AudioFlinger.cpp revision a189a6883ee55cf62da1d7bf5bf5a8ab501938a4
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// Priorities for requestPriority 165static const int kPriorityAudioApp = 2; 166static const int kPriorityFastMixer = 3; 167 168// ---------------------------------------------------------------------------- 169 170#ifdef ADD_BATTERY_DATA 171// To collect the amplifier usage 172static void addBatteryData(uint32_t params) { 173 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 174 if (service == NULL) { 175 // it already logged 176 return; 177 } 178 179 service->addBatteryData(params); 180} 181#endif 182 183static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 184{ 185 const hw_module_t *mod; 186 int rc; 187 188 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 189 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 190 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 191 if (rc) { 192 goto out; 193 } 194 rc = audio_hw_device_open(mod, dev); 195 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 196 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 197 if (rc) { 198 goto out; 199 } 200 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 201 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 202 rc = BAD_VALUE; 203 goto out; 204 } 205 return 0; 206 207out: 208 *dev = NULL; 209 return rc; 210} 211 212// ---------------------------------------------------------------------------- 213 214AudioFlinger::AudioFlinger() 215 : BnAudioFlinger(), 216 mPrimaryHardwareDev(NULL), 217 mHardwareStatus(AUDIO_HW_IDLE), 218 mMasterVolume(1.0f), 219 mMasterVolumeSW(1.0f), 220 mMasterVolumeSupportLvl(MVS_NONE), 221 mMasterMute(false), 222 mNextUniqueId(1), 223 mMode(AUDIO_MODE_INVALID), 224 mBtNrecIsOff(false) 225{ 226} 227 228void AudioFlinger::onFirstRef() 229{ 230 int rc = 0; 231 232 Mutex::Autolock _l(mLock); 233 234 /* TODO: move all this work into an Init() function */ 235 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 236 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 237 uint32_t int_val; 238 if (1 == sscanf(val_str, "%u", &int_val)) { 239 mStandbyTimeInNsecs = milliseconds(int_val); 240 ALOGI("Using %u mSec as standby time.", int_val); 241 } else { 242 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 243 ALOGI("Using default %u mSec as standby time.", 244 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 245 } 246 } 247 248 mMode = AUDIO_MODE_NORMAL; 249} 250 251AudioFlinger::~AudioFlinger() 252{ 253 while (!mRecordThreads.isEmpty()) { 254 // closeInput() will remove first entry from mRecordThreads 255 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 256 } 257 while (!mPlaybackThreads.isEmpty()) { 258 // closeOutput() will remove first entry from mPlaybackThreads 259 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 260 } 261 262 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 263 // no mHardwareLock needed, as there are no other references to this 264 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 265 delete mAudioHwDevs.valueAt(i); 266 } 267} 268 269static const char * const audio_interfaces[] = { 270 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 271 AUDIO_HARDWARE_MODULE_ID_A2DP, 272 AUDIO_HARDWARE_MODULE_ID_USB, 273}; 274#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 275 276audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 } else { 286 // check a match for the requested module handle 287 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 288 if (audioHwdevice != NULL) { 289 return audioHwdevice->hwDevice(); 290 } 291 } 292 // then try to find a module supporting the requested device. 293 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 294 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 295 if ((dev->get_supported_devices(dev) & devices) == devices) 296 return dev; 297 } 298 299 return NULL; 300} 301 302status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 303{ 304 const size_t SIZE = 256; 305 char buffer[SIZE]; 306 String8 result; 307 308 result.append("Clients:\n"); 309 for (size_t i = 0; i < mClients.size(); ++i) { 310 sp<Client> client = mClients.valueAt(i).promote(); 311 if (client != 0) { 312 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 313 result.append(buffer); 314 } 315 } 316 317 result.append("Global session refs:\n"); 318 result.append(" session pid count\n"); 319 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 320 AudioSessionRef *r = mAudioSessionRefs[i]; 321 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 322 result.append(buffer); 323 } 324 write(fd, result.string(), result.size()); 325 return NO_ERROR; 326} 327 328 329status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 330{ 331 const size_t SIZE = 256; 332 char buffer[SIZE]; 333 String8 result; 334 hardware_call_state hardwareStatus = mHardwareStatus; 335 336 snprintf(buffer, SIZE, "Hardware status: %d\n" 337 "Standby Time mSec: %u\n", 338 hardwareStatus, 339 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 340 result.append(buffer); 341 write(fd, result.string(), result.size()); 342 return NO_ERROR; 343} 344 345status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 346{ 347 const size_t SIZE = 256; 348 char buffer[SIZE]; 349 String8 result; 350 snprintf(buffer, SIZE, "Permission Denial: " 351 "can't dump AudioFlinger from pid=%d, uid=%d\n", 352 IPCThreadState::self()->getCallingPid(), 353 IPCThreadState::self()->getCallingUid()); 354 result.append(buffer); 355 write(fd, result.string(), result.size()); 356 return NO_ERROR; 357} 358 359static bool tryLock(Mutex& mutex) 360{ 361 bool locked = false; 362 for (int i = 0; i < kDumpLockRetries; ++i) { 363 if (mutex.tryLock() == NO_ERROR) { 364 locked = true; 365 break; 366 } 367 usleep(kDumpLockSleepUs); 368 } 369 return locked; 370} 371 372status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 373{ 374 if (!dumpAllowed()) { 375 dumpPermissionDenial(fd, args); 376 } else { 377 // get state of hardware lock 378 bool hardwareLocked = tryLock(mHardwareLock); 379 if (!hardwareLocked) { 380 String8 result(kHardwareLockedString); 381 write(fd, result.string(), result.size()); 382 } else { 383 mHardwareLock.unlock(); 384 } 385 386 bool locked = tryLock(mLock); 387 388 // failed to lock - AudioFlinger is probably deadlocked 389 if (!locked) { 390 String8 result(kDeadlockedString); 391 write(fd, result.string(), result.size()); 392 } 393 394 dumpClients(fd, args); 395 dumpInternals(fd, args); 396 397 // dump playback threads 398 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 399 mPlaybackThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump record threads 403 for (size_t i = 0; i < mRecordThreads.size(); i++) { 404 mRecordThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump all hardware devs 408 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 409 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 410 dev->dump(dev, fd); 411 } 412 if (locked) mLock.unlock(); 413 } 414 return NO_ERROR; 415} 416 417sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 418{ 419 // If pid is already in the mClients wp<> map, then use that entry 420 // (for which promote() is always != 0), otherwise create a new entry and Client. 421 sp<Client> client = mClients.valueFor(pid).promote(); 422 if (client == 0) { 423 client = new Client(this, pid); 424 mClients.add(pid, client); 425 } 426 427 return client; 428} 429 430// IAudioFlinger interface 431 432 433sp<IAudioTrack> AudioFlinger::createTrack( 434 pid_t pid, 435 audio_stream_type_t streamType, 436 uint32_t sampleRate, 437 audio_format_t format, 438 audio_channel_mask_t channelMask, 439 int frameCount, 440 IAudioFlinger::track_flags_t flags, 441 const sp<IMemory>& sharedBuffer, 442 audio_io_handle_t output, 443 pid_t tid, 444 int *sessionId, 445 status_t *status) 446{ 447 sp<PlaybackThread::Track> track; 448 sp<TrackHandle> trackHandle; 449 sp<Client> client; 450 status_t lStatus; 451 int lSessionId; 452 453 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 454 // but if someone uses binder directly they could bypass that and cause us to crash 455 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 456 ALOGE("createTrack() invalid stream type %d", streamType); 457 lStatus = BAD_VALUE; 458 goto Exit; 459 } 460 461 { 462 Mutex::Autolock _l(mLock); 463 PlaybackThread *thread = checkPlaybackThread_l(output); 464 PlaybackThread *effectThread = NULL; 465 if (thread == NULL) { 466 ALOGE("unknown output thread"); 467 lStatus = BAD_VALUE; 468 goto Exit; 469 } 470 471 client = registerPid_l(pid); 472 473 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 474 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 475 // check if an effect chain with the same session ID is present on another 476 // output thread and move it here. 477 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 478 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 479 if (mPlaybackThreads.keyAt(i) != output) { 480 uint32_t sessions = t->hasAudioSession(*sessionId); 481 if (sessions & PlaybackThread::EFFECT_SESSION) { 482 effectThread = t.get(); 483 break; 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 508 // Look for sync events awaiting for a session to be used. 509 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 510 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 511 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 512 if (lStatus == NO_ERROR) { 513 track->setSyncEvent(mPendingSyncEvents[i]); 514 } else { 515 mPendingSyncEvents[i]->cancel(); 516 } 517 mPendingSyncEvents.removeAt(i); 518 i--; 519 } 520 } 521 } 522 } 523 if (lStatus == NO_ERROR) { 524 trackHandle = new TrackHandle(track); 525 } else { 526 // remove local strong reference to Client before deleting the Track so that the Client 527 // destructor is called by the TrackBase destructor with mLock held 528 client.clear(); 529 track.clear(); 530 } 531 532Exit: 533 if (status != NULL) { 534 *status = lStatus; 535 } 536 return trackHandle; 537} 538 539uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 540{ 541 Mutex::Autolock _l(mLock); 542 PlaybackThread *thread = checkPlaybackThread_l(output); 543 if (thread == NULL) { 544 ALOGW("sampleRate() unknown thread %d", output); 545 return 0; 546 } 547 return thread->sampleRate(); 548} 549 550int AudioFlinger::channelCount(audio_io_handle_t output) const 551{ 552 Mutex::Autolock _l(mLock); 553 PlaybackThread *thread = checkPlaybackThread_l(output); 554 if (thread == NULL) { 555 ALOGW("channelCount() unknown thread %d", output); 556 return 0; 557 } 558 return thread->channelCount(); 559} 560 561audio_format_t AudioFlinger::format(audio_io_handle_t output) const 562{ 563 Mutex::Autolock _l(mLock); 564 PlaybackThread *thread = checkPlaybackThread_l(output); 565 if (thread == NULL) { 566 ALOGW("format() unknown thread %d", output); 567 return AUDIO_FORMAT_INVALID; 568 } 569 return thread->format(); 570} 571 572size_t AudioFlinger::frameCount(audio_io_handle_t output) const 573{ 574 Mutex::Autolock _l(mLock); 575 PlaybackThread *thread = checkPlaybackThread_l(output); 576 if (thread == NULL) { 577 ALOGW("frameCount() unknown thread %d", output); 578 return 0; 579 } 580 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 581 // should examine all callers and fix them to handle smaller counts 582 return thread->frameCount(); 583} 584 585uint32_t AudioFlinger::latency(audio_io_handle_t output) const 586{ 587 Mutex::Autolock _l(mLock); 588 PlaybackThread *thread = checkPlaybackThread_l(output); 589 if (thread == NULL) { 590 ALOGW("latency() unknown thread %d", output); 591 return 0; 592 } 593 return thread->latency(); 594} 595 596status_t AudioFlinger::setMasterVolume(float value) 597{ 598 status_t ret = initCheck(); 599 if (ret != NO_ERROR) { 600 return ret; 601 } 602 603 // check calling permissions 604 if (!settingsAllowed()) { 605 return PERMISSION_DENIED; 606 } 607 608 float swmv = value; 609 610 Mutex::Autolock _l(mLock); 611 612 // when hw supports master volume, don't scale in sw mixer 613 if (MVS_NONE != mMasterVolumeSupportLvl) { 614 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 615 AutoMutex lock(mHardwareLock); 616 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 617 618 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 619 if (NULL != dev->set_master_volume) { 620 dev->set_master_volume(dev, value); 621 } 622 mHardwareStatus = AUDIO_HW_IDLE; 623 } 624 625 swmv = 1.0; 626 } 627 628 mMasterVolume = value; 629 mMasterVolumeSW = swmv; 630 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 631 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 632 633 return NO_ERROR; 634} 635 636status_t AudioFlinger::setMode(audio_mode_t mode) 637{ 638 status_t ret = initCheck(); 639 if (ret != NO_ERROR) { 640 return ret; 641 } 642 643 // check calling permissions 644 if (!settingsAllowed()) { 645 return PERMISSION_DENIED; 646 } 647 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 648 ALOGW("Illegal value: setMode(%d)", mode); 649 return BAD_VALUE; 650 } 651 652 { // scope for the lock 653 AutoMutex lock(mHardwareLock); 654 mHardwareStatus = AUDIO_HW_SET_MODE; 655 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 656 mHardwareStatus = AUDIO_HW_IDLE; 657 } 658 659 if (NO_ERROR == ret) { 660 Mutex::Autolock _l(mLock); 661 mMode = mode; 662 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 663 mPlaybackThreads.valueAt(i)->setMode(mode); 664 } 665 666 return ret; 667} 668 669status_t AudioFlinger::setMicMute(bool state) 670{ 671 status_t ret = initCheck(); 672 if (ret != NO_ERROR) { 673 return ret; 674 } 675 676 // check calling permissions 677 if (!settingsAllowed()) { 678 return PERMISSION_DENIED; 679 } 680 681 AutoMutex lock(mHardwareLock); 682 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 683 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 684 mHardwareStatus = AUDIO_HW_IDLE; 685 return ret; 686} 687 688bool AudioFlinger::getMicMute() const 689{ 690 status_t ret = initCheck(); 691 if (ret != NO_ERROR) { 692 return false; 693 } 694 695 bool state = AUDIO_MODE_INVALID; 696 AutoMutex lock(mHardwareLock); 697 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 698 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 699 mHardwareStatus = AUDIO_HW_IDLE; 700 return state; 701} 702 703status_t AudioFlinger::setMasterMute(bool muted) 704{ 705 // check calling permissions 706 if (!settingsAllowed()) { 707 return PERMISSION_DENIED; 708 } 709 710 Mutex::Autolock _l(mLock); 711 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 712 mMasterMute = muted; 713 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 714 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 715 716 return NO_ERROR; 717} 718 719float AudioFlinger::masterVolume() const 720{ 721 Mutex::Autolock _l(mLock); 722 return masterVolume_l(); 723} 724 725float AudioFlinger::masterVolumeSW() const 726{ 727 Mutex::Autolock _l(mLock); 728 return masterVolumeSW_l(); 729} 730 731bool AudioFlinger::masterMute() const 732{ 733 Mutex::Autolock _l(mLock); 734 return masterMute_l(); 735} 736 737float AudioFlinger::masterVolume_l() const 738{ 739 if (MVS_FULL == mMasterVolumeSupportLvl) { 740 float ret_val; 741 AutoMutex lock(mHardwareLock); 742 743 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 744 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 745 (NULL != mPrimaryHardwareDev->get_master_volume), 746 "can't get master volume"); 747 748 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 749 mHardwareStatus = AUDIO_HW_IDLE; 750 return ret_val; 751 } 752 753 return mMasterVolume; 754} 755 756status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 757 audio_io_handle_t output) 758{ 759 // check calling permissions 760 if (!settingsAllowed()) { 761 return PERMISSION_DENIED; 762 } 763 764 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 765 ALOGE("setStreamVolume() invalid stream %d", stream); 766 return BAD_VALUE; 767 } 768 769 AutoMutex lock(mLock); 770 PlaybackThread *thread = NULL; 771 if (output) { 772 thread = checkPlaybackThread_l(output); 773 if (thread == NULL) { 774 return BAD_VALUE; 775 } 776 } 777 778 mStreamTypes[stream].volume = value; 779 780 if (thread == NULL) { 781 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 782 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 783 } 784 } else { 785 thread->setStreamVolume(stream, value); 786 } 787 788 return NO_ERROR; 789} 790 791status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 792{ 793 // check calling permissions 794 if (!settingsAllowed()) { 795 return PERMISSION_DENIED; 796 } 797 798 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 799 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 800 ALOGE("setStreamMute() invalid stream %d", stream); 801 return BAD_VALUE; 802 } 803 804 AutoMutex lock(mLock); 805 mStreamTypes[stream].mute = muted; 806 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 807 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 808 809 return NO_ERROR; 810} 811 812float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 813{ 814 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 815 return 0.0f; 816 } 817 818 AutoMutex lock(mLock); 819 float volume; 820 if (output) { 821 PlaybackThread *thread = checkPlaybackThread_l(output); 822 if (thread == NULL) { 823 return 0.0f; 824 } 825 volume = thread->streamVolume(stream); 826 } else { 827 volume = streamVolume_l(stream); 828 } 829 830 return volume; 831} 832 833bool AudioFlinger::streamMute(audio_stream_type_t