = IDLE; 8451 break; 8452 case ACTIVE: 8453 mState = STOPPING; 8454 break; 8455 case DESTROYED: 8456 return NO_ERROR; // simply ignore as we are being destroyed 8457 } 8458 for (size_t i = 1; i < mHandles.size(); i++) { 8459 EffectHandle *h = mHandles[i]; 8460 if (h != NULL && !h->destroyed_l()) { 8461 h->setEnabled(enabled); 8462 } 8463 } 8464 } 8465 return NO_ERROR; 8466} 8467 8468bool AudioFlinger::EffectModule::isEnabled() const 8469{ 8470 switch (mState) { 8471 case RESTART: 8472 case STARTING: 8473 case ACTIVE: 8474 return true; 8475 case IDLE: 8476 case STOPPING: 8477 case STOPPED: 8478 case DESTROYED: 8479 default: 8480 return false; 8481 } 8482} 8483 8484bool AudioFlinger::EffectModule::isProcessEnabled() const 8485{ 8486 switch (mState) { 8487 case RESTART: 8488 case ACTIVE: 8489 case STOPPING: 8490 case STOPPED: 8491 return true; 8492 case IDLE: 8493 case STARTING: 8494 case DESTROYED: 8495 default: 8496 return false; 8497 } 8498} 8499 8500status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8501{ 8502 Mutex::Autolock _l(mLock); 8503 status_t status = NO_ERROR; 8504 8505 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8506 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8507 if (isProcessEnabled() && 8508 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8509 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8510 status_t cmdStatus; 8511 uint32_t volume[2]; 8512 uint32_t *pVolume = NULL; 8513 uint32_t size = sizeof(volume); 8514 volume[0] = *left; 8515 volume[1] = *right; 8516 if (controller) { 8517 pVolume = volume; 8518 } 8519 status = (*mEffectInterface)->command(mEffectInterface, 8520 EFFECT_CMD_SET_VOLUME, 8521 size, 8522 volume, 8523 &size, 8524 pVolume); 8525 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8526 *left = volume[0]; 8527 *right = volume[1]; 8528 } 8529 } 8530 return status; 8531} 8532 8533status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8534{ 8535 Mutex::Autolock _l(mLock); 8536 status_t status = NO_ERROR; 8537 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8538 // audio pre processing modules on RecordThread can receive both output and 8539 // input device indication in the same call 8540 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8541 if (dev) { 8542 status_t cmdStatus; 8543 uint32_t size = sizeof(status_t); 8544 8545 status = (*mEffectInterface)->command(mEffectInterface, 8546 EFFECT_CMD_SET_DEVICE, 8547 sizeof(uint32_t), 8548 &dev, 8549 &size, 8550 &cmdStatus); 8551 if (status == NO_ERROR) { 8552 status = cmdStatus; 8553 } 8554 } 8555 dev = device & AUDIO_DEVICE_IN_ALL; 8556 if (dev) { 8557 status_t cmdStatus; 8558 uint32_t size = sizeof(status_t); 8559 8560 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8561 EFFECT_CMD_SET_INPUT_DEVICE, 8562 sizeof(uint32_t), 8563 &dev, 8564 &size, 8565 &cmdStatus); 8566 if (status2 == NO_ERROR) { 8567 status2 = cmdStatus; 8568 } 8569 if (status == NO_ERROR) { 8570 status = status2; 8571 } 8572 } 8573 } 8574 return status; 8575} 8576 8577status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8578{ 8579 Mutex::Autolock _l(mLock); 8580 status_t status = NO_ERROR; 8581 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8582 status_t cmdStatus; 8583 uint32_t size = sizeof(status_t); 8584 status = (*mEffectInterface)->command(mEffectInterface, 8585 EFFECT_CMD_SET_AUDIO_MODE, 8586 sizeof(audio_mode_t), 8587 &mode, 8588 &size, 8589 &cmdStatus); 8590 if (status == NO_ERROR) { 8591 status = cmdStatus; 8592 } 8593 } 8594 return status; 8595} 8596 8597void AudioFlinger::EffectModule::setSuspended(bool suspended) 8598{ 8599 Mutex::Autolock _l(mLock); 8600 mSuspended = suspended; 8601} 8602 8603bool AudioFlinger::EffectModule::suspended() const 8604{ 8605 Mutex::Autolock _l(mLock); 8606 return mSuspended; 8607} 8608 8609bool AudioFlinger::EffectModule::purgeHandles() 8610{ 8611 bool enabled = false; 8612 Mutex::Autolock _l(mLock); 8613 for (size_t i = 0; i < mHandles.size(); i++) { 8614 EffectHandle *handle = mHandles[i]; 8615 if (handle != NULL && !handle->destroyed_l()) { 8616 handle->effect().clear(); 