stream) const 834{ 835 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 836 return true; 837 } 838 839 AutoMutex lock(mLock); 840 return streamMute_l(stream); 841} 842 843status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 844{ 845 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 846 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 847 // check calling permissions 848 if (!settingsAllowed()) { 849 return PERMISSION_DENIED; 850 } 851 852 // ioHandle == 0 means the parameters are global to the audio hardware interface 853 if (ioHandle == 0) { 854 Mutex::Autolock _l(mLock); 855 status_t final_result = NO_ERROR; 856 { 857 AutoMutex lock(mHardwareLock); 858 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 859 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 860 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 861 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 862 final_result = result ?: final_result; 863 } 864 mHardwareStatus = AUDIO_HW_IDLE; 865 } 866 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 867 AudioParameter param = AudioParameter(keyValuePairs); 868 String8 value; 869 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 870 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 871 if (mBtNrecIsOff != btNrecIsOff) { 872 for (size_t i = 0; i < mRecordThreads.size(); i++) { 873 sp<RecordThread> thread = mRecordThreads.valueAt(i); 874 RecordThread::RecordTrack *track = thread->track(); 875 if (track != NULL) { 876 audio_devices_t device = (audio_devices_t)( 877 thread->device() & AUDIO_DEVICE_IN_ALL); 878 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 879 thread->setEffectSuspended(FX_IID_AEC, 880 suspend, 881 track->sessionId()); 882 thread->setEffectSuspended(FX_IID_NS, 883 suspend, 884 track->sessionId()); 885 } 886 } 887 mBtNrecIsOff = btNrecIsOff; 888 } 889 } 890 String8 screenState; 891 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 892 bool isOff = screenState == "off"; 893 if (isOff != (gScreenState & 1)) { 894 gScreenState = ((gScreenState & ~1) + 2) | isOff; 895 } 896 } 897 return final_result; 898 } 899 900 // hold a strong ref on thread in case closeOutput() or closeInput() is called 901 // and the thread is exited once the lock is released 902 sp<ThreadBase> thread; 903 { 904 Mutex::Autolock _l(mLock); 905 thread = checkPlaybackThread_l(ioHandle); 906 if (thread == 0) { 907 thread = checkRecordThread_l(ioHandle); 908 } else if (thread == primaryPlaybackThread_l()) { 909 // indicate output device change to all input threads for pre processing 910 AudioParameter param = AudioParameter(keyValuePairs); 911 int value; 912 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 913 (value != 0)) { 914 for (size_t i = 0; i < mRecordThreads.size(); i++) { 915 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 916 } 917 } 918 } 919 } 920 if (thread != 0) { 921 return thread->setParameters(keyValuePairs); 922 } 923 return BAD_VALUE; 924} 925 926String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 927{ 928// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 929// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 930 931 Mutex::Autolock _l(mLock); 932 933 if (ioHandle == 0) { 934 String8 out_s8; 935 936 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 937 char *s; 938 { 939 AutoMutex lock(mHardwareLock); 940 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 941 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 942 s = dev->get_parameters(dev, keys.string()); 943 mHardwareStatus = AUDIO_HW_IDLE; 944 } 945 out_s8 += String8(s ? s : ""); 946 free(s); 947 } 948 return out_s8; 949 } 950 951 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 952 if (playbackThread != NULL) { 953 return playbackThread->getParameters(keys); 954 } 955 RecordThread *recordThread = checkRecordThread_l(ioHandle); 956 if (recordThread != NULL) { 957 return recordThread->getParameters(keys); 958 } 959 return String8(""); 960} 961 962size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 963 audio_channel_mask_t channelMask) const 964{ 965 status_t ret = initCheck(); 966 if (ret != NO_ERROR) { 967 return 0; 968 } 969 970 AutoMutex lock(mHardwareLock); 971 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 972 struct audio_config config = { 973 sample_rate: sampleRate, 974 channel_mask: channelMask, 975 format: format, 976 }; 977 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 978 mHardwareStatus = AUDIO_HW_IDLE; 979 return size; 980} 981 982unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 983{ 984 Mutex::Autolock _l(mLock); 985 986 RecordThread *recordThread = checkRecordThread_l(ioHandle); 987 if (recordThread != NULL) { 988 return recordThread->getInputFramesLost(); 989 } 990 return 0; 991} 992 993status_t AudioFlinger::setVoiceVolume(float value) 994{ 995 status_t ret = initCheck(); 996 if (ret != NO_ERROR) { 997 return ret; 998 } 999 1000 // check calling permissions 1001 if (!settingsAllowed()) { 1002 return PERMISSION_DENIED; 1003 } 1004 1005 AutoMutex lock(mHardwareLock); 1006 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1007 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1008 mHardwareStatus = AUDIO_HW_IDLE; 1009 1010 return ret; 1011} 1012 1013status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1014 audio_io_handle_t output) const 1015{ 1016 status_t status; 1017 1018 Mutex::Autolock _l(mLock); 1019 1020 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1021 if (playbackThread != NULL) { 1022 return playbackThread->getRenderPosition(halFrames, dspFrames); 1023 } 1024 1025 return BAD_VALUE; 1026} 1027 1028void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1029{ 1030 1031 Mutex::Autolock _l(mLock); 1032 1033 pid_t pid = IPCThreadState::self()->getCallingPid(); 1034 if (mNotificationClients.indexOfKey(pid) < 0) { 1035 sp<NotificationClient> notificationClient = new NotificationClient(this, 1036 client, 1037 pid); 1038 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1039 1040 mNotificationClients.add(pid, notificationClient); 1041 1042 sp<IBinder> binder = client->asBinder(); 1043 binder->linkToDeath(notificationClient); 1044 1045 // the config change is always sent from playback or record threads to avoid deadlock 1046 // with AudioSystem::gLock 1047 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1048 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1049 } 1050 1051 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1052 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1053 } 1054 } 1055} 1056 1057void AudioFlinger::removeNotificationClient(pid_t pid) 1058{ 1059 Mutex::Autolock _l(mLock); 1060 1061 mNotificationClients.removeItem(pid); 1062 1063 ALOGV("%d died, releasing its sessions", pid); 1064 size_t num = mAudioSessionRefs.size(); 1065 bool removed = false; 1066 for (size_t i = 0; i< num; ) { 1067 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1068 ALOGV(" pid %d @ %d", ref->mPid, i); 1069 if (ref->mPid == pid) { 1070 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1071 mAudioSessionRefs.removeAt(i); 1072 delete ref; 1073 removed = true; 1074 num--; 1075 } else { 1076 i++; 1077 } 1078 } 1079 if (removed) { 1080 purgeStaleEffects_l(); 1081 } 1082} 1083 1084// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1085void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1086{ 1087 size_t size = mNotificationClients.size(); 1088 for (size_t i = 0; i < size; i++) { 1089 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1090 param2); 1091 } 1092} 1093 1094// removeClient_l() must be called with AudioFlinger::mLock held 1095void AudioFlinger::removeClient_l(pid_t pid) 1096{ 1097 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1098 mClients.removeItem(pid); 1099} 1100 1101// getEffectThread_l() must be called with AudioFlinger::mLock held 1102sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1103{ 1104 sp<PlaybackThread> thread; 1105 1106 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1107 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1108 ALOG_ASSERT(thread == 0); 1109 thread = mPlaybackThreads.valueAt(i); 1110 } 1111 } 1112 1113 return thread; 1114} 1115 1116// ---------------------------------------------------------------------------- 1117 1118AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1119 uint32_t device, type_t type) 1120 : Thread(false), 1121 mType(type), 1122 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1123 // mChannelMask 1124 mChannelCount(0), 1125 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1126 mParamStatus(NO_ERROR), 1127 mStandby(false), mDevice((audio_devices_t) device), mId(id), 1128 mDeathRecipient(new PMDeathRecipient(this)) 1129{ 1130} 1131 1132AudioFlinger::ThreadBase::~ThreadBase() 1133{ 1134 mParamCond.broadcast(); 1135 // do not lock the mutex in destructor 1136 releaseWakeLock_l(); 1137 if (mPowerManager != 0) { 1138 sp<IBinder> binder = mPowerManager->asBinder(); 1139 binder->unlinkToDeath(mDeathRecipient); 1140 } 1141} 1142 1143void AudioFlinger::ThreadBase::exit() 1144{ 1145 ALOGV("ThreadBase::exit"); 1146 { 1147 // This lock prevents the following race in thread (uniprocessor for illustration): 1148 // if (!exitPending()) { 1149 // // context switch from here to exit() 1150 // // exit() calls requestExit(), what exitPending() observes 1151 // // exit() calls signal(), which is dropped since no waiters 1152 // // context switch back from exit() to here 1153 // mWaitWorkCV.wait(...); 1154 // // now thread is hung 1155 // } 1156 AutoMutex lock(mLock); 1157 requestExit(); 1158 mWaitWorkCV.signal(); 1159 } 1160 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1161 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1162 requestExitAndWait(); 1163} 1164 1165status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1166{ 1167 status_t status; 1168 1169 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1170 Mutex::Autolock _l(mLock); 1171 1172 mNewParameters.add(keyValuePairs); 1173 mWaitWorkCV.signal(); 1174 // wait condition with timeout in case the thread loop has exited 1175 // before the request could be processed 1176 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1177 status = mParamStatus; 1178 mWaitWorkCV.signal(); 1179 } else { 1180 status = TIMED_OUT; 1181 } 1182 return status; 1183} 1184 1185void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1186{ 1187 Mutex::Autolock _l(mLock); 1188 sendConfigEvent_l(event, param); 1189} 1190 1191// sendConfigEvent_l() must be called with ThreadBase::mLock held 1192void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1193{ 1194 ConfigEvent configEvent; 1195 configEvent.mEvent = event; 1196 configEvent.mParam = param; 1197 mConfigEvents.add(configEvent); 1198 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1199 mWaitWorkCV.signal(); 1200} 1201 1202void AudioFlinger::ThreadBase::processConfigEvents() 1203{ 1204 mLock.lock(); 1205 while (!mConfigEvents.isEmpty()) { 1206 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1207 ConfigEvent configEvent = mConfigEvents[0]; 1208 mConfigEvents.removeAt(0); 1209 // release mLock before locking AudioFlinger mLock: lock order is always 1210 // AudioFlinger then ThreadBase to avoid cross deadlock 1211 mLock.unlock(); 1212 mAudioFlinger->mLock.lock(); 1213 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1214 mAudioFlinger->mLock.unlock(); 1215 mLock.lock(); 1216 } 1217 mLock.unlock(); 1218} 1219 1220status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1221{ 1222 const size_t SIZE = 256; 1223 char buffer[SIZE]; 1224 String8 result; 1225 1226 bool locked = tryLock(mLock); 1227 if (!locked) { 1228 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1229 write(fd, buffer, strlen(buffer)); 1230 } 1231 1232 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1233 result.append(buffer); 1234 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1235 result.append(buffer); 1236 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1237 result.append(buffer); 1238 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1239 result.append(buffer); 1240 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1241 result.append(buffer); 1242 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1243 result.append(buffer); 1244 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1245 result.append(buffer); 1246 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1247 result.append(buffer); 1248 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1249 result.append(buffer); 1250 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1251 result.append(buffer); 1252 1253 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1254 result.append(buffer); 1255 result.append(" Index Command"); 1256 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1257 snprintf(buffer, SIZE, "\n %02d ", i); 1258 result.append(buffer); 1259 result.append(mNewParameters[i]); 1260 } 1261 1262 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1263 result.append(buffer); 1264 snprintf(buffer, SIZE, " Index event param\n"); 1265 result.append(buffer); 1266 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1267 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1268 result.append(buffer); 1269 } 1270 result.append("\n"); 1271 1272 write(fd, result.string(), result.size()); 1273 1274 if (locked) { 1275 mLock.unlock(); 1276 } 1277 return NO_ERROR; 1278} 1279 1280status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1281{ 1282 const size_t SIZE = 256; 1283 char buffer[SIZE]; 1284 String8 result; 1285 1286 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1287 write(fd, buffer, strlen(buffer)); 1288 1289 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1290 sp<EffectChain> chain = mEffectChains[i]; 1291 if (chain != 0) { 1292 chain->dump(fd, args); 1293 } 1294 } 1295 return NO_ERROR; 1296} 1297 1298void AudioFlinger::ThreadBase::acquireWakeLock() 1299{ 1300 Mutex::Autolock _l(mLock); 1301 acquireWakeLock_l(); 1302} 1303 1304void AudioFlinger::ThreadBase::acquireWakeLock_l() 1305{ 1306 if (mPowerManager == 0) { 1307 // use checkService() to avoid blocking if power service is not up yet 1308 sp<IBinder> binder = 1309 defaultServiceManager()->checkService(String16("power")); 1310 if (binder == 0) { 1311 ALOGW("Thread %s cannot connect to the power manager service", mName); 1312 } else { 1313 mPowerManager = interface_cast<IPowerManager>(binder); 1314 binder->linkToDeath(mDeathRecipient); 1315 } 1316 } 1317 if (mPowerManager != 0) { 1318 sp<IBinder> binder = new BBinder(); 1319 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1320 binder, 1321 String16(mName)); 1322 if (status == NO_ERROR) { 1323 mWakeLockToken = binder; 1324 } 1325 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1326 } 1327} 1328 1329void AudioFlinger::ThreadBase::releaseWakeLock() 1330{ 1331 Mutex::Autolock _l(mLock); 1332 releaseWakeLock_l(); 1333} 1334 1335void AudioFlinger::ThreadBase::releaseWakeLock_l() 1336{ 1337 if (mWakeLockToken != 0) { 1338 ALOGV("releaseWakeLock_l() %s", mName); 1339 if (mPowerManager != 0) { 1340 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1341 } 1342 mWakeLockToken.clear(); 1343 } 1344} 1345 1346void AudioFlinger::ThreadBase::clearPowerManager() 1347{ 1348 Mutex::Autolock _l(mLock); 1349 releaseWakeLock_l(); 1350 mPowerManager.clear(); 1351} 1352 1353void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1354{ 1355 sp<ThreadBase> thread = mThread.promote(); 1356 if (thread != 0) { 1357 thread->clearPowerManager(); 1358 } 1359 ALOGW("power manager service died !!!"); 1360} 1361 1362void AudioFlinger::ThreadBase::setEffectSuspended( 1363 const effect_uuid_t *type, bool suspend, int sessionId) 1364{ 1365 Mutex::Autolock _l(mLock); 1366 setEffectSuspended_l(type, suspend, sessionId); 1367} 1368 1369void AudioFlinger::ThreadBase::setEffectSuspended_l( 1370 const effect_uuid_t *type, bool suspend, int sessionId) 1371{ 1372 sp<EffectChain> chain = getEffectChain_l(sessionId); 1373 if (chain != 0) { 1374 if (type != NULL) { 1375 chain->setEffectSuspended_l(type, suspend); 1376 } else { 1377 chain->setEffectSuspendedAll_l(suspend); 1378 } 1379 } 1380 1381 updateSuspendedSessions_l(type, suspend, sessionId); 1382} 1383 1384void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1385{ 1386 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1387 if (index < 0) { 1388 return; 1389 } 1390 1391 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1392 mSuspendedSessions.editValueAt(index); 1393 1394 for (size_t i = 0; i < sessionEffects.size(); i++) { 1395 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1396 for (int j = 0; j < desc->mRefCount; j++) { 1397 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1398 chain->setEffectSuspendedAll_l(true); 1399 } else { 1400 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1401 desc->mType.timeLow); 1402 chain->setEffectSuspended_l(&desc->mType, true); 1403 } 1404 } 1405 } 1406} 1407 1408void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1409 bool suspend, 1410 int sessionId) 1411{ 1412 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1413 1414 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1415 1416 if (suspend) { 1417 if (index >= 0) { 1418 sessionEffects = mSuspendedSessions.editValueAt(index); 1419 } else { 1420 mSuspendedSessions.add(sessionId, sessionEffects); 1421 } 1422 } else { 1423 if (index < 0) { 1424 return; 1425 } 1426 sessionEffects = mSuspendedSessions.editValueAt(index); 1427 } 1428 1429 1430 int key = EffectChain::kKeyForSuspendAll; 1431 if (type != NULL) { 1432 key = type->timeLow; 1433 } 1434 index = sessionEffects.indexOfKey(key); 1435 1436 sp<SuspendedSessionDesc> desc; 1437 if (suspend) { 1438 if (index >= 0) { 1439 desc = sessionEffects.valueAt(index); 1440 } else { 1441 desc = new SuspendedSessionDesc(); 1442 if (type != NULL) { 1443 desc->mType = *type; 1444 } 1445 sessionEffects.add(key, desc); 1446 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1447 } 1448 desc->mRefCount++; 1449 } else { 1450 if (index < 0) { 1451 return; 1452 } 1453 desc = sessionEffects.valueAt(index); 1454 if (--desc->mRefCount == 0) { 1455 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1456 sessionEffects.removeItemsAt(index); 1457 if (sessionEffects.isEmpty()) { 1458 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1459 sessionId); 1460 mSuspendedSessions.removeItem(sessionId); 1461 } 1462 } 1463 } 1464 if (!sessionEffects.isEmpty()) { 1465 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1466 } 1467} 1468 1469void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1470 bool enabled, 1471 int sessionId) 1472{ 1473 Mutex::Autolock _l(mLock); 1474 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1475} 1476 1477void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1478 bool enabled, 1479 int sessionId) 1480{ 1481 if (mType != RECORD) { 1482 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1483 // another session. This gives the priority to well behaved effect control panels 1484 // and applications not using global effects. 1485 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1486 // global effects 1487 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1488 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1489 } 1490 } 1491 1492 sp<EffectChain> chain = getEffectChain_l(sessionId); 1493 if (chain != 0) { 1494 chain->checkSuspendOnEffectEnabled(effect, enabled); 1495 } 1496} 1497 1498// ---------------------------------------------------------------------------- 1499 1500AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1501 AudioStreamOut* output, 1502 audio_io_handle_t id, 1503 uint32_t device, 1504 type_t type) 1505 : ThreadBase(audioFlinger, id, device, type), 1506 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1507 // Assumes constructor is called by AudioFlinger with it's mLock held, 1508 // but it would be safer to explicitly pass initial masterMute as parameter 1509 mMasterMute(audioFlinger->masterMute_l()), 1510 // mStreamTypes[] initialized in constructor body 1511 mOutput(output), 1512 // Assumes constructor is called by AudioFlinger with it's mLock held, 1513 // but it would be safer to explicitly pass initial masterVolume as parameter 1514 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1515 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1516 mMixerStatus(MIXER_IDLE), 1517 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1518 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1519 mScreenState(gScreenState), 1520 // index 0 is reserved for normal mixer's submix 1521 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1522{ 1523 snprintf(mName, kNameLength, "AudioOut_%X", id); 1524 1525 readOutputParameters(); 1526 1527 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1528 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1529 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1530 stream = (audio_stream_type_t) (stream + 1)) { 1531 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1532 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1533 } 1534 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1535 // because mAudioFlinger doesn't have one to copy from 1536} 1537 1538AudioFlinger::PlaybackThread::~PlaybackThread() 1539{ 1540 delete [] mMixBuffer; 1541} 1542 1543status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1544{ 1545 dumpInternals(fd, args); 1546 dumpTracks(fd, args); 1547 dumpEffectChains(fd, args); 1548 return NO_ERROR; 1549} 1550 1551status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1552{ 1553 const size_t SIZE = 256; 1554 char buffer[SIZE]; 1555 String8 result; 1556 1557 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1558 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1559 const stream_type_t *st = &mStreamTypes[i]; 1560 if (i > 0) { 1561 result.appendFormat(", "); 1562 } 1563 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1564 if (st->mute) { 1565 result.append("M"); 1566 } 1567 } 1568 result.append("\n"); 1569 write(fd, result.string(), result.length()); 1570 result.clear(); 1571 1572 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1573 result.append(buffer); 1574 Track::appendDumpHeader(result); 1575 for (size_t i = 0; i < mTracks.size(); ++i) { 1576 sp<Track> track = mTracks[i]; 1577 if (track != 0) { 1578 track->dump(buffer, SIZE); 1579 result.append(buffer); 1580 } 1581 } 1582 1583 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1584 result.append(buffer); 1585 Track::appendDumpHeader(result); 1586 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1587 sp<Track> track = mActiveTracks[i].promote(); 1588 if (track != 0) { 1589 track->dump(buffer, SIZE); 1590 result.append(buffer); 1591 } 1592 } 1593 write(fd, result.string(), result.size()); 1594 1595 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1596 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1597 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1598 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1599 1600 return NO_ERROR; 1601} 1602 1603status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1604{ 1605 const size_t SIZE = 256; 1606 char buffer[SIZE]; 1607 String8 result; 1608 1609 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1610 result.append(buffer); 1611 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1612 result.append(buffer); 1613 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1614 result.append(buffer); 1615 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1616 result.append(buffer); 1617 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1618 result.append(buffer); 1619 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1620 result.append(buffer); 1621 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1622 result.append(buffer); 1623 write(fd, result.string(), result.size()); 1624 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1625 1626 dumpBase(fd, args); 1627 1628 return NO_ERROR; 1629} 1630 1631// Thread virtuals 1632status_t AudioFlinger::PlaybackThread::readyToRun() 1633{ 1634 status_t status = initCheck(); 1635 if (status == NO_ERROR) { 1636 ALOGI("AudioFlinger's thread %p ready to run", this); 1637 } else { 1638 ALOGE("No working audio driver found."); 1639 } 1640 return status; 1641} 1642 1643void AudioFlinger::PlaybackThread::onFirstRef() 1644{ 1645 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1646} 1647 1648// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1649sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1650 const sp<AudioFlinger::Client>& client, 1651 audio_stream_type_t streamType, 1652 uint32_t sampleRate, 1653 audio_format_t format, 1654 audio_channel_mask_t channelMask, 1655 int frameCount, 1656 const sp<IMemory>& sharedBuffer, 1657 int sessionId, 1658 IAudioFlinger::track_flags_t flags, 1659 pid_t tid, 1660 status_t *status) 1661{ 1662 sp<Track> track; 1663 status_t lStatus; 1664 1665 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1666 1667 // client expresses a preference for FAST, but we get the final say 1668 if (flags & IAudioFlinger::TRACK_FAST) { 1669 if ( 1670 // not timed 1671 (!isTimed) && 1672 // either of these use cases: 1673 ( 1674 // use case 1: shared buffer with any frame count 1675 ( 1676 (sharedBuffer != 0) 1677 ) || 1678 // use case 2: callback handler and frame count is default or at least as large as HAL 1679 ( 1680 (tid != -1) && 1681 ((frameCount == 0) || 1682 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1683 ) 1684 ) && 1685 // PCM data 1686 audio_is_linear_pcm(format) && 1687 // mono or stereo 1688 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1689 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1690#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1691 // hardware sample rate 1692 (sampleRate == mSampleRate) && 1693#endif 1694 // normal mixer has an associated fast mixer 1695 hasFastMixer() && 1696 // there are sufficient fast track slots available 1697 (mFastTrackAvailMask != 0) 1698 // FIXME test that MixerThread for this fast track has a capable output HAL 1699 // FIXME add a permission test also? 1700 ) { 1701 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1702 if (frameCount == 0) { 1703 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1704 } 1705 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1706 frameCount, mFrameCount); 1707 } else { 1708 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1709 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1710 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1711 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1712 audio_is_linear_pcm(format), 1713 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1714 flags &= ~IAudioFlinger::TRACK_FAST; 1715 // For compatibility with AudioTrack calculation, buffer depth is forced 1716 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1717 // This is probably too conservative, but legacy application code may depend on it. 1718 // If you change this calculation, also review the start threshold which is related. 1719 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1720 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1721 if (minBufCount < 2) { 1722 minBufCount = 2; 1723 } 1724 int minFrameCount = mNormalFrameCount * minBufCount; 1725 if (frameCount < minFrameCount) { 1726 frameCount = minFrameCount; 1727 } 1728 } 1729 } 1730 1731 if (mType == DIRECT) { 1732 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1733 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1734 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1735 "for output %p with format %d", 1736 sampleRate, format, channelMask, mOutput, mFormat); 1737 lStatus = BAD_VALUE; 1738 goto Exit; 1739 } 1740 } 1741 } else { 1742 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1743 if (sampleRate > mSampleRate*2) { 1744 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1745 lStatus = BAD_VALUE; 1746 goto Exit; 1747 } 1748 } 1749 1750 lStatus = initCheck(); 1751 if (lStatus != NO_ERROR) { 1752 ALOGE("Audio driver not initialized."); 1753 goto Exit; 1754 } 1755 1756 { // scope for mLock 1757 Mutex::Autolock _l(mLock); 1758 1759 // all tracks in same audio session must share the same routing strategy otherwise 1760 // conflicts will happen when tracks are moved from one output to another by audio policy 1761 // manager 1762 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1763 for (size_t i = 0; i < mTracks.size(); ++i) { 1764 sp<Track> t = mTracks[i]; 1765 if (t != 0 && !t->isOutputTrack()) { 1766 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1767 if (sessionId == t->sessionId() && strategy != actual) { 1768 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1769 strategy, actual); 1770 lStatus = BAD_VALUE; 1771 goto Exit; 1772 } 1773 } 1774 } 1775 1776 if (!isTimed) { 1777 track = new Track(this, client, streamType, sampleRate, format, 1778 channelMask, frameCount, sharedBuffer, sessionId, flags); 1779 } else { 1780 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1781 channelMask, frameCount, sharedBuffer, sessionId); 1782 } 1783 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1784 lStatus = NO_MEMORY; 1785 goto Exit; 1786 } 1787 mTracks.add(track); 1788 1789 sp<EffectChain> chain = getEffectChain_l(sessionId); 1790 if (chain != 0) { 1791 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1792 track->setMainBuffer(chain->inBuffer()); 1793 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1794 chain->incTrackCnt(); 1795 } 1796 } 1797 1798 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1799 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1800 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1801 // so ask activity manager to do this on our behalf 1802 int err = requestPriority(callingPid, tid, kPriorityAudioApp); 1803 if (err != 0) { 1804 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1805 kPriorityAudioApp, callingPid, tid, err); 1806 } 1807 } 1808 1809 lStatus = NO_ERROR; 1810 1811Exit: 1812 if (status) { 1813 *status = lStatus; 1814 } 1815 return track; 1816} 1817 1818uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1819{ 1820 if (mFastMixer != NULL) { 1821 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1822 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1823 } 1824 return latency; 1825} 1826 1827uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1828{ 1829 return latency; 1830} 1831 1832uint32_t AudioFlinger::PlaybackThread::latency() const 1833{ 1834 Mutex::Autolock _l(mLock); 1835 return latency_l(); 1836} 1837uint32_t AudioFlinger::PlaybackThread::latency_l() const 1838{ 1839 if (initCheck() == NO_ERROR) { 1840 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1841 } else { 1842 return 0; 1843 } 1844} 1845 1846void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1847{ 1848 Mutex::Autolock _l(mLock); 1849 mMasterVolume = value; 1850} 1851 1852void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1853{ 1854 Mutex::Autolock _l(mLock); 1855 setMasterMute_l(muted); 1856} 1857 1858void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1859{ 1860 Mutex::Autolock _l(mLock); 1861 mStreamTypes[stream].volume = value; 1862} 1863 1864void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1865{ 1866 Mutex::Autolock _l(mLock); 1867 mStreamTypes[stream].mute = muted; 1868} 1869 1870float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1871{ 1872 Mutex::Autolock _l(mLock); 1873 return mStreamTypes[stream].volume; 1874} 1875 1876// addTrack_l() must be called with ThreadBase::mLock held 1877status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1878{ 1879 status_t status = ALREADY_EXISTS; 1880 1881 // set retry count for buffer fill 1882 track->mRetryCount = kMaxTrackStartupRetries; 1883 if (mActiveTracks.indexOf(track) < 0) { 1884 // the track is newly added, make sure it fills up all its 1885 // buffers before playing. This is to ensure the client will 1886 // effectively get the latency it requested. 1887 track->mFillingUpStatus = Track::FS_FILLING; 1888 track->mResetDone = false; 1889 track->mPresentationCompleteFrames = 0; 1890 mActiveTracks.add(track); 1891 if (track->mainBuffer() != mMixBuffer) { 1892 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1893 if (chain != 0) { 1894 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1895 chain->incActiveTrackCnt(); 1896 } 1897 } 1898 1899 status = NO_ERROR; 1900 } 1901 1902 ALOGV("mWaitWorkCV.broadcast"); 1903 mWaitWorkCV.broadcast(); 1904 1905 return status; 1906} 1907 1908// destroyTrack_l() must be called with ThreadBase::mLock held 1909void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1910{ 1911 track->mState = TrackBase::TERMINATED; 1912 // active tracks are removed by threadLoop() 1913 if (mActiveTracks.indexOf(track) < 0) { 1914 removeTrack_l(track); 1915 } 1916} 1917 1918void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1919{ 1920 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1921 mTracks.remove(track); 1922 deleteTrackName_l(track->name()); 1923 // redundant as track is about to be destroyed, for dumpsys only 1924 track->mName = -1; 1925 if (track->isFastTrack()) { 1926 int index = track->mFastIndex; 1927 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1928 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1929 mFastTrackAvailMask |= 1 << index; 1930 // redundant as track is about to be destroyed, for dumpsys only 1931 track->mFastIndex = -1; 1932 } 1933 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1934 if (chain != 0) { 1935 chain->decTrackCnt(); 1936 } 1937} 1938 1939String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1940{ 1941 String8 out_s8 = String8(""); 1942 char *s; 1943 1944 Mutex::Autolock _l(mLock); 1945 if (initCheck() != NO_ERROR) { 1946 return out_s8; 1947 } 1948 1949 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1950 out_s8 = String8(s); 1951 free(s); 1952 return out_s8; 1953} 1954 1955// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1956void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1957 AudioSystem::OutputDescriptor desc; 1958 void *param2 = NULL; 1959 1960 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1961 1962 switch (event) { 1963 case AudioSystem::OUTPUT_OPENED: 1964 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1965 desc.channels = mChannelMask; 1966 desc.samplingRate = mSampleRate; 1967 desc.format = mFormat; 1968 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1969 desc.latency = latency(); 1970 param2 = &desc; 1971 break; 1972 1973 case AudioSystem::STREAM_CONFIG_CHANGED: 1974 param2 = ¶m; 1975 case AudioSystem::OUTPUT_CLOSED: 1976 default: 1977 break; 1978 } 1979 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1980} 1981 1982void AudioFlinger::PlaybackThread::readOutputParameters() 1983{ 1984 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1985 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1986 mChannelCount = (uint16_t)popcount(mChannelMask); 1987 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1988 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1989 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1990 if (mFrameCount & 15) { 1991 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1992 mFrameCount); 1993 } 1994 1995 // Calculate size of normal mix buffer relative to the HAL output buffer size 1996 double multiplier = 1.0; 1997 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1998 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1999 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 2000 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 2001 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 2002 maxNormalFrameCount = maxNormalFrameCount & ~15; 2003 if (maxNormalFrameCount < minNormalFrameCount) { 2004 maxNormalFrameCount = minNormalFrameCount; 2005 } 2006 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 2007 if (multiplier <= 1.0) { 2008 multiplier = 1.0; 2009 } else if (multiplier <= 2.0) { 2010 if (2 * mFrameCount <= maxNormalFrameCount) { 2011 multiplier = 2.0; 2012 } else { 2013 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2014 } 2015 } else { 2016 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2017 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2018 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2019 // FIXME this rounding up should not be done if no HAL SRC 2020 uint32_t truncMult = (uint32_t) multiplier; 2021 if ((truncMult & 1)) { 2022 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2023 ++truncMult; 2024 } 2025 } 2026 multiplier = (double) truncMult; 2027 } 2028 } 2029 mNormalFrameCount = multiplier * mFrameCount; 2030 // round up to nearest 16 frames to satisfy AudioMixer 2031 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2032 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2033 2034 delete[] mMixBuffer; 2035 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2036 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2037 2038 // force reconfiguration of effect chains and engines to take new buffer size and audio 2039 // parameters into account 2040 // Note that mLock is not held when readOutputParameters() is called from the constructor 2041 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2042 // matter. 