8617 if (handle->hasControl()) { 8618 enabled = handle->enabled(); 8619 } 8620 } 8621 } 8622 return enabled; 8623} 8624 8625status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8626{ 8627 const size_t SIZE = 256; 8628 char buffer[SIZE]; 8629 String8 result; 8630 8631 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8632 result.append(buffer); 8633 8634 bool locked = tryLock(mLock); 8635 // failed to lock - AudioFlinger is probably deadlocked 8636 if (!locked) { 8637 result.append("\t\tCould not lock Fx mutex:\n"); 8638 } 8639 8640 result.append("\t\tSession Status State Engine:\n"); 8641 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8642 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8643 result.append(buffer); 8644 8645 result.append("\t\tDescriptor:\n"); 8646 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8647 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8648 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8649 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8650 result.append(buffer); 8651 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8652 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8653 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8654 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8655 result.append(buffer); 8656 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8657 mDescriptor.apiVersion, 8658 mDescriptor.flags); 8659 result.append(buffer); 8660 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8661 mDescriptor.name); 8662 result.append(buffer); 8663 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8664 mDescriptor.implementor); 8665 result.append(buffer); 8666 8667 result.append("\t\t- Input configuration:\n"); 8668 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8669 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8670 (uint32_t)mConfig.inputCfg.buffer.raw, 8671 mConfig.inputCfg.buffer.frameCount, 8672 mConfig.inputCfg.samplingRate, 8673 mConfig.inputCfg.channels, 8674 mConfig.inputCfg.format); 8675 result.append(buffer); 8676 8677 result.append("\t\t- Output configuration:\n"); 8678 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8679 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8680 (uint32_t)mConfig.outputCfg.buffer.raw, 8681 mConfig.outputCfg.buffer.frameCount, 8682 mConfig.outputCfg.samplingRate, 8683 mConfig.outputCfg.channels, 8684 mConfig.outputCfg.format); 8685 result.append(buffer); 8686 8687 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8688 result.append(buffer); 8689 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8690 for (size_t i = 0; i < mHandles.size(); ++i) { 8691 EffectHandle *handle = mHandles[i]; 8692 if (handle != NULL && !handle->destroyed_l()) { 8693 handle->dump(buffer, SIZE); 8694 result.append(buffer); 8695 } 8696 } 8697 8698 result.append("\n"); 8699 8700 write(fd, result.string(), result.length()); 8701 8702 if (locked) { 8703 mLock.unlock(); 8704 } 8705 8706 return NO_ERROR; 8707} 8708 8709// ---------------------------------------------------------------------------- 8710// EffectHandle implementation 8711// ---------------------------------------------------------------------------- 8712 8713#undef LOG_TAG 8714#define LOG_TAG "AudioFlinger::EffectHandle" 8715 8716AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8717 const sp<AudioFlinger::Client>& client, 8718 const sp<IEffectClient>& effectClient, 8719 int32_t priority) 8720 : BnEffect(), 8721 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8722 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8723{ 8724 ALOGV("constructor %p", this); 8725 8726 if (client == 0) { 8727 return; 8728 } 8729 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8730 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8731 if (mCblkMemory != 0) { 8732 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8733 8734 if (mCblk != NULL) { 8735 new(mCblk) effect_param_cblk_t(); 8736 mBuffer = (uint8_t *)mCblk + bufOffset; 8737 } 8738 } else { 8739 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8740 return; 8741 } 8742} 8743 