2043 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2044 Vector< sp<EffectChain> > effectChains = mEffectChains; 2045 for (size_t i = 0; i < effectChains.size(); i ++) { 2046 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2047 } 2048} 2049 2050 2051status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2052{ 2053 if (halFrames == NULL || dspFrames == NULL) { 2054 return BAD_VALUE; 2055 } 2056 Mutex::Autolock _l(mLock); 2057 if (initCheck() != NO_ERROR) { 2058 return INVALID_OPERATION; 2059 } 2060 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2061 2062 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2063} 2064 2065uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2066{ 2067 Mutex::Autolock _l(mLock); 2068 uint32_t result = 0; 2069 if (getEffectChain_l(sessionId) != 0) { 2070 result = EFFECT_SESSION; 2071 } 2072 2073 for (size_t i = 0; i < mTracks.size(); ++i) { 2074 sp<Track> track = mTracks[i]; 2075 if (sessionId == track->sessionId() && 2076 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2077 result |= TRACK_SESSION; 2078 break; 2079 } 2080 } 2081 2082 return result; 2083} 2084 2085uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2086{ 2087 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2088 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2089 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2090 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2091 } 2092 for (size_t i = 0; i < mTracks.size(); i++) { 2093 sp<Track> track = mTracks[i]; 2094 if (sessionId == track->sessionId() && 2095 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2096 return AudioSystem::getStrategyForStream(track->streamType()); 2097 } 2098 } 2099 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2100} 2101 2102 2103AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2104{ 2105 Mutex::Autolock _l(mLock); 2106 return mOutput; 2107} 2108 2109AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2110{ 2111 Mutex::Autolock _l(mLock); 2112 AudioStreamOut *output = mOutput; 2113 mOutput = NULL; 2114 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2115 // must push a NULL and wait for ack 2116 mOutputSink.clear(); 2117 mPipeSink.clear(); 2118 mNormalSink.clear(); 2119 return output; 2120} 2121 2122// this method must always be called either with ThreadBase mLock held or inside the thread loop 2123audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2124{ 2125 if (mOutput == NULL) { 2126 return NULL; 2127 } 2128 return &mOutput->stream->common; 2129} 2130 2131uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2132{ 2133 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2134} 2135 2136status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2137{ 2138 if (!isValidSyncEvent(event)) { 2139 return BAD_VALUE; 2140 } 2141 2142 Mutex::Autolock _l(mLock); 2143 2144 for (size_t i = 0; i < mTracks.size(); ++i) { 2145 sp<Track> track = mTracks[i]; 2146 if (event->triggerSession() == track->sessionId()) { 2147 track->setSyncEvent(event); 2148 return NO_ERROR; 2149 } 2150 } 2151 2152 return NAME_NOT_FOUND; 2153} 2154 2155bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2156{ 2157 switch (event->type()) { 2158 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2159 return true; 2160 default: 2161 break; 2162 } 2163 return false; 2164} 2165 2166void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2167{ 2168 size_t count = tracksToRemove.size(); 2169 if (CC_UNLIKELY(count)) { 2170 for (size_t i = 0 ; i < count ; i++) { 2171 const sp<Track>& track = tracksToRemove.itemAt(i); 2172 if ((track->sharedBuffer() != 0) && 2173 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2174 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2175 } 2176 } 2177 } 2178 2179} 2180 2181// ---------------------------------------------------------------------------- 2182 2183AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2184 audio_io_handle_t id, uint32_t device, type_t type) 2185 : PlaybackThread(audioFlinger, output, id, device, type), 2186 // mAudioMixer below 2187 // mFastMixer below 2188 mFastMixerFutex(0) 2189 // mOutputSink below 2190 // mPipeSink below 2191 // mNormalSink below 2192{ 2193 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2194 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2195 "mFrameCount=%d, mNormalFrameCount=%d", 2196 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2197 mNormalFrameCount); 2198 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2199 2200 // FIXME - Current mixer implementation only supports stereo output 2201 if (mChannelCount != FCC_2) { 2202 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2203 } 2204 2205 // create an NBAIO sink for the HAL output stream, and negotiate 2206 mOutputSink = new AudioStreamOutSink(output->stream); 2207 size_t numCounterOffers = 0; 2208 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2209 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2210 ALOG_ASSERT(index == 0); 2211 2212 // initialize fast mixer depending on configuration 2213 bool initFastMixer; 2214 switch (kUseFastMixer) { 2215 case FastMixer_Never: 2216 initFastMixer = false; 2217 break; 2218 case FastMixer_Always: 2219 initFastMixer = true; 2220 break; 2221 case FastMixer_Static: 2222 case FastMixer_Dynamic: 2223 initFastMixer = mFrameCount < mNormalFrameCount; 2224 break; 2225 } 2226 if (initFastMixer) { 2227 2228 // create a MonoPipe to connect our submix to FastMixer 2229 NBAIO_Format format = mOutputSink->format(); 2230 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2231 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2232 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2233 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2234 const NBAIO_Format offers[1] = {format}; 2235 size_t numCounterOffers = 0; 2236 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2237 ALOG_ASSERT(index == 0); 2238 monoPipe->setAvgFrames((mScreenState & 1) ? 2239 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2240 mPipeSink = monoPipe; 2241 2242#ifdef TEE_SINK_FRAMES 2243 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2244 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2245 numCounterOffers = 0; 2246 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2247 ALOG_ASSERT(index == 0); 2248 mTeeSink = teeSink; 2249 PipeReader *teeSource = new PipeReader(*teeSink); 2250 numCounterOffers = 0; 2251 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2252 ALOG_ASSERT(index == 0); 2253 mTeeSource = teeSource; 2254#endif 2255 2256 // create fast mixer and configure it initially with just one fast track for our submix 2257 mFastMixer = new FastMixer(); 2258 FastMixerStateQueue *sq = mFastMixer->sq(); 2259#ifdef STATE_QUEUE_DUMP 2260 sq->setObserverDump(&mStateQueueObserverDump); 2261 sq->setMutatorDump(&mStateQueueMutatorDump); 2262#endif 2263 FastMixerState *state = sq->begin(); 2264 FastTrack *fastTrack = &state->mFastTracks[0]; 2265 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2266 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2267 fastTrack->mVolumeProvider = NULL; 2268 fastTrack->mGeneration++; 2269 state->mFastTracksGen++; 2270 state->mTrackMask = 1; 2271 // fast mixer will use the HAL output sink 2272 state->mOutputSink = mOutputSink.get(); 2273 state->mOutputSinkGen++; 2274 state->mFrameCount = mFrameCount; 2275 state->mCommand = FastMixerState::COLD_IDLE; 2276 // already done in constructor initialization list 2277 //mFastMixerFutex = 0; 2278 state->mColdFutexAddr = &mFastMixerFutex; 2279 state->mColdGen++; 2280 state->mDumpState = &mFastMixerDumpState; 2281 state->mTeeSink = mTeeSink.get(); 2282 sq->end(); 2283 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2284 2285 // start the fast mixer 2286 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2287 pid_t tid = mFastMixer->getTid(); 2288 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2289 if (err != 0) { 2290 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2291 kPriorityFastMixer, getpid_cached, tid, err); 2292 } 2293 2294#ifdef AUDIO_WATCHDOG 2295 // create and start the watchdog 2296 mAudioWatchdog = new AudioWatchdog(); 2297 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2298 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2299 tid = mAudioWatchdog->getTid(); 2300 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2301 if (err != 0) { 2302 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2303 kPriorityFastMixer, getpid_cached, tid, err); 2304 } 2305#endif 2306 2307 } else { 2308 mFastMixer = NULL; 2309 } 2310 2311 switch (kUseFastMixer) { 2312 case FastMixer_Never: 2313 case FastMixer_Dynamic: 2314 mNormalSink = mOutputSink; 2315 break; 2316 case FastMixer_Always: 2317 mNormalSink = mPipeSink; 2318 break; 2319 case FastMixer_Static: 2320 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2321 break; 2322 } 2323} 2324 2325AudioFlinger::MixerThread::~MixerThread() 2326{ 2327 if (mFastMixer != NULL) { 2328 FastMixerStateQueue *sq = mFastMixer->sq(); 2329 FastMixerState *state = sq->begin(); 2330 if (state->mCommand == FastMixerState::COLD_IDLE) { 2331 int32_t old = android_atomic_inc(&mFastMixerFutex); 2332 if (old == -1) { 2333 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2334 } 2335 } 2336 state->mCommand = FastMixerState::EXIT; 2337 sq->end(); 2338 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2339 mFastMixer->join(); 2340 // Though the fast mixer thread has exited, it's state queue is still valid. 2341 // We'll use that extract the final state which contains one remaining fast track 2342 // corresponding to our sub-mix. 2343 state = sq->begin(); 2344 ALOG_ASSERT(state->mTrackMask == 1); 2345 FastTrack *fastTrack = &state->mFastTracks[0]; 2346 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2347 delete fastTrack->mBufferProvider; 2348 sq->end(false /*didModify*/); 2349 delete mFastMixer; 2350 if (mAudioWatchdog != 0) { 2351 mAudioWatchdog->requestExit(); 2352 mAudioWatchdog->requestExitAndWait(); 2353 mAudioWatchdog.clear(); 2354 } 2355 } 2356 delete mAudioMixer; 2357} 2358 2359class CpuStats { 2360public: 2361 CpuStats(); 2362 void sample(const String8 &title); 2363#ifdef DEBUG_CPU_USAGE 2364private: 2365 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2366 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2367 2368 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2369 2370 int mCpuNum; // thread's current CPU number 2371 int mCpukHz; // frequency of thread's current CPU in kHz 2372#endif 2373}; 2374 2375CpuStats::CpuStats() 2376#ifdef DEBUG_CPU_USAGE 2377 : mCpuNum(-1), mCpukHz(-1) 2378#endif 2379{ 2380} 2381 2382void CpuStats::sample(const String8 &title) { 2383#ifdef DEBUG_CPU_USAGE 2384 // get current thread's delta CPU time in wall clock ns 2385 double wcNs; 2386 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2387 2388 // record sample for wall clock statistics 2389 if (valid) { 2390 mWcStats.sample(wcNs); 2391 } 2392 2393 // get the current CPU number 2394 int cpuNum = sched_getcpu(); 2395 2396 // get the current CPU frequency in kHz 2397 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2398 2399 // check if either CPU number or frequency changed 2400 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2401 mCpuNum = cpuNum; 2402 mCpukHz = cpukHz; 2403 // ignore sample for purposes of cycles 2404 valid = false; 2405 } 2406 2407 // if no change in CPU number or frequency, then record sample for cycle statistics 2408 if (valid && mCpukHz > 0) { 2409 double cycles = wcNs * cpukHz * 0.000001; 2410 mHzStats.sample(cycles); 2411 } 2412 2413 unsigned n = mWcStats.n(); 2414 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2415 if ((n & 127) == 1) { 2416 long long elapsed = mCpuUsage.elapsed(); 2417 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2418 double perLoop = elapsed / (double) n; 2419 double perLoop100 = perLoop * 0.01; 2420 double perLoop1k = perLoop * 0.001; 2421 double mean = mWcStats.mean(); 2422 double stddev = mWcStats.stddev(); 2423 double minimum = mWcStats.minimum(); 2424 double maximum = mWcStats.maximum(); 2425 double meanCycles = mHzStats.mean(); 2426 double stddevCycles = mHzStats.stddev(); 2427 double minCycles = mHzStats.minimum(); 2428 double maxCycles = mHzStats.maximum(); 2429 mCpuUsage.resetElapsed(); 2430 mWcStats.reset(); 2431 mHzStats.reset(); 2432 ALOGD("CPU usage for %s over past %.1f secs\n" 2433 " (%u mixer loops at %.1f mean ms per loop):\n" 2434 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2435 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2436 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2437 title.string(), 2438 elapsed * .000000001, n, perLoop * .000001, 2439 mean * .001, 2440 stddev * .001, 2441 minimum * .001, 2442 maximum * .001, 2443 mean / perLoop100, 2444 stddev / perLoop100, 2445 minimum / perLoop100, 2446 maximum / perLoop100, 2447 meanCycles / perLoop1k, 2448 stddevCycles / perLoop1k, 2449 minCycles / perLoop1k, 2450 maxCycles / perLoop1k); 2451 2452 } 2453 } 2454#endif 2455}; 2456 2457void AudioFlinger::PlaybackThread::checkSilentMode_l() 2458{ 2459 if (!mMasterMute) { 2460 char value[PROPERTY_VALUE_MAX]; 2461 if (property_get("ro.audio.silent", value, "0") > 0) { 2462 char *endptr; 2463 unsigned long ul = strtoul(value, &endptr, 0); 2464 if (*endptr == '\0' && ul != 0) { 2465 ALOGD("Silence is golden"); 2466 // The setprop command will not allow a property to be changed after 2467 // the first time it is set, so we don't have to worry about un-muting. 2468 setMasterMute_l(true); 2469 } 2470 } 2471 } 2472} 2473 2474bool AudioFlinger::PlaybackThread::threadLoop() 2475{ 2476 Vector< sp<Track> > tracksToRemove; 2477 2478 standbyTime = systemTime(); 2479 2480 // MIXER 2481 nsecs_t lastWarning = 0; 2482 2483 // DUPLICATING 2484 // FIXME could this be made local to while loop? 2485 writeFrames = 0; 2486 2487 cacheParameters_l(); 2488 sleepTime = idleSleepTime; 2489 2490 if (mType == MIXER) { 2491 sleepTimeShift = 0; 2492 } 2493 2494 CpuStats cpuStats; 2495 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2496 2497 acquireWakeLock(); 2498 2499 while (!exitPending()) 2500 { 2501 cpuStats.sample(myName); 2502 2503 Vector< sp<EffectChain> > effectChains; 2504 2505 processConfigEvents(); 2506 2507 { // scope for mLock 2508 2509 Mutex::Autolock _l(mLock); 2510 2511 if (checkForNewParameters_l()) { 2512 cacheParameters_l(); 2513 } 2514 2515 saveOutputTracks(); 2516 2517 // put audio hardware into standby after short delay 2518 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2519 isSuspended())) { 2520 if (!mStandby) { 2521 2522 threadLoop_standby(); 2523 2524 mStandby = true; 2525 mBytesWritten = 0; 2526 } 2527 2528 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2529 // we're about to wait, flush the binder command buffer 2530 IPCThreadState::self()->flushCommands(); 2531 2532 clearOutputTracks(); 2533 2534 if (exitPending()) break; 2535 2536 releaseWakeLock_l(); 2537 // wait until we have something to do... 2538 ALOGV("%s going to sleep", myName.string()); 2539 mWaitWorkCV.wait(mLock); 2540 ALOGV("%s waking up", myName.string()); 2541 acquireWakeLock_l(); 2542 2543 mMixerStatus = MIXER_IDLE; 2544 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2545 2546 checkSilentMode_l(); 2547 2548 standbyTime = systemTime() + standbyDelay; 2549 sleepTime = idleSleepTime; 2550 if (mType == MIXER) { 2551 sleepTimeShift = 0; 2552 } 2553 2554 continue; 2555 } 2556 } 2557 2558 // mMixerStatusIgnoringFastTracks is also updated internally 2559 mMixerStatus = prepareTracks_l(&tracksToRemove); 2560 2561 // prevent any changes in effect chain list and in each effect chain 2562 // during mixing and effect process as the audio buffers could be deleted 2563 // or modified if an effect is created or deleted 2564 lockEffectChains_l(effectChains); 2565 } 2566 2567 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2568 threadLoop_mix(); 2569 } else { 2570 threadLoop_sleepTime(); 2571 } 2572 2573 if (isSuspended()) { 2574 sleepTime = suspendSleepTimeUs(); 2575 } 2576 2577 // only process effects if we're going to write 2578 if (sleepTime == 0) { 2579 for (size_t i = 0; i < effectChains.size(); i ++) { 2580 effectChains[i]->process_l(); 2581 } 2582 } 2583 2584 // enable changes in effect chain 2585 unlockEffectChains(effectChains); 2586 2587 // sleepTime == 0 means we must write to audio hardware 2588 if (sleepTime == 0) { 2589 2590 threadLoop_write(); 2591 2592if (mType == MIXER) { 2593 // write blocked detection 2594 nsecs_t now = systemTime(); 2595 nsecs_t delta = now - mLastWriteTime; 2596 if (!mStandby && delta > maxPeriod) { 2597 mNumDelayedWrites++; 2598 if ((now - lastWarning) > kWarningThrottleNs) { 2599#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2600 ScopedTrace st(ATRACE_TAG, "underrun"); 2601#endif 2602 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2603 ns2ms(delta), mNumDelayedWrites, this); 2604 lastWarning = now; 2605 } 2606 } 2607} 2608 2609 mStandby = false; 2610 } else { 2611 usleep(sleepTime); 2612 } 2613 2614 // Finally let go of removed track(s), without the lock held 2615 // since we can't guarantee the destructors won't acquire that 2616 // same lock. This will also mutate and push a new fast mixer state. 2617 threadLoop_removeTracks(tracksToRemove); 2618 tracksToRemove.clear(); 2619 2620 // FIXME I don't understand the need for this here; 2621 // it was in the original code but maybe the 2622 // assignment in saveOutputTracks() makes this unnecessary? 2623 clearOutputTracks(); 2624 2625 // Effect chains will be actually deleted here if they were removed from 2626 // mEffectChains list during mixing or effects processing 2627 effectChains.clear(); 2628 2629 // FIXME Note that the above .clear() is no longer necessary since effectChains 2630 // is now local to this block, but will keep it for now (at least until merge done). 2631 } 2632 2633 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2634 if (mType == MIXER || mType == DIRECT) { 2635 // put output stream into standby mode 2636 if (!mStandby) { 2637 mOutput->stream->common.standby(&mOutput->stream->common); 2638 } 2639 } 2640 2641 releaseWakeLock(); 2642 2643 ALOGV("Thread %p type %d exiting", this, mType); 2644 return false; 2645} 2646 2647void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2648{ 2649 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2650} 2651 2652void AudioFlinger::MixerThread::threadLoop_write() 2653{ 2654 // FIXME we should only do one push per cycle; confirm this is true 2655 // Start the fast mixer if it's not already running 2656 if (mFastMixer != NULL) { 2657 FastMixerStateQueue *sq = mFastMixer->sq(); 2658 FastMixerState *state = sq->begin(); 2659 if (state->mCommand != FastMixerState::MIX_WRITE && 2660 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2661 if (state->mCommand == FastMixerState::COLD_IDLE) { 2662 int32_t old = android_atomic_inc(&mFastMixerFutex); 2663 if (old == -1) { 2664 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2665 } 2666 if (mAudioWatchdog != 0) { 2667 mAudioWatchdog->resume(); 2668 } 2669 } 2670 state->mCommand = FastMixerState::MIX_WRITE; 2671 sq->end(); 2672 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2673 if (kUseFastMixer == FastMixer_Dynamic) { 2674 mNormalSink = mPipeSink; 2675 } 2676 } else { 2677 sq->end(false /*didModify*/); 2678 } 2679 } 2680 PlaybackThread::threadLoop_write(); 2681} 2682 2683// shared by MIXER and DIRECT, overridden by DUPLICATING 2684void AudioFlinger::PlaybackThread::threadLoop_write() 2685{ 2686 // FIXME rewrite to reduce number of system calls 2687 mLastWriteTime = systemTime(); 2688 mInWrite = true; 2689 int bytesWritten; 2690 2691 // If an NBAIO sink is present, use it to write the normal mixer's submix 2692 if (mNormalSink != 0) { 2693#define mBitShift 2 // FIXME 2694 size_t count = mixBufferSize >> mBitShift; 2695#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2696 Tracer::traceBegin(ATRACE_TAG, "write"); 2697#endif 2698 // update the setpoint when gScreenState changes 2699 uint32_t screenState = gScreenState; 2700 if (screenState != mScreenState) { 2701 mScreenState = screenState; 2702 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2703 if (pipe != NULL) { 2704 pipe->setAvgFrames((mScreenState & 1) ? 2705 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2706 } 2707 } 2708 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2709#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2710 Tracer::traceEnd(ATRACE_TAG); 2711#endif 2712 if (framesWritten > 0) { 2713 bytesWritten = framesWritten << mBitShift; 2714 } else { 2715 bytesWritten = framesWritten; 2716 } 2717 // otherwise use the HAL / AudioStreamOut directly 2718 } else { 2719 // Direct output thread. 2720 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2721 } 2722 2723 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2724 mNumWrites++; 2725 mInWrite = false; 2726} 2727 2728void AudioFlinger::MixerThread::threadLoop_standby() 2729{ 2730 // Idle the fast mixer if it's currently running 2731 if (mFastMixer != NULL) { 2732 FastMixerStateQueue *sq = mFastMixer->sq(); 2733 FastMixerState *state = sq->begin(); 2734 if (!(state->mCommand & FastMixerState::IDLE)) { 2735 state->mCommand = FastMixerState::COLD_IDLE; 2736 state->mColdFutexAddr = &mFastMixerFutex; 2737 state->mColdGen++; 2738 mFastMixerFutex = 0; 2739 sq->end(); 2740 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2741 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2742 if (kUseFastMixer == FastMixer_Dynamic) { 2743 mNormalSink = mOutputSink; 2744 } 2745 if (mAudioWatchdog != 0) { 2746 mAudioWatchdog->pause(); 2747 } 2748 } else { 2749 sq->end(false /*didModify*/); 2750 } 2751 } 2752 PlaybackThread::threadLoop_standby(); 2753} 2754 2755// shared by MIXER and DIRECT, overridden by DUPLICATING 2756void AudioFlinger::PlaybackThread::threadLoop_standby() 2757{ 2758 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2759 mOutput->stream->common.standby(&mOutput->stream->common); 2760} 2761 2762void AudioFlinger::MixerThread::threadLoop_mix() 2763{ 2764 // obtain the presentation timestamp of the next output buffer 2765 int64_t pts; 2766 status_t status = INVALID_OPERATION; 2767 2768 if (NULL != mOutput->stream->get_next_write_timestamp) { 2769 status = mOutput->stream->get_next_write_timestamp( 2770 mOutput->stream, &pts); 2771 } 2772 2773 if (status != NO_ERROR) { 2774 pts = AudioBufferProvider::kInvalidPTS; 2775 } 2776 2777 // mix buffers... 2778 mAudioMixer->process(pts); 2779 // increase sleep time progressively when application underrun condition clears. 2780 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2781 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2782 // such that we would underrun the audio HAL. 2783 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2784 sleepTimeShift--; 2785 } 2786 sleepTime = 0; 2787 standbyTime = systemTime() + standbyDelay; 2788 //TODO: delay standby when effects have a tail 2789} 2790 2791void AudioFlinger::MixerThread::threadLoop_sleepTime() 2792{ 2793 // If no tracks are ready, sleep once for the duration of an output 2794 // buffer size, then write 0s to the output 2795 if (sleepTime == 0) { 2796 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2797 sleepTime = activeSleepTime >> sleepTimeShift; 2798 if (sleepTime < kMinThreadSleepTimeUs) { 2799 sleepTime = kMinThreadSleepTimeUs; 2800 } 2801 // reduce sleep time in case of consecutive application underruns to avoid 2802 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2803 // duration we would end up writing less data than needed by the audio HAL if 2804 // the condition persists. 2805 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2806 sleepTimeShift++; 2807 } 2808 } else { 2809 sleepTime = idleSleepTime; 2810 } 2811 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2812 memset (mMixBuffer, 0, mixBufferSize); 2813 sleepTime = 0; 2814 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2815 } 2816 // TODO add standby time extension fct of effect tail 2817} 2818 2819// prepareTracks_l() must be called with ThreadBase::mLock held 2820AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2821 Vector< sp<Track> > *tracksToRemove) 2822{ 2823 2824 mixer_state mixerStatus = MIXER_IDLE; 2825 // find out which tracks need to be processed 2826 size_t count = mActiveTracks.size(); 2827 size_t mixedTracks = 0; 2828 size_t tracksWithEffect = 0; 2829 // counts only _active_ fast tracks 2830 size_t fastTracks = 0; 2831 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2832 2833 float masterVolume = mMasterVolume; 2834 bool masterMute = mMasterMute; 2835 2836 if (masterMute) { 2837 masterVolume = 0; 2838 } 2839 // Delegate master volume control to effect in output mix effect chain if needed 2840 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2841 if (chain != 0) { 2842 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2843 chain->setVolume_l(&v, &v); 2844 masterVolume = (float)((v + (1 << 23)) >> 24); 2845 chain.clear(); 2846 } 2847 2848 // prepare a new state to push 2849 FastMixerStateQueue *sq = NULL; 2850 FastMixerState *state = NULL; 2851 bool didModify = false; 2852 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2853 if (mFastMixer != NULL) { 2854 sq = mFastMixer->sq(); 2855 state = sq->begin(); 2856 } 2857 2858 for (size_t i=0 ; i<count ; i++) { 2859 sp<Track> t = mActiveTracks[i].promote(); 2860 if (t == 0) continue; 2861 2862 // this const just means the local variable doesn't change 2863 Track* const track = t.get(); 2864 2865 // process fast tracks 2866 if (track->isFastTrack()) { 2867 2868 // It's theoretically possible (though unlikely) for a fast track to be created 2869 // and then removed within the same normal mix cycle. This is not a problem, as 2870 // the track never becomes active so it's fast mixer slot is never touched. 2871 // The converse, of removing an (active) track and then creating a new track 2872 // at the identical fast mixer slot within the same normal mix cycle, 2873 // is impossible because the slot isn't marked available until the end of each cycle. 2874 int j = track->mFastIndex; 2875 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2876 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2877 FastTrack *fastTrack = &state->mFastTracks[j]; 2878 2879 // Determine whether the track is currently in underrun condition, 2880 // and whether it had a recent underrun. 2881 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2882 FastTrackUnderruns underruns = ftDump->mUnderruns; 2883 uint32_t recentFull = (underruns.mBitFields.mFull - 2884 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2885 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2886 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2887 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2888 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2889 uint32_t recentUnderruns = recentPartial + recentEmpty; 2890 track->mObservedUnderruns = underruns; 2891 // don't count underruns that occur while stopping or pausing 2892 // or stopped which can occur when flush() is called while active 2893 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2894 track->mUnderrunCount += recentUnderruns; 2895 } 2896 2897 // This is similar to the state machine for normal tracks, 2898 // with a few modifications for fast tracks. 2899 bool isActive = true; 2900 switch (track->mState) { 2901 case TrackBase::STOPPING_1: 2902 // track stays active in STOPPING_1 state until first underrun 2903 if (recentUnderruns > 0) { 2904 track->mState = TrackBase::STOPPING_2; 2905 } 2906 break; 2907 case TrackBase::PAUSING: 2908 // ramp down is not yet implemented 2909 track->setPaused(); 2910 break; 2911 case TrackBase::RESUMING: 2912 // ramp up is not yet implemented 2913 track->mState = TrackBase::ACTIVE; 2914 break; 2915 case TrackBase::ACTIVE: 2916 if (recentFull > 0 || recentPartial > 0) { 2917 // track has provided at least some frames recently: reset retry count 2918 track->mRetryCount = kMaxTrackRetries; 2919 } 2920 if (recentUnderruns == 0) { 2921 // no recent underruns: stay active 2922 break; 2923 } 2924 // there has recently been an underrun of some kind 2925 if (track->sharedBuffer() == 0) { 2926 // were any of the recent underruns "empty" (no frames available)? 2927 if (recentEmpty == 0) { 2928 // no, then ignore the partial underruns as they are allowed indefinitely 2929 break; 2930 } 2931 // there has recently been an "empty" underrun: decrement the retry counter 2932 if (--(track->mRetryCount) > 0) { 2933 break; 2934 } 2935 // indicate to client process that the track was disabled because of underrun; 2936 // it will then automatically call start() when data is available 2937 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2938 // remove from active list, but state remains ACTIVE [confusing but true] 2939 isActive = false; 2940 break; 2941 } 2942 // fall through 2943 case TrackBase::STOPPING_2: 2944 case TrackBase::PAUSED: 2945 case TrackBase::TERMINATED: 2946 case TrackBase::STOPPED: 2947 case TrackBase::FLUSHED: // flush() while active 2948 // Check for presentation complete if track is inactive 2949 // We have consumed all the buffers of this track. 2950 // This would be incomplete if we auto-paused on underrun 2951 { 2952 size_t audioHALFrames = 2953 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2954 size_t framesWritten = 2955 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2956 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2957 // track stays in active list until presentation is complete 2958 break; 2959 } 2960 } 2961 if (track->isStopping_2()) { 2962 track->mState = TrackBase::STOPPED; 2963 } 2964 if (track->isStopped()) { 2965 // Can't reset directly, as fast mixer is still polling this track 2966 // track->reset(); 2967 // So instead mark this track as needing to be reset after push with ack 2968 resetMask |= 1 << i; 2969 } 2970 isActive = false; 2971 break; 2972 case TrackBase::IDLE: 2973 default: 2974 LOG_FATAL("unexpected track state %d", track->mState); 2975 } 2976 2977 if (isActive) { 2978 // was it previously inactive? 2979 if (!(state->mTrackMask & (1 << j))) { 2980 ExtendedAudioBufferProvider *eabp = track; 2981 VolumeProvider *vp = track; 2982 fastTrack->mBufferProvider = eabp; 2983 fastTrack->mVolumeProvider = vp; 2984 fastTrack->mSampleRate = track->mSampleRate; 2985 fastTrack->mChannelMask = track->mChannelMask; 2986 fastTrack->mGeneration++; 2987 state->mTrackMask |= 1 << j; 2988 didModify = true; 2989 // no acknowledgement required for newly active tracks 2990 } 2991 // cache the combined master volume and stream type volume for fast mixer; this 2992 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2993 track->mCachedVolume = track->isMuted() ? 2994 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2995 ++fastTracks; 2996 } else { 2997 // was it previously active? 2998 if (state->mTrackMask & (1 << j)) { 2999 fastTrack->mBufferProvider = NULL; 3000 fastTrack->mGeneration++; 3001 state->mTrackMask &= ~(1 << j); 3002 didModify = true; 3003 // If any fast tracks were removed, we must wait for acknowledgement 3004 // because we're about to decrement the last sp<> on those tracks. 3005 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3006 } else { 3007 LOG_FATAL("fast track %d should have been active", j); 3008 } 3009 tracksToRemove->add(track); 3010 // Avoids a misleading display in dumpsys 3011 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3012 } 3013 continue; 3014 } 3015 3016 { // local variable scope to avoid goto warning 3017 3018 audio_track_cblk_t* cblk = track->cblk(); 3019 3020 // The first time a track is added we wait 3021 // for all its buffers to be filled before processing it 3022 int name = track->name(); 3023 // make sure that we have enough frames to mix one full buffer. 