8744AudioFlinger::EffectHandle::~EffectHandle() 8745{ 8746 ALOGV("Destructor %p", this); 8747 8748 if (mEffect == 0) { 8749 mDestroyed = true; 8750 return; 8751 } 8752 mEffect->lock(); 8753 mDestroyed = true; 8754 mEffect->unlock(); 8755 disconnect(false); 8756} 8757 8758status_t AudioFlinger::EffectHandle::enable() 8759{ 8760 ALOGV("enable %p", this); 8761 if (!mHasControl) return INVALID_OPERATION; 8762 if (mEffect == 0) return DEAD_OBJECT; 8763 8764 if (mEnabled) { 8765 return NO_ERROR; 8766 } 8767 8768 mEnabled = true; 8769 8770 sp<ThreadBase> thread = mEffect->thread().promote(); 8771 if (thread != 0) { 8772 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8773 } 8774 8775 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8776 if (mEffect->suspended()) { 8777 return NO_ERROR; 8778 } 8779 8780 status_t status = mEffect->setEnabled(true); 8781 if (status != NO_ERROR) { 8782 if (thread != 0) { 8783 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8784 } 8785 mEnabled = false; 8786 } 8787 return status; 8788} 8789 8790status_t AudioFlinger::EffectHandle::disable() 8791{ 8792 ALOGV("disable %p", this); 8793 if (!mHasControl) return INVALID_OPERATION; 8794 if (mEffect == 0) return DEAD_OBJECT; 8795 8796 if (!mEnabled) { 8797 return NO_ERROR; 8798 } 8799 mEnabled = false; 8800 8801 if (mEffect->suspended()) { 8802 return NO_ERROR; 8803 } 8804 8805 status_t status = mEffect->setEnabled(false); 8806 8807 sp<ThreadBase> thread = mEffect->thread().promote(); 8808 if (thread != 0) { 8809 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8810 } 8811 8812 return status; 8813} 8814 8815void AudioFlinger::EffectHandle::disconnect() 8816{ 8817 disconnect(true); 8818} 8819 8820void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8821{ 8822 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8823 if (mEffect == 0) { 8824 return; 8825 } 8826 // restore suspended effects if the disconnected handle was enabled and the last one. 8827 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8828 sp<ThreadBase> thread = mEffect->thread().promote(); 8829 if (thread != 0) { 8830 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8831 } 8832 } 8833 8834 // release sp on module => module destructor can be called now 8835 mEffect.clear(); 8836 if (mClient != 0) { 8837 if (mCblk != NULL) { 8838 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8839 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8840 } 8841 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8842 // Client destructor must run with AudioFlinger mutex locked 8843 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8844 mClient.clear(); 8845 } 8846} 8847 8848status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8849 uint32_t cmdSize, 8850 void *pCmdData, 8851 uint32_t *replySize, 8852 void *pReplyData) 8853{ 8854// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8855// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8856 8857 // only get parameter command is permitted for applications not controlling the effect 8858 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8859 return INVALID_OPERATION; 8860 } 8861 if (mEffect == 0) return DEAD_OBJECT; 8862 if (mClient == 0) return INVALID_OPERATION; 8863 8864 // handle commands that are not forwarded transparently to effect engine 8865 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8866 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8867 // no risk to block the whole media server process or mixer threads is we are stuck here 8868 Mutex::Autolock _l(mCblk->lock); 8869 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8870 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8871 mCblk->serverIndex = 0; 8872 mCblk->clientIndex = 0; 8873 return BAD_VALUE; 8874 } 8875 status_t status = NO_ERROR; 8876 while (mCblk->serverIndex < mCblk->clientIndex) { 8877 int reply; 8878 uint32_t rsize = sizeof(int); 8879 int *p = (int *)(mBuffer + mCblk->serverIndex); 8880 int size = *p++; 8881 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8882 ALOGW("command(): invalid parameter block size"); 8883 