3024 // enforce this condition only once to enable draining the buffer in case the client 3025 // app does not call stop() and relies on underrun to stop: 3026 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3027 // during last round 3028 uint32_t minFrames = 1; 3029 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3030 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3031 if (t->sampleRate() == (int)mSampleRate) { 3032 minFrames = mNormalFrameCount; 3033 } else { 3034 // +1 for rounding and +1 for additional sample needed for interpolation 3035 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3036 // add frames already consumed but not yet released by the resampler 3037 // because cblk->framesReady() will include these frames 3038 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3039 // the minimum track buffer size is normally twice the number of frames necessary 3040 // to fill one buffer and the resampler should not leave more than one buffer worth 3041 // of unreleased frames after each pass, but just in case... 3042 ALOG_ASSERT(minFrames <= cblk->frameCount); 3043 } 3044 } 3045 if ((track->framesReady() >= minFrames) && track->isReady() && 3046 !track->isPaused() && !track->isTerminated()) 3047 { 3048 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3049 3050 mixedTracks++; 3051 3052 // track->mainBuffer() != mMixBuffer means there is an effect chain 3053 // connected to the track 3054 chain.clear(); 3055 if (track->mainBuffer() != mMixBuffer) { 3056 chain = getEffectChain_l(track->sessionId()); 3057 // Delegate volume control to effect in track effect chain if needed 3058 if (chain != 0) { 3059 tracksWithEffect++; 3060 } else { 3061 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3062 name, track->sessionId()); 3063 } 3064 } 3065 3066 3067 int param = AudioMixer::VOLUME; 3068 if (track->mFillingUpStatus == Track::FS_FILLED) { 3069 // no ramp for the first volume setting 3070 track->mFillingUpStatus = Track::FS_ACTIVE; 3071 if (track->mState == TrackBase::RESUMING) { 3072 track->mState = TrackBase::ACTIVE; 3073 param = AudioMixer::RAMP_VOLUME; 3074 } 3075 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3076 } else if (cblk->server != 0) { 3077 // If the track is stopped before the first frame was mixed, 3078 // do not apply ramp 3079 param = AudioMixer::RAMP_VOLUME; 3080 } 3081 3082 // compute volume for this track 3083 uint32_t vl, vr, va; 3084 if (track->isMuted() || track->isPausing() || 3085 mStreamTypes[track->streamType()].mute) { 3086 vl = vr = va = 0; 3087 if (track->isPausing()) { 3088 track->setPaused(); 3089 } 3090 } else { 3091 3092 // read original volumes with volume control 3093 float typeVolume = mStreamTypes[track->streamType()].volume; 3094 float v = masterVolume * typeVolume; 3095 uint32_t vlr = cblk->getVolumeLR(); 3096 vl = vlr & 0xFFFF; 3097 vr = vlr >> 16; 3098 // track volumes come from shared memory, so can't be trusted and must be clamped 3099 if (vl > MAX_GAIN_INT) { 3100 ALOGV("Track left volume out of range: %04X", vl); 3101 vl = MAX_GAIN_INT; 3102 } 3103 if (vr > MAX_GAIN_INT) { 3104 ALOGV("Track right volume out of range: %04X", vr); 3105 vr = MAX_GAIN_INT; 3106 } 3107 // now apply the master volume and stream type volume 3108 vl = (uint32_t)(v * vl) << 12; 3109 vr = (uint32_t)(v * vr) << 12; 3110 // assuming master volume and stream type volume each go up to 1.0, 3111 // vl and vr are now in 8.24 format 3112 3113 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3114 // send level comes from shared memory and so may be corrupt 3115 if (sendLevel > MAX_GAIN_INT) { 3116 ALOGV("Track send level out of range: %04X", sendLevel); 3117 sendLevel = MAX_GAIN_INT; 3118 } 3119 va = (uint32_t)(v * sendLevel); 3120 } 3121 // Delegate volume control to effect in track effect chain if needed 3122 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3123 // Do not ramp volume if volume is controlled by effect 3124 param = AudioMixer::VOLUME; 3125 track->mHasVolumeController = true; 3126 } else { 3127 // force no volume ramp when volume controller was just disabled or removed 3128 // from effect chain to avoid volume spike 3129 if (track->mHasVolumeController) { 3130 param = AudioMixer::VOLUME; 3131 } 3132 track->mHasVolumeController = false; 3133 } 3134 3135 // Convert volumes from 8.24 to 4.12 format 3136 // This additional clamping is needed in case chain->setVolume_l() overshot 3137 vl = (vl + (1 << 11)) >> 12; 3138 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3139 vr = (vr + (1 << 11)) >> 12; 3140 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3141 3142 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3143 3144 // XXX: these things DON'T need to be done each time 3145 mAudioMixer->setBufferProvider(name, track); 3146 mAudioMixer->enable(name); 3147 3148 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3149 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3150 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3151 mAudioMixer->setParameter( 3152 name, 3153 AudioMixer::TRACK, 3154 AudioMixer::FORMAT, (void *)track->format()); 3155 mAudioMixer->setParameter( 3156 name, 3157 AudioMixer::TRACK, 3158 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3159 mAudioMixer->setParameter( 3160 name, 3161 AudioMixer::RESAMPLE, 3162 AudioMixer::SAMPLE_RATE, 3163 (void *)(cblk->sampleRate)); 3164 mAudioMixer->setParameter( 3165 name, 3166 AudioMixer::TRACK, 3167 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3168 mAudioMixer->setParameter( 3169 name, 3170 AudioMixer::TRACK, 3171 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3172 3173 // reset retry count 3174 track->mRetryCount = kMaxTrackRetries; 3175 3176 // If one track is ready, set the mixer ready if: 3177 // - the mixer was not ready during previous round OR 3178 // - no other track is not ready 3179 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3180 mixerStatus != MIXER_TRACKS_ENABLED) { 3181 mixerStatus = MIXER_TRACKS_READY; 3182 } 3183 } else { 3184 // clear effect chain input buffer if an active track underruns to avoid sending 3185 // previous audio buffer again to effects 3186 chain = getEffectChain_l(track->sessionId()); 3187 if (chain != 0) { 3188 chain->clearInputBuffer(); 3189 } 3190 3191 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3192 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3193 track->isStopped() || track->isPaused()) { 3194 // We have consumed all the buffers of this track. 3195 // Remove it from the list of active tracks. 3196 // TODO: use actual buffer filling status instead of latency when available from 3197 // audio HAL 3198 size_t audioHALFrames = 3199 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3200 size_t framesWritten = 3201 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3202 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3203 if (track->isStopped()) { 3204 track->reset(); 3205 } 3206 tracksToRemove->add(track); 3207 } 3208 } else { 3209 track->mUnderrunCount++; 3210 // No buffers for this track. Give it a few chances to 3211 // fill a buffer, then remove it from active list. 3212 if (--(track->mRetryCount) <= 0) { 3213 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3214 tracksToRemove->add(track); 3215 // indicate to client process that the track was disabled because of underrun; 3216 // it will then automatically call start() when data is available 3217 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3218 // If one track is not ready, mark the mixer also not ready if: 3219 // - the mixer was ready during previous round OR 3220 // - no other track is ready 3221 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3222 mixerStatus != MIXER_TRACKS_READY) { 3223 mixerStatus = MIXER_TRACKS_ENABLED; 3224 } 3225 } 3226 mAudioMixer->disable(name); 3227 } 3228 3229 } // local variable scope to avoid goto warning 3230track_is_ready: ; 3231 3232 } 3233 3234 // Push the new FastMixer state if necessary 3235 bool pauseAudioWatchdog = false; 3236 if (didModify) { 3237 state->mFastTracksGen++; 3238 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3239 if (kUseFastMixer == FastMixer_Dynamic && 3240 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3241 state->mCommand = FastMixerState::COLD_IDLE; 3242 state->mColdFutexAddr = &mFastMixerFutex; 3243 state->mColdGen++; 3244 mFastMixerFutex = 0; 3245 if (kUseFastMixer == FastMixer_Dynamic) { 3246 mNormalSink = mOutputSink; 3247 } 3248 // If we go into cold idle, need to wait for acknowledgement 3249 // so that fast mixer stops doing I/O. 3250 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3251 pauseAudioWatchdog = true; 3252 } 3253 sq->end(); 3254 } 3255 if (sq != NULL) { 3256 sq->end(didModify); 3257 sq->push(block); 3258 } 3259 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3260 mAudioWatchdog->pause(); 3261 } 3262 3263 // Now perform the deferred reset on fast tracks that have stopped 3264 while (resetMask != 0) { 3265 size_t i = __builtin_ctz(resetMask); 3266 ALOG_ASSERT(i < count); 3267 resetMask &= ~(1 << i); 3268 sp<Track> t = mActiveTracks[i].promote(); 3269 if (t == 0) continue; 3270 Track* track = t.get(); 3271 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3272 track->reset(); 3273 } 3274 3275 // remove all the tracks that need to be... 3276 count = tracksToRemove->size(); 3277 if (CC_UNLIKELY(count)) { 3278 for (size_t i=0 ; i<count ; i++) { 3279 const sp<Track>& track = tracksToRemove->itemAt(i); 3280 mActiveTracks.remove(track); 3281 if (track->mainBuffer() != mMixBuffer) { 3282 chain = getEffectChain_l(track->sessionId()); 3283 if (chain != 0) { 3284 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3285 chain->decActiveTrackCnt(); 3286 } 3287 } 3288 if (track->isTerminated()) { 3289 removeTrack_l(track); 3290 } 3291 } 3292 } 3293 3294 // mix buffer must be cleared if all tracks are connected to an 3295 // effect chain as in this case the mixer will not write to 3296 // mix buffer and track effects will accumulate into it 3297 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3298 // FIXME as a performance optimization, should remember previous zero status 3299 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3300 } 3301 3302 // if any fast tracks, then status is ready 3303 mMixerStatusIgnoringFastTracks = mixerStatus; 3304 if (fastTracks > 0) { 3305 mixerStatus = MIXER_TRACKS_READY; 3306 } 3307 return mixerStatus; 3308} 3309 3310/* 3311The derived values that are cached: 3312 - mixBufferSize from frame count * frame size 3313 - activeSleepTime from activeSleepTimeUs() 3314 - idleSleepTime from idleSleepTimeUs() 3315 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3316 - maxPeriod from frame count and sample rate (MIXER only) 3317 3318The parameters that affect these derived values are: 3319 - frame count 3320 - frame size 3321 - sample rate 3322 - device type: A2DP or not 3323 - device latency 3324 - format: PCM or not 3325 - active sleep time 3326 - idle sleep time 3327*/ 3328 3329void AudioFlinger::PlaybackThread::cacheParameters_l() 3330{ 3331 mixBufferSize = mNormalFrameCount * mFrameSize; 3332 activeSleepTime = activeSleepTimeUs(); 3333 idleSleepTime = idleSleepTimeUs(); 3334} 3335 3336void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3337{ 3338 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3339 this, streamType, mTracks.size()); 3340 Mutex::Autolock _l(mLock); 3341 3342 size_t size = mTracks.size(); 3343 for (size_t i = 0; i < size; i++) { 3344 sp<Track> t = mTracks[i]; 3345 if (t->streamType() == streamType) { 3346 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3347 t->mCblk->cv.signal(); 3348 } 3349 } 3350} 3351 3352// getTrackName_l() must be called with ThreadBase::mLock held 3353int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3354{ 3355 return mAudioMixer->getTrackName(channelMask); 3356} 3357 3358// deleteTrackName_l() must be called with ThreadBase::mLock held 3359void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3360{ 3361 ALOGV("remove track (%d) and delete from mixer", name); 3362 mAudioMixer->deleteTrackName(name); 3363} 3364 3365// checkForNewParameters_l() must be called with ThreadBase::mLock held 3366bool AudioFlinger::MixerThread::checkForNewParameters_l() 3367{ 3368 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3369 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3370 bool reconfig = false; 3371 3372 while (!mNewParameters.isEmpty()) { 3373 3374 if (mFastMixer != NULL) { 3375 FastMixerStateQueue *sq = mFastMixer->sq(); 3376 FastMixerState *state = sq->begin(); 3377 if (!(state->mCommand & FastMixerState::IDLE)) { 3378 previousCommand = state->mCommand; 3379 state->mCommand = FastMixerState::HOT_IDLE; 3380 sq->end(); 3381 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3382 } else { 3383 sq->end(false /*didModify*/); 3384 } 3385 } 3386 3387 status_t status = NO_ERROR; 3388 String8 keyValuePair = mNewParameters[0]; 3389 AudioParameter param = AudioParameter(keyValuePair); 3390 int value; 3391 3392 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3393 reconfig = true; 3394 } 3395 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3396 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3397 status = BAD_VALUE; 3398 } else { 3399 reconfig = true; 3400 } 3401 } 3402 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3403 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3404 status = BAD_VALUE; 3405 } else { 3406 reconfig = true; 3407 } 3408 } 3409 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3410 // do not accept frame count changes if tracks are open as the track buffer 3411 // size depends on frame count and correct behavior would not be guaranteed 3412 // if frame count is changed after track creation 3413 if (!mTracks.isEmpty()) { 3414 status = INVALID_OPERATION; 3415 } else { 3416 reconfig = true; 3417 } 3418 } 3419 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3420#ifdef ADD_BATTERY_DATA 3421 // when changing the audio output device, call addBatteryData to notify 3422 // the change 3423 if ((int)mDevice != value) { 3424 uint32_t params = 0; 3425 // check whether speaker is on 3426 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3427 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3428 } 3429 3430 int deviceWithoutSpeaker 3431 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3432 // check if any other device (except speaker) is on 3433 if (value & deviceWithoutSpeaker ) { 3434 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3435 } 3436 3437 if (params != 0) { 3438 addBatteryData(params); 3439 } 3440 } 3441#endif 3442 3443 // forward device change to effects that have requested to be 3444 // aware of attached audio device. 3445 mDevice = (audio_devices_t) value; 3446 for (size_t i = 0; i < mEffectChains.size(); i++) { 3447 mEffectChains[i]->setDevice_l(mDevice); 3448 } 3449 } 3450 3451 if (status == NO_ERROR) { 3452 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3453 keyValuePair.string()); 3454 if (!mStandby && status == INVALID_OPERATION) { 3455 mOutput->stream->common.standby(&mOutput->stream->common); 3456 mStandby = true; 3457 mBytesWritten = 0; 3458 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3459 keyValuePair.string()); 3460 } 3461 if (status == NO_ERROR && reconfig) { 3462 delete mAudioMixer; 3463 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3464 mAudioMixer = NULL; 3465 readOutputParameters(); 3466 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3467 for (size_t i = 0; i < mTracks.size() ; i++) { 3468 int name = getTrackName_l(mTracks[i]->mChannelMask); 3469 if (name < 0) break; 3470 mTracks[i]->mName = name; 3471 // limit track sample rate to 2 x new output sample rate 3472 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3473 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3474 } 3475 } 3476 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3477 } 3478 } 3479 3480 mNewParameters.removeAt(0); 3481 3482 mParamStatus = status; 3483 mParamCond.signal(); 3484 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3485 // already timed out waiting for the status and will never signal the condition. 3486 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3487 } 3488 3489 if (!(previousCommand & FastMixerState::IDLE)) { 3490 ALOG_ASSERT(mFastMixer != NULL); 3491 FastMixerStateQueue *sq = mFastMixer->sq(); 3492 FastMixerState *state = sq->begin(); 3493 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3494 state->mCommand = previousCommand; 3495 sq->end(); 3496 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3497 } 3498 3499 return reconfig; 3500} 3501 3502status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3503{ 3504 const size_t SIZE = 256; 3505 char buffer[SIZE]; 3506 String8 result; 3507 3508 PlaybackThread::dumpInternals(fd, args); 3509 3510 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3511 result.append(buffer); 3512 write(fd, result.string(), result.size()); 3513 3514 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3515 FastMixerDumpState copy = mFastMixerDumpState; 3516 copy.dump(fd); 3517 3518#ifdef STATE_QUEUE_DUMP 3519 // Similar for state queue 3520 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3521 observerCopy.dump(fd); 3522 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3523 mutatorCopy.dump(fd); 3524#endif 3525 3526 // Write the tee output to a .wav file 3527 NBAIO_Source *teeSource = mTeeSource.get(); 3528 if (teeSource != NULL) { 3529 char teePath[64]; 3530 struct timeval tv; 3531 gettimeofday(&tv, NULL); 3532 struct tm tm; 3533 localtime_r(&tv.tv_sec, &tm); 3534 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3535 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3536 if (teeFd >= 0) { 3537 char wavHeader[44]; 3538 memcpy(wavHeader, 3539 "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", 3540 sizeof(wavHeader)); 3541 NBAIO_Format format = teeSource->format(); 3542 unsigned channelCount = Format_channelCount(format); 3543 ALOG_ASSERT(channelCount <= FCC_2); 3544 unsigned sampleRate = Format_sampleRate(format); 3545 wavHeader[22] = channelCount; // number of channels 3546 wavHeader[24] = sampleRate; // sample rate 3547 wavHeader[25] = sampleRate >> 8; 3548 wavHeader[32] = channelCount * 2; // block alignment 3549 write(teeFd, wavHeader, sizeof(wavHeader)); 3550 size_t total = 0; 3551 bool firstRead = true; 3552 for (;;) { 3553#define TEE_SINK_READ 1024 3554 short buffer[TEE_SINK_READ * FCC_2]; 3555 size_t count = TEE_SINK_READ; 3556 ssize_t actual = teeSource->read(buffer, count); 3557 bool wasFirstRead = firstRead; 3558 firstRead = false; 3559 if (actual <= 0) { 3560 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3561 continue; 3562 } 3563 break; 3564 } 3565 ALOG_ASSERT(actual <= (ssize_t)count); 3566 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3567 total += actual; 3568 } 3569 lseek(teeFd, (off_t) 4, SEEK_SET); 3570 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3571 write(teeFd, &temp, sizeof(temp)); 3572 lseek(teeFd, (off_t) 40, SEEK_SET); 3573 temp = total * channelCount * sizeof(short); 3574 write(teeFd, &temp, sizeof(temp)); 3575 close(teeFd); 3576 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3577 } else { 3578 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3579 } 3580 } 3581 3582 if (mAudioWatchdog != 0) { 3583 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3584 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3585 wdCopy.dump(fd); 3586 } 3587 3588 return NO_ERROR; 3589} 3590 3591uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3592{ 3593 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3594} 3595 3596uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3597{ 3598 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3599} 3600 3601void AudioFlinger::MixerThread::cacheParameters_l() 3602{ 3603 PlaybackThread::cacheParameters_l(); 3604 3605 // FIXME: Relaxed timing because of a certain device that can't meet latency 3606 // Should be reduced to 2x after the vendor fixes the driver issue 3607 // increase threshold again due to low power audio mode. The way this warning 3608 // threshold is calculated and its usefulness should be reconsidered anyway. 3609 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3610} 3611 3612// ---------------------------------------------------------------------------- 3613AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3614 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3615 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3616 // mLeftVolFloat, mRightVolFloat 3617{ 3618} 3619 3620AudioFlinger::DirectOutputThread::~DirectOutputThread() 3621{ 3622} 3623 3624AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3625 Vector< sp<Track> > *tracksToRemove 3626) 3627{ 3628 sp<Track> trackToRemove; 3629 3630 mixer_state mixerStatus = MIXER_IDLE; 3631 3632 // find out which tracks need to be processed 3633 if (mActiveTracks.size() != 0) { 3634 sp<Track> t = mActiveTracks[0].promote(); 3635 // The track died recently 3636 if (t == 0) return MIXER_IDLE; 3637 3638 Track* const track = t.get(); 3639 audio_track_cblk_t* cblk = track->cblk(); 3640 3641 // The first time a track is added we wait 3642 // for all its buffers to be filled before processing it 3643 uint32_t minFrames; 3644 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3645 minFrames = mNormalFrameCount; 3646 } else { 3647 minFrames = 1; 3648 } 3649 if ((track->framesReady() >= minFrames) && track->isReady() && 3650 !track->isPaused() && !track->isTerminated()) 3651 { 3652 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3653 3654 if (track->mFillingUpStatus == Track::FS_FILLED) { 3655 track->mFillingUpStatus = Track::FS_ACTIVE; 3656 mLeftVolFloat = mRightVolFloat = 0; 3657 if (track->mState == TrackBase::RESUMING) { 3658 track->mState = TrackBase::ACTIVE; 3659 } 3660 } 3661 3662 // compute volume for this track 3663 float left, right; 3664 if (track->isMuted() || mMasterMute || track->isPausing() || 3665 mStreamTypes[track->streamType()].mute) { 3666 left = right = 0; 3667 if (track->isPausing()) { 3668 track->setPaused(); 3669 } 3670 } else { 3671 float typeVolume = mStreamTypes[track->streamType()].volume; 3672 float v = mMasterVolume * typeVolume; 3673 uint32_t vlr = cblk->getVolumeLR(); 3674 float v_clamped = v * (vlr & 0xFFFF); 3675 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3676 left = v_clamped/MAX_GAIN; 3677 v_clamped = v * (vlr >> 16); 3678 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3679 right = v_clamped/MAX_GAIN; 3680 } 3681 3682 if (left != mLeftVolFloat || right != mRightVolFloat) { 3683 mLeftVolFloat = left; 3684 mRightVolFloat = right; 3685 3686 // Convert volumes from float to 8.24 3687 uint32_t vl = (uint32_t)(left * (1 << 24)); 3688 uint32_t vr = (uint32_t)(right * (1 << 24)); 3689 3690 // Delegate volume control to effect in track effect chain if needed 3691 // only one effect chain can be present on DirectOutputThread, so if 3692 // there is one, the track is connected to it 3693 if (!mEffectChains.isEmpty()) { 3694 // Do not ramp volume if volume is controlled by effect 3695 mEffectChains[0]->setVolume_l(&vl, &vr); 3696 left = (float)vl / (1 << 24); 3697 right = (float)vr / (1 << 24); 3698 } 3699 mOutput->stream->set_volume(mOutput->stream, left, right); 3700 } 3701 3702 // reset retry count 3703 track->mRetryCount = kMaxTrackRetriesDirect; 3704 mActiveTrack = t; 3705 mixerStatus = MIXER_TRACKS_READY; 3706 } else { 3707 // clear effect chain input buffer if an active track underruns to avoid sending 3708 // previous audio buffer again to effects 3709 if (!mEffectChains.isEmpty()) { 3710 mEffectChains[0]->clearInputBuffer(); 3711 } 3712 3713 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3714 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3715 track->isStopped() || track->isPaused()) { 3716 // We have consumed all the buffers of this track. 3717 // Remove it from the list of active tracks. 3718 // TODO: implement behavior for compressed audio 3719 size_t audioHALFrames = 3720 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3721 size_t framesWritten = 3722 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3723 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3724 if (track->isStopped()) { 3725 track->reset(); 3726 } 3727 trackToRemove = track; 3728 } 3729 } else { 3730 // No buffers for this track. Give it a few chances to 3731 // fill a buffer, then remove it from active list. 3732 if (--(track->mRetryCount) <= 0) { 3733 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3734 trackToRemove = track; 3735 } else { 3736 mixerStatus = MIXER_TRACKS_ENABLED; 3737 } 3738 } 3739 } 3740 } 3741 3742 // FIXME merge this with similar code for removing multiple tracks 3743 // remove all the tracks that need to be... 3744 if (CC_UNLIKELY(trackToRemove != 0)) { 3745 tracksToRemove->add(trackToRemove); 3746 mActiveTracks.remove(trackToRemove); 3747 if (!mEffectChains.isEmpty()) { 3748 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3749 trackToRemove->sessionId()); 3750 mEffectChains[0]->decActiveTrackCnt(); 3751 } 3752 if (trackToRemove->isTerminated()) { 3753 removeTrack_l(trackToRemove); 3754 } 3755 } 3756 3757 return mixerStatus; 3758} 3759 3760void AudioFlinger::DirectOutputThread::threadLoop_mix() 3761{ 3762 AudioBufferProvider::Buffer buffer; 3763 size_t frameCount = mFrameCount; 3764 int8_t *curBuf = (int8_t *)mMixBuffer; 3765 // output audio to hardware 3766 while (frameCount) { 3767 buffer.frameCount = frameCount; 3768 mActiveTrack->getNextBuffer(&buffer); 3769 if (CC_UNLIKELY(buffer.raw == NULL)) { 3770 memset(curBuf, 0, frameCount * mFrameSize); 3771 break; 3772 } 3773 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3774 frameCount -= buffer.frameCount; 3775 curBuf += buffer.frameCount * mFrameSize; 3776 mActiveTrack->releaseBuffer(&buffer); 3777 } 3778 sleepTime = 0; 3779 standbyTime = systemTime() + standbyDelay; 3780 mActiveTrack.clear(); 3781 3782} 3783 3784void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3785{ 3786 if (sleepTime == 0) { 3787 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3788 sleepTime = activeSleepTime; 3789 } else { 3790 sleepTime = idleSleepTime; 3791 } 3792 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3793 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3794 sleepTime = 0; 3795 } 3796} 3797 3798// getTrackName_l() must be called with ThreadBase::mLock held 3799int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3800{ 3801 return 0; 3802} 3803 3804// deleteTrackName_l() must be called with ThreadBase::mLock held 3805void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3806{ 3807} 3808 3809// checkForNewParameters_l() must be called with ThreadBase::mLock held 3810bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3811{ 3812 bool reconfig = false; 3813 3814 while (!mNewParameters.isEmpty()) { 3815 status_t status = NO_ERROR; 3816 String8 keyValuePair = mNewParameters[0]; 3817 AudioParameter param = AudioParameter(keyValuePair); 3818 int value; 3819 3820 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3821 // do not accept frame count changes if tracks are open as the track buffer 3822 // size depends on frame count and correct behavior would not be garantied 3823 // if frame count is changed after track creation 3824 if (!mTracks.isEmpty()) { 3825 status = INVALID_OPERATION; 3826 } else { 3827 reconfig = true; 3828 } 3829 } 3830 if (status == NO_ERROR) { 3831 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3832 keyValuePair.string()); 3833 if (!mStandby && status == INVALID_OPERATION) { 3834 mOutput->stream->common.standby(&mOutput->stream->common); 3835 mStandby = true; 3836 mBytesWritten = 0; 3837 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3838 keyValuePair.string()); 3839 } 3840 if (status == NO_ERROR && reconfig) { 3841 readOutputParameters(); 3842 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3843 } 3844 } 3845 3846 mNewParameters.removeAt(0); 3847 3848 mParamStatus = status; 3849 mParamCond.signal(); 3850 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3851 // already timed out waiting for the status and will never signal the condition. 3852 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3853 } 3854 return reconfig; 3855} 3856 3857uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3858{ 3859 uint32_t time; 3860 if (audio_is_linear_pcm(mFormat)) { 3861 time = PlaybackThread::activeSleepTimeUs(); 3862 } else { 3863 time = 10000; 3864 } 3865 return time; 3866} 3867 3868uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3869{ 3870 uint32_t time; 3871 if (audio_is_linear_pcm(mFormat)) { 3872 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3873 } else { 3874 time = 10000; 3875 } 3876 return time; 3877} 3878 3879uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3880{ 3881 uint32_t time; 3882 if (audio_is_linear_pcm(mFormat)) { 3883 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3884 } else { 3885 time = 10000; 3886 } 3887 return time; 3888} 3889 3890void AudioFlinger::DirectOutputThread::cacheParameters_l() 3891{ 3892 PlaybackThread::cacheParameters_l(); 3893 3894 // use shorter standby delay as on normal output to release 3895 // hardware resources as soon as possible 3896 standbyDelay = microseconds(activeSleepTime*2); 3897} 3898 3899// ---------------------------------------------------------------------------- 3900 3901AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3902 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3903 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3904 mWaitTimeMs(UINT_MAX) 3905{ 3906 addOutputTrack(mainThread); 3907} 3908 3909AudioFlinger::DuplicatingThread::~DuplicatingThread() 3910{ 3911 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3912 mOutputTracks[i]->destroy(); 3913 } 3914} 3915 3916void AudioFlinger::DuplicatingThread::threadLoop_mix() 3917{ 3918 // mix buffers... 3919 if (outputsReady(outputTracks)) { 3920 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3921 } else { 3922 memset(mMixBuffer, 0, mixBufferSize); 3923 } 3924 sleepTime = 0; 3925 writeFrames = mNormalFrameCount; 3926 standbyTime = systemTime() + standbyDelay; 3927} 3928 3929void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3930{ 3931 if (sleepTime == 0) { 3932 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3933 sleepTime = activeSleepTime; 3934 } else { 3935 sleepTime = idleSleepTime; 3936 } 3937 } else if (mBytesWritten != 0) { 3938 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3939 writeFrames = mNormalFrameCount; 3940 memset(mMixBuffer, 0, mixBufferSize); 3941 } else { 3942 // flush remaining overflow buffers in output tracks 3943 writeFrames = 0; 3944 } 3945 sleepTime = 0; 3946 } 3947} 3948 3949void AudioFlinger::DuplicatingThread::threadLoop_write() 3950{ 3951 for (size_t i = 0; i < outputTracks.size(); i++) { 3952 outputTracks[i]->write(mMixBuffer, writeFrames); 3953 } 3954 mBytesWritten += mixBufferSize; 3955} 3956 3957void AudioFlinger::DuplicatingThread::threadLoop_standby() 3958{ 3959 // DuplicatingThread implements standby by stopping all tracks 3960 for (size_t i = 0; i < outputTracks.size(); i++) { 3961 outputTracks[i]->stop(); 3962 } 3963} 3964 3965void AudioFlinger::DuplicatingThread::saveOutputTracks() 3966{ 3967 outputTracks = mOutputTracks; 3968} 3969 3970void AudioFlinger::DuplicatingThread::clearOutputTracks() 3971{ 3972 outputTracks.clear(); 3973} 3974 3975void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3976{ 3977 Mutex::Autolock _l(mLock); 3978 // FIXME explain this formula 3979 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3980 OutputTrack *outputTrack = new OutputTrack(thread, 3981 this, 3982 mSampleRate, 3983 mFormat, 3984 mChannelMask, 3985 frameCount); 3986 if (outputTrack->cblk() != NULL) { 3987 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3988 mOutputTracks.add(outputTrack); 3989 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3990 updateWaitTime_l(); 3991 } 3992} 3993 3994void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3995{ 3996 Mutex::Autolock _l(mLock); 3997 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3998 if (mOutputTracks[i]->thread() == thread) { 3999 mOutputTracks[i]->destroy(); 4000 mOutputTracks.removeAt(i); 4001 updateWaitTime_l(); 4002 return; 4003 } 4004 } 4005 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 4006} 4007 4008// caller must hold mLock 4009void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4010{ 4011 mWaitTimeMs = UINT_MAX; 4012 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4013 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4014 if (strong != 0) { 4015 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4016 if (waitTimeMs < mWaitTimeMs) { 4017 mWaitTimeMs = waitTimeMs; 4018 } 4019 } 4020 } 4021} 4022 4023 4024bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4025{ 4026 for (size_t i = 0; i < outputTracks.size(); i++) { 4027 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4028 if (thread == 0) { 4029 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4030 return false; 4031 } 4032 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4033 // see note at standby() declaration 4034 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4035 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4036 return false; 4037 } 4038 } 4039 return true; 4040} 4041 4042uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4043{ 4044 return (mWaitTimeMs * 1000) / 2; 4045} 4046 4047void AudioFlinger::DuplicatingThread::cacheParameters_l() 4048{ 4049 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4050 updateWaitTime_l(); 4051 4052 MixerThread::cacheParameters_l(); 4053} 4054 4055// ---------------------------------------------------------------------------- 4056 4057// TrackBase constructor must be called with AudioFlinger::mLock held 4058AudioFlinger::ThreadBase::TrackBase::TrackBase( 4059 ThreadBase *thread, 4060 const sp<Client>& client, 4061 uint32_t sampleRate, 4062 audio_format_t format, 4063 audio_channel_mask_t channelMask, 4064 int frameCount, 4065 const sp<IMemory>& sharedBuffer, 4066 int sessionId) 4067 : RefBase(), 4068 mThread(thread), 4069 mClient(client), 4070 mCblk(NULL), 4071 // mBuffer 4072 // mBufferEnd 4073 mFrameCount(0), 4074 mState(IDLE), 4075 mSampleRate(sampleRate), 4076 mFormat(format), 4077 mStepServerFailed(false), 4078 mSessionId(sessionId) 4079 // mChannelCount 4080 // mChannelMask 4081{ 4082 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4083 4084 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4085 size_t size = sizeof(audio_track_cblk_t); 4086 uint8_t channelCount = popcount(channelMask); 4087 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4088 if (sharedBuffer == 0) { 4089 size += bufferSize; 4090 } 4091 4092 if (client != NULL) { 4093 mCblkMemory = client->heap()->allocate(size); 4094 if (mCblkMemory != 0) { 4095 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4096 if (mCblk != NULL) { // construct the shared structure in-place. 4097 new(mCblk) audio_track_cblk_t(); 4098 // clear all buffers 4099 mCblk->frameCount = frameCount; 4100 mCblk->sampleRate = sampleRate; 4101// uncomment the following lines to quickly test 32-bit wraparound 4102// mCblk->user = 0xffff0000; 4103// mCblk->server = 0xffff0000; 4104// mCblk->userBase = 0xffff0000; 4105// mCblk->serverBase = 0xffff0000; 4106 mChannelCount = channelCount; 4107 mChannelMask = channelMask; 4108 if (sharedBuffer == 0) { 4109 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4110 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4111 // Force underrun condition to avoid false underrun callback until first data is 4112 // written to buffer (other flags are cleared) 4113 mCblk->flags = CBLK_UNDERRUN_ON; 4114 } else { 4115 mBuffer = sharedBuffer->pointer(); 4116 } 4117 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4118 } 4119 } else { 4120 ALOGE("not enough memory for AudioTrack size=%u", size); 4121 client->heap()->dump("AudioTrack"); 4122 return; 4123 } 4124 } else { 4125 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4126 // construct the shared structure in-place. 