break; 8884 } 8885 effect_param_t *param = (effect_param_t *)p; 8886 if (param->psize == 0 || param->vsize == 0) { 8887 ALOGW("command(): null parameter or value size"); 8888 mCblk->serverIndex += size; 8889 continue; 8890 } 8891 uint32_t psize = sizeof(effect_param_t) + 8892 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8893 param->vsize; 8894 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8895 psize, 8896 p, 8897 &rsize, 8898 &reply); 8899 // stop at first error encountered 8900 if (ret != NO_ERROR) { 8901 status = ret; 8902 *(int *)pReplyData = reply; 8903 break; 8904 } else if (reply != NO_ERROR) { 8905 *(int *)pReplyData = reply; 8906 break; 8907 } 8908 mCblk->serverIndex += size; 8909 } 8910 mCblk->serverIndex = 0; 8911 mCblk->clientIndex = 0; 8912 return status; 8913 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8914 *(int *)pReplyData = NO_ERROR; 8915 return enable(); 8916 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8917 *(int *)pReplyData = NO_ERROR; 8918 return disable(); 8919 } 8920 8921 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8922} 8923 8924void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8925{ 8926 ALOGV("setControl %p control %d", this, hasControl); 8927 8928 mHasControl = hasControl; 8929 mEnabled = enabled; 8930 8931 if (signal && mEffectClient != 0) { 8932 mEffectClient->controlStatusChanged(hasControl); 8933 } 8934} 8935 8936void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8937 uint32_t cmdSize, 8938 void *pCmdData, 8939 uint32_t replySize, 8940 void *pReplyData) 8941{ 8942 if (mEffectClient != 0) { 8943 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8944 } 8945} 8946 8947 8948 8949void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8950{ 8951 if (mEffectClient != 0) { 8952 mEffectClient->enableStatusChanged(enabled); 8953 } 8954} 8955 8956status_t AudioFlinger::EffectHandle::onTransact( 8957 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8958{ 8959 return BnEffect::onTransact(code, data, reply, flags); 8960} 8961 8962 8963void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8964{ 8965 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8966 8967 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8968 (mClient == 0) ? getpid_cached : mClient->pid(), 8969 mPriority, 8970 mHasControl, 8971 !locked, 8972 mCblk ? mCblk->clientIndex : 0, 8973 mCblk ? mCblk->serverIndex : 0 8974 ); 8975 8976 if (locked) { 8977 mCblk->lock.unlock(); 8978 } 8979} 8980 8981#undef LOG_TAG 8982#define LOG_TAG "AudioFlinger::EffectChain" 8983 8984AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8985 int sessionId) 8986 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8987 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8988 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8989{ 8990 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8991 if (thread == NULL) { 8992 return; 8993 } 8994 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8995 thread->frameCount(); 8996} 8997 8998AudioFlinger::EffectChain::~EffectChain() 8999{ 9000 if (mOwnInBuffer) { 9001 delete mInBuffer; 9002 } 9003 9004} 9005 9006// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9007sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9008{ 9009 size_t size = mEffects.size(); 9010 9011 for (size_t i = 0; i < size; i++) { 9012 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9013 return mEffects[i]; 9014 } 9015 } 9016 return 0; 9017} 9018 9019// getEffectFromId_l() must be called with ThreadBase::mLock held 9020sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9021{ 9022 size_t size = mEffects.size(); 9023 9024 for (size_t i = 0; i < size; i++) { 9025 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9026 if (id == 0 || mEffects[i]->id() == id) { 9027 return mEffects[i]; 9028 } 9029 } 9030 return 0; 9031} 9032 9033// getEffectFromType_l() must be called with ThreadBase::mLock held 9034sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9035 const effect_uuid_t *type) 9036{ 9037 size_t size = mEffects.size(); 9038 9039 for (size_t i = 0; i < size; i++) { 9040 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9041 return mEffects[i]; 9042 } 9043 } 9044 return 0; 9045} 9046 9047void AudioFlinger::EffectChain::clearInputBuffer() 9048{ 9049 Mutex::Autolock _l(mLock); 9050 sp<ThreadBase> thread = mThread.promote(); 9051 if (thread == 0) { 9052 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9053 return; 9054 } 9055 clearInputBuffer_l(thread); 9056} 9057 9058// Must be called with EffectChain::mLock locked 9059void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9060{ 9061 size_t numSamples = thread->frameCount() * thread->channelCount(); 9062 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9063 9064} 9065 9066// Must be called with EffectChain::mLock locked 9067void AudioFlinger::EffectChain::process_l() 9068{ 9069 sp<ThreadBase> thread = mThread.promote(); 9070 if (thread == 0) { 9071 ALOGW("process_l(): cannot promote mixer thread"); 9072 return; 9073 } 9074 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9075 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9076 // always process effects unless no more tracks are on the session and the effect tail 9077 // has been rendered 9078 bool doProcess = true; 9079 if (!isGlobalSession) { 9080 bool tracksOnSession = (trackCnt() != 0); 9081 9082 if (!tracksOnSession && mTailBufferCount == 0) { 9083 doProcess = false; 9084 } 9085 9086 if (activeTrackCnt() == 0) { 9087 // if no track is active and the effect tail has not been rendered, 9088 // the input buffer must be cleared here as the mixer process will not do it 9089 if (tracksOnSession || mTailBufferCount > 0) { 9090 clearInputBuffer_l(thread); 9091 if (mTailBufferCount > 0) { 9092 mTailBufferCount--; 9093 } 9094 } 9095 } 9096 } 9097 9098 size_t size = mEffects.size(); 9099 if (doProcess) { 9100 for (size_t i = 0; i < size; i++) { 9101 mEffects[i]->process(); 9102 } 9103 } 9104 for (size_t i = 0; i < size; i++) { 9105 mEffects[i]->updateState(); 9106 } 9107} 9108 9109// addEffect_l() must be called with PlaybackThread::mLock held 9110status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9111{ 9112 effect_descriptor_t desc = effect->desc(); 9113 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9114 9115 Mutex::Autolock _l(mLock); 9116 effect->setChain(this); 9117 sp<ThreadBase> thread = mThread.promote(); 9118 if (thread == 0) { 9119 return NO_INIT; 9120 } 9121 effect->setThread(thread); 9122 9123 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9124 // Auxiliary effects are inserted at the beginning of mEffects vector as 9125 // they are processed first and accumulated in chain input buffer 9126 mEffects.insertAt(effect, 0); 9127 9128 // the input buffer for auxiliary effect contains mono samples in 9129 // 32 bit format. This is to avoid saturation in AudoMixer 9130 // accumulation stage. Saturation is done in EffectModule::process() before 9131 // calling the process in effect engine 9132 size_t numSamples = thread->frameCount(); 9133 int32_t *buffer = new int32_t[numSamples]; 9134 memset(buffer, 0, numSamples * sizeof(int32_t)); 9135 effect->setInBuffer((int16_t *)buffer); 9136 // auxiliary effects output samples to chain input buffer for further processing 9137 // by insert effects 9138 effect->setOutBuffer(mInBuffer); 9139 } else { 9140 // Insert effects are inserted at the end of mEffects vector as they are processed 9141 // after track and auxiliary effects. 9142 // Insert effect order as a function of indicated preference: 9143 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9144 // another effect is present 9145 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9146 // last effect claiming first position 9147 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9148 // first effect claiming last position 9149 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9150 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9151 // already present 9152 9153 size_t size = mEffects.size(); 9154 size_t idx_insert = size; 9155 ssize_t idx_insert_first = -1; 9156 ssize_t idx_insert_last = -1; 9157 9158 for (size_t i = 0; i < size; i++) { 9159 effect_descriptor_t d = mEffects[i]->desc(); 9160 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9161 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9162 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9163 // check invalid effect chaining combinations 9164 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9165 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9166 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9167 return INVALID_OPERATION; 9168 } 9169 // remember position of first insert effect and by default 9170 // select this as insert position for new effect 9171 if (idx_insert == size) { 9172 idx_insert = i; 9173 } 9174 // remember position of last insert effect claiming 9175 // first position 9176 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9177 idx_insert_first = i; 9178 } 9179 // remember position of first insert effect claiming 9180 // last position 9181 if (iPref == EFFECT_FLAG_INSERT_LAST && 9182 idx_insert_last == -1) { 9183 idx_insert_last = i; 9184 } 9185 } 9186 } 9187 9188 // modify idx_insert from first position if needed 9189 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9190 if (idx_insert_last != -1) { 9191 idx_insert = idx_insert_last; 9192 } else { 9193 idx_insert = size; 9194 } 9195 } else { 9196 if (idx_insert_first != -1) { 9197 idx_insert = idx_insert_first + 1; 9198 } 9199 } 9200 9201 // always read samples from chain input buffer 9202 effect->setInBuffer(mInBuffer); 9203 9204 // if last effect in the chain, output samples to chain 9205 // output buffer, otherwise to chain input buffer 9206 if (idx_insert == size) { 9207 if (idx_insert != 0) { 9208 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9209 mEffects[idx_insert-1]->configure(); 9210 } 9211 effect->setOutBuffer(mOutBuffer); 9212 } else { 9213 effect->setOutBuffer(mInBuffer); 9214 } 9215 mEffects.insertAt(effect, idx_insert); 9216 9217 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9218 } 9219 effect->configure(); 9220 return NO_ERROR; 9221} 9222 9223// removeEffect_l() must be called with PlaybackThread::mLock held 9224size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9225{ 9226 Mutex::Autolock _l(mLock); 9227 size_t size = mEffects.size(); 9228 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9229 9230 for (size_t i = 0; i < size; i++) { 9231 if (effect == mEffects[i]) { 9232 // calling stop here will remove pre-processing effect from the audio HAL. 9233 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9234 // the middle of a read from audio HAL 9235 if (mEffects[i]->state() == EffectModule::ACTIVE || 9236 mEffects[i]->state() == EffectModule::STOPPING) { 9237 mEffects[i]->stop(); 9238 } 9239 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9240 delete[] effect->inBuffer(); 9241 } else { 9242 if (i == size - 1 && i != 0) { 9243 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9244 mEffects[i - 1]->configure(); 9245 } 9246 } 9247 mEffects.removeAt(i); 9248 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9249 break; 9250 } 9251 } 9252 9253 return mEffects.size(); 9254} 9255 9256// setDevice_l() must be called with PlaybackThread::mLock held 9257void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9258{ 9259 size_t size = mEffects.size(); 9260 for (size_t i = 0; i < size; i++) { 9261 mEffects[i]->setDevice(device); 9262 } 9263} 9264 9265// setMode_l() must be called with PlaybackThread::mLock held 9266void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9267{ 9268 size_t size = mEffects.size(); 9269 for (size_t i = 0; i < size; i++) { 9270 mEffects[i]->setMode(mode); 9271 } 9272} 9273 9274// setVolume_l() must be called with PlaybackThread::mLock held 9275bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9276{ 9277 uint32_t newLeft = *left; 9278 uint32_t newRight = *right; 9279 bool hasControl = false; 9280 int ctrlIdx = -1; 9281 size_t size = mEffects.size(); 9282 9283 // first update volume controller 9284 for (size_t i = size; i > 0; i--) { 9285 if (mEffects[i - 1]->isProcessEnabled() && 9286 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9287 ctrlIdx = i - 1; 9288 hasControl = true; 9289 break; 9290 } 9291 } 9292 9293 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9294 if (hasControl) { 9295 *left = mNewLeftVolume; 9296 *right = mNewRightVolume; 9297 } 9298 return hasControl; 9299 } 9300 9301 mVolumeCtrlIdx = ctrlIdx; 9302 mLeftVolume = newLeft; 9303 mRightVolume = newRight; 9304 9305 // second get volume update from volume controller 9306 if (ctrlIdx >= 0) { 9307 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9308 mNewLeftVolume = newLeft; 9309 mNewRightVolume = newRight; 9310 } 9311 // then indicate volume to all other effects in chain. 9312 // Pass altered volume to effects before volume controller 9313 // and requested volume to effects after controller 9314 uint32_t lVol = newLeft; 9315 uint32_t rVol = newRight; 9316 9317 for (size_t i = 0; i < size; i++) { 9318 if ((int)i == ctrlIdx) continue; 9319 // this also works for ctrlIdx == -1 when there is no volume controller 9320 if ((int)i > ctrlIdx) { 9321 lVol = *left; 9322 rVol = *right; 9323 } 9324 mEffects[i]->setVolume(&lVol, &rVol, false); 9325 } 9326 *left = newLeft; 9327 *right = newRight; 9328 9329 return hasControl; 9330} 9331 9332status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9333{ 9334 const size_t SIZE = 256; 9335 char buffer[SIZE]; 9336 String8 result; 9337 9338 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9339 result.append(buffer); 9340 9341 bool locked = tryLock(mLock); 9342 // failed to lock - AudioFlinger is probably deadlocked 9343 if (!locked) { 9344 result.append("\tCould not lock mutex:\n"); 9345 } 9346 9347 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9348 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9349 mEffects.size(), 9350 (uint32_t)mInBuffer, 9351 (uint32_t)mOutBuffer, 9352 mActiveTrackCnt); 9353 result.append(buffer); 9354 write(fd, result.string(), result.size()); 9355 9356 for (size_t i = 0; i < mEffects.size(); ++i) { 9357 sp<EffectModule> effect = mEffects[i]; 9358 if (effect != 0) { 9359 effect->dump(fd, args); 9360 } 9361 } 9362 9363 if (locked) { 9364 mLock.unlock(); 9365 } 9366 9367 return NO_ERROR; 9368} 9369 9370// must be called with ThreadBase::mLock held 9371void AudioFlinger::EffectChain::setEffectSuspended_l( 9372 const effect_uuid_t *type, bool suspend) 9373{ 9374 sp<SuspendedEffectDesc> desc; 9375 // use effect type UUID timelow as key as there is no real risk of identical 9376 // timeLow fields among effect type UUIDs. 9377 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9378 if (suspend) { 9379 if (index >= 0) { 9380 desc = mSuspendedEffects.valueAt(index); 9381 } else { 9382 desc = new SuspendedEffectDesc(); 9383 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9384 mSuspendedEffects.add(type->timeLow, desc); 9385 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9386 } 9387 if (desc->mRefCount++ == 0) { 9388 sp<EffectModule> effect = getEffectIfEnabled(type); 9389 if (effect != 0) { 9390 desc->mEffect = effect; 9391 effect->setSuspended(true); 9392 effect->setEnabled(false); 9393 } 9394 } 9395 } else { 9396 if (index < 0) { 9397 return; 9398 } 9399 desc = mSuspendedEffects.valueAt(index); 9400 if (desc->mRefCount <= 0) { 9401 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9402 desc->mRefCount = 1; 9403 } 9404 if (--desc->mRefCount == 0) { 9405 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9406 if (desc->mEffect != 0) { 9407 sp<EffectModule> effect = desc->mEffect.promote(); 9408 if (effect != 0) { 9409 effect->setSuspended(false); 9410 effect->lock(); 9411 EffectHandle *handle = effect->controlHandle_l(); 9412 if (handle != NULL && !handle->destroyed_l()) { 9413 effect->setEnabled_l(handle->enabled()); 9414 } 9415 effect->unlock(); 9416 } 9417 desc->mEffect.clear(); 9418 } 9419 mSuspendedEffects.removeItemsAt(index); 9420 } 9421 } 9422} 9423 9424// must be called with ThreadBase::mLock held 9425void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9426{ 9427 sp<SuspendedEffectDesc> desc; 9428 9429 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9430 if (suspend) { 9431 if (index >= 0) { 9432 desc = mSuspendedEffects.valueAt(index); 9433 } else { 9434 desc = new SuspendedEffectDesc(); 9435 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9436 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9437 } 9438 if (desc->mRefCount++ == 0) { 9439 Vector< sp<EffectModule> > effects; 9440 getSuspendEligibleEffects(effects); 9441 for (size_t i = 0; i < effects.size(); i++) { 9442 setEffectSuspended_l(&effects[i]->desc().type, true); 9443 } 9444 } 9445 } else { 9446 if (index < 0) { 9447 return; 9448 } 9449 desc = mSuspendedEffects.valueAt(index); 9450 if (desc->mRefCount <= 0) { 9451 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9452 desc->mRefCount = 1; 9453 } 9454 if (--desc->mRefCount == 0) { 9455 Vector<const effect_uuid_t *> types; 9456 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9457 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9458 continue; 9459 } 9460 types.add(&mSuspendedEffects.valueAt(i)->mType); 9461 } 9462 for (size_t i = 0; i < types.size(); i++) { 9463 setEffectSuspended_l(types[i], false); 9464 } 9465 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9466 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9467 } 9468 } 9469} 9470 9471 9472// The volume effect is used for automated tests only 9473#ifndef OPENSL_ES_H_ 9474static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9475 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9476const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9477#endif //OPENSL_ES_H_ 9478 9479bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9480{ 9481 // auxiliary effects and visualizer are never suspended on output mix 9482 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9483 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9484 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9485 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9486 return false; 9487 } 9488 return true; 9489} 9490 9491void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9492{ 9493 effects.clear(); 9494 for (size_t i = 0; i < mEffects.size(); i++) { 9495 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9496 effects.add(mEffects[i]); 9497 } 9498 } 9499} 9500 9501sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9502 const effect_uuid_t *type) 9503{ 9504 sp<EffectModule> effect = getEffectFromType_l(type); 9505 return effect != 0 && effect->isEnabled() ? effect : 0; 9506} 9507 9508void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9509 bool enabled) 9510{ 9511 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9512 if (enabled) { 9513 if (index < 0) { 9514 // if the effect is not suspend check if all effects are suspended 9515 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9516 if (index < 0) { 9517 return; 9518 } 9519 if (!isEffectEligibleForSuspend(effect->desc())) { 9520 return; 9521 } 9522 setEffectSuspended_l(&effect->desc().type, enabled); 9523 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9524 if (index < 0) { 9525 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9526 return; 9527 } 9528 } 9529 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9530 effect->desc().type.timeLow); 9531 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9532 // if effect is requested to suspended but was not yet enabled, supend it now. 9533 if (desc->mEffect == 0) { 9534 desc->mEffect = effect; 9535 effect->setEnabled(false); 9536 effect->setSuspended(true); 9537 } 9538 } else { 9539 if (index < 0) { 9540 return; 9541 } 9542 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9543 effect->desc().type.timeLow); 9544 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9545 desc->mEffect.clear(); 9546 effect->setSuspended(false); 9547 } 9548} 9549 9550#undef LOG_TAG 9551#define LOG_TAG "AudioFlinger" 9552 9553// ---------------------------------------------------------------------------- 9554 9555status_t AudioFlinger::onTransact( 9556 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9557{ 9558 return BnAudioFlinger::onTransact(code, data, reply, flags); 9559} 9560 9561}; // namespace android 9562