4127 new(mCblk) audio_track_cblk_t(); 4128 // clear all buffers 4129 mCblk->frameCount = frameCount; 4130 mCblk->sampleRate = sampleRate; 4131// uncomment the following lines to quickly test 32-bit wraparound 4132// mCblk->user = 0xffff0000; 4133// mCblk->server = 0xffff0000; 4134// mCblk->userBase = 0xffff0000; 4135// mCblk->serverBase = 0xffff0000; 4136 mChannelCount = channelCount; 4137 mChannelMask = channelMask; 4138 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4139 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4140 // Force underrun condition to avoid false underrun callback until first data is 4141 // written to buffer (other flags are cleared) 4142 mCblk->flags = CBLK_UNDERRUN_ON; 4143 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4144 } 4145} 4146 4147AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4148{ 4149 if (mCblk != NULL) { 4150 if (mClient == 0) { 4151 delete mCblk; 4152 } else { 4153 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4154 } 4155 } 4156 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4157 if (mClient != 0) { 4158 // Client destructor must run with AudioFlinger mutex locked 4159 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4160 // If the client's reference count drops to zero, the associated destructor 4161 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4162 // relying on the automatic clear() at end of scope. 4163 mClient.clear(); 4164 } 4165} 4166 4167// AudioBufferProvider interface 4168// getNextBuffer() = 0; 4169// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4170void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4171{ 4172 buffer->raw = NULL; 4173 mFrameCount = buffer->frameCount; 4174 // FIXME See note at getNextBuffer() 4175 (void) step(); // ignore return value of step() 4176 buffer->frameCount = 0; 4177} 4178 4179bool AudioFlinger::ThreadBase::TrackBase::step() { 4180 bool result; 4181 audio_track_cblk_t* cblk = this->cblk(); 4182 4183 result = cblk->stepServer(mFrameCount); 4184 if (!result) { 4185 ALOGV("stepServer failed acquiring cblk mutex"); 4186 mStepServerFailed = true; 4187 } 4188 return result; 4189} 4190 4191void AudioFlinger::ThreadBase::TrackBase::reset() { 4192 audio_track_cblk_t* cblk = this->cblk(); 4193 4194 cblk->user = 0; 4195 cblk->server = 0; 4196 cblk->userBase = 0; 4197 cblk->serverBase = 0; 4198 mStepServerFailed = false; 4199 ALOGV("TrackBase::reset"); 4200} 4201 4202int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4203 return (int)mCblk->sampleRate; 4204} 4205 4206void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4207 audio_track_cblk_t* cblk = this->cblk(); 4208 size_t frameSize = cblk->frameSize; 4209 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4210 int8_t *bufferEnd = bufferStart + frames * frameSize; 4211 4212 // Check validity of returned pointer in case the track control block would have been corrupted. 4213 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4214 "TrackBase::getBuffer buffer out of range:\n" 4215 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4216 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4217 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4218 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4219 4220 return bufferStart; 4221} 4222 4223status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4224{ 4225 mSyncEvents.add(event); 4226 return NO_ERROR; 4227} 4228 4229// ---------------------------------------------------------------------------- 4230 4231// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4232AudioFlinger::PlaybackThread::Track::Track( 4233 PlaybackThread *thread, 4234 const sp<Client>& client, 4235 audio_stream_type_t streamType, 4236 uint32_t sampleRate, 4237 audio_format_t format, 4238 audio_channel_mask_t channelMask, 4239 int frameCount, 4240 const sp<IMemory>& sharedBuffer, 4241 int sessionId, 4242 IAudioFlinger::track_flags_t flags) 4243 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4244 mMute(false), 4245 mFillingUpStatus(FS_INVALID), 4246 // mRetryCount initialized later when needed 4247 mSharedBuffer(sharedBuffer), 4248 mStreamType(streamType), 4249 mName(-1), // see note below 4250 mMainBuffer(thread->mixBuffer()), 4251 mAuxBuffer(NULL), 4252 mAuxEffectId(0), mHasVolumeController(false), 4253 mPresentationCompleteFrames(0), 4254 mFlags(flags), 4255 mFastIndex(-1), 4256 mUnderrunCount(0), 4257 mCachedVolume(1.0) 4258{ 4259 if (mCblk != NULL) { 4260 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4261 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4262 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4263 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4264 mName = thread->getTrackName_l(channelMask); 4265 mCblk->mName = mName; 4266 if (mName < 0) { 4267 ALOGE("no more track names available"); 4268 return; 4269 } 4270 // only allocate a fast track index if we were able to allocate a normal track name 4271 if (flags & IAudioFlinger::TRACK_FAST) { 4272 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4273 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4274 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4275 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4276 // FIXME This is too eager. We allocate a fast track index before the 4277 // fast track becomes active. Since fast tracks are a scarce resource, 4278 // this means we are potentially denying other more important fast tracks from 4279 // being created. It would be better to allocate the index dynamically. 4280 mFastIndex = i; 4281 mCblk->mName = i; 4282 // Read the initial underruns because this field is never cleared by the fast mixer 4283 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4284 thread->mFastTrackAvailMask &= ~(1 << i); 4285 } 4286 } 4287 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4288} 4289 4290AudioFlinger::PlaybackThread::Track::~Track() 4291{ 4292 ALOGV("PlaybackThread::Track destructor"); 4293 sp<ThreadBase> thread = mThread.promote(); 4294 if (thread != 0) { 4295 Mutex::Autolock _l(thread->mLock); 4296 mState = TERMINATED; 4297 } 4298} 4299 4300void AudioFlinger::PlaybackThread::Track::destroy() 4301{ 4302 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4303 // by removing it from mTracks vector, so there is a risk that this Tracks's 4304 // destructor is called. As the destructor needs to lock mLock, 4305 // we must acquire a strong reference on this Track before locking mLock 4306 // here so that the destructor is called only when exiting this function. 4307 // On the other hand, as long as Track::destroy() is only called by 4308 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4309 // this Track with its member mTrack. 4310 sp<Track> keep(this); 4311 { // scope for mLock 4312 sp<ThreadBase> thread = mThread.promote(); 4313 if (thread != 0) { 4314 if (!isOutputTrack()) { 4315 if (mState == ACTIVE || mState == RESUMING) { 4316 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4317 4318#ifdef ADD_BATTERY_DATA 4319 // to track the speaker usage 4320 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4321#endif 4322 } 4323 AudioSystem::releaseOutput(thread->id()); 4324 } 4325 Mutex::Autolock _l(thread->mLock); 4326 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4327 playbackThread->destroyTrack_l(this); 4328 } 4329 } 4330} 4331 4332/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4333{ 4334 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4335 " Server User Main buf Aux Buf Flags Underruns\n"); 4336} 4337 4338void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4339{ 4340 uint32_t vlr = mCblk->getVolumeLR(); 4341 if (isFastTrack()) { 4342 sprintf(buffer, " F %2d", mFastIndex); 4343 } else { 4344 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4345 } 4346 track_state state = mState; 4347 char stateChar; 4348 switch (state) { 4349 case IDLE: 4350 stateChar = 'I'; 4351 break; 4352 case TERMINATED: 4353 stateChar = 'T'; 4354 break; 4355 case STOPPING_1: 4356 stateChar = 's'; 4357 break; 4358 case STOPPING_2: 4359 stateChar = '5'; 4360 break; 4361 case STOPPED: 4362 stateChar = 'S'; 4363 break; 4364 case RESUMING: 4365 stateChar = 'R'; 4366 break; 4367 case ACTIVE: 4368 stateChar = 'A'; 4369 break; 4370 case PAUSING: 4371 stateChar = 'p'; 4372 break; 4373 case PAUSED: 4374 stateChar = 'P'; 4375 break; 4376 case FLUSHED: 4377 stateChar = 'F'; 4378 break; 4379 default: 4380 stateChar = '?'; 4381 break; 4382 } 4383 char nowInUnderrun; 4384 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4385 case UNDERRUN_FULL: 4386 nowInUnderrun = ' '; 4387 break; 4388 case UNDERRUN_PARTIAL: 4389 nowInUnderrun = '<'; 4390 break; 4391 case UNDERRUN_EMPTY: 4392 nowInUnderrun = '*'; 4393 break; 4394 default: 4395 nowInUnderrun = '?'; 4396 break; 4397 } 4398 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4399 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4400 (mClient == 0) ? getpid_cached : mClient->pid(), 4401 mStreamType, 4402 mFormat, 4403 mChannelMask, 4404 mSessionId, 4405 mFrameCount, 4406 mCblk->frameCount, 4407 stateChar, 4408 mMute, 4409 mFillingUpStatus, 4410 mCblk->sampleRate, 4411 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4412 20.0 * log10((vlr >> 16) / 4096.0), 4413 mCblk->server, 4414 mCblk->user, 4415 (int)mMainBuffer, 4416 (int)mAuxBuffer, 4417 mCblk->flags, 4418 mUnderrunCount, 4419 nowInUnderrun); 4420} 4421 4422// AudioBufferProvider interface 4423status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4424 AudioBufferProvider::Buffer* buffer, int64_t pts) 4425{ 4426 audio_track_cblk_t* cblk = this->cblk(); 4427 uint32_t framesReady; 4428 uint32_t framesReq = buffer->frameCount; 4429 4430 // Check if last stepServer failed, try to step now 4431 if (mStepServerFailed) { 4432 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4433 // Since the fast mixer is higher priority than client callback thread, 4434 // it does not result in priority inversion for client. 4435 // But a non-blocking solution would be preferable to avoid 4436 // fast mixer being unable to tryLock(), and 4437 // to avoid the extra context switches if the client wakes up, 4438 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4439 if (!step()) goto getNextBuffer_exit; 4440 ALOGV("stepServer recovered"); 4441 mStepServerFailed = false; 4442 } 4443 4444 // FIXME Same as above 4445 framesReady = cblk->framesReady(); 4446 4447 if (CC_LIKELY(framesReady)) { 4448 uint32_t s = cblk->server; 4449 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4450 4451 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4452 if (framesReq > framesReady) { 4453 framesReq = framesReady; 4454 } 4455 if (framesReq > bufferEnd - s) { 4456 framesReq = bufferEnd - s; 4457 } 4458 4459 buffer->raw = getBuffer(s, framesReq); 4460 buffer->frameCount = framesReq; 4461 return NO_ERROR; 4462 } 4463 4464getNextBuffer_exit: 4465 buffer->raw = NULL; 4466 buffer->frameCount = 0; 4467 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4468 return NOT_ENOUGH_DATA; 4469} 4470 4471// Note that framesReady() takes a mutex on the control block using tryLock(). 4472// This could result in priority inversion if framesReady() is called by the normal mixer, 4473// as the normal mixer thread runs at lower 4474// priority than the client's callback thread: there is a short window within framesReady() 4475// during which the normal mixer could be preempted, and the client callback would block. 4476// Another problem can occur if framesReady() is called by the fast mixer: 4477// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4478// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4479size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4480 return mCblk->framesReady(); 4481} 4482 4483// Don't call for fast tracks; the framesReady() could result in priority inversion 4484bool AudioFlinger::PlaybackThread::Track::isReady() const { 4485 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4486 4487 if (framesReady() >= mCblk->frameCount || 4488 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4489 mFillingUpStatus = FS_FILLED; 4490 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4491 return true; 4492 } 4493 return false; 4494} 4495 4496status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4497 int triggerSession) 4498{ 4499 status_t status = NO_ERROR; 4500 ALOGV("start(%d), calling pid %d session %d", 4501 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4502 4503 sp<ThreadBase> thread = mThread.promote(); 4504 if (thread != 0) { 4505 Mutex::Autolock _l(thread->mLock); 4506 track_state state = mState; 4507 // here the track could be either new, or restarted 4508 // in both cases "unstop" the track 4509 if (mState == PAUSED) { 4510 mState = TrackBase::RESUMING; 4511 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4512 } else { 4513 mState = TrackBase::ACTIVE; 4514 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4515 } 4516 4517 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4518 thread->mLock.unlock(); 4519 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4520 thread->mLock.lock(); 4521 4522#ifdef ADD_BATTERY_DATA 4523 // to track the speaker usage 4524 if (status == NO_ERROR) { 4525 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4526 } 4527#endif 4528 } 4529 if (status == NO_ERROR) { 4530 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4531 playbackThread->addTrack_l(this); 4532 } else { 4533 mState = state; 4534 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4535 } 4536 } else { 4537 status = BAD_VALUE; 4538 } 4539 return status; 4540} 4541 4542void AudioFlinger::PlaybackThread::Track::stop() 4543{ 4544 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4545 sp<ThreadBase> thread = mThread.promote(); 4546 if (thread != 0) { 4547 Mutex::Autolock _l(thread->mLock); 4548 track_state state = mState; 4549 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4550 // If the track is not active (PAUSED and buffers full), flush buffers 4551 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4552 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4553 reset(); 4554 mState = STOPPED; 4555 } else if (!isFastTrack()) { 4556 mState = STOPPED; 4557 } else { 4558 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4559 // and then to STOPPED and reset() when presentation is complete 4560 mState = STOPPING_1; 4561 } 4562 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4563 } 4564 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4565 thread->mLock.unlock(); 4566 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4567 thread->mLock.lock(); 4568 4569#ifdef ADD_BATTERY_DATA 4570 // to track the speaker usage 4571 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4572#endif 4573 } 4574 } 4575} 4576 4577void AudioFlinger::PlaybackThread::Track::pause() 4578{ 4579 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4580 sp<ThreadBase> thread = mThread.promote(); 4581 if (thread != 0) { 4582 Mutex::Autolock _l(thread->mLock); 4583 if (mState == ACTIVE || mState == RESUMING) { 4584 mState = PAUSING; 4585 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4586 if (!isOutputTrack()) { 4587 thread->mLock.unlock(); 4588 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4589 thread->mLock.lock(); 4590 4591#ifdef ADD_BATTERY_DATA 4592 // to track the speaker usage 4593 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4594#endif 4595 } 4596 } 4597 } 4598} 4599 4600void AudioFlinger::PlaybackThread::Track::flush() 4601{ 4602 ALOGV("flush(%d)", mName); 4603 sp<ThreadBase> thread = mThread.promote(); 4604 if (thread != 0) { 4605 Mutex::Autolock _l(thread->mLock); 4606 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4607 mState != PAUSING) { 4608 return; 4609 } 4610 // No point remaining in PAUSED state after a flush => go to 4611 // FLUSHED state 4612 mState = FLUSHED; 4613 // do not reset the track if it is still in the process of being stopped or paused. 4614 // this will be done by prepareTracks_l() when the track is stopped. 4615 // prepareTracks_l() will see mState == FLUSHED, then 4616 // remove from active track list, reset(), and trigger presentation complete 4617 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4618 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4619 reset(); 4620 } 4621 } 4622} 4623 4624void AudioFlinger::PlaybackThread::Track::reset() 4625{ 4626 // Do not reset twice to avoid discarding data written just after a flush and before 4627 // the audioflinger thread detects the track is stopped. 4628 if (!mResetDone) { 4629 TrackBase::reset(); 4630 // Force underrun condition to avoid false underrun callback until first data is 4631 // written to buffer 4632 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4633 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4634 mFillingUpStatus = FS_FILLING; 4635 mResetDone = true; 4636 if (mState == FLUSHED) { 4637 mState = IDLE; 4638 } 4639 } 4640} 4641 4642void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4643{ 4644 mMute = muted; 4645} 4646 4647status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4648{ 4649 status_t status = DEAD_OBJECT; 4650 sp<ThreadBase> thread = mThread.promote(); 4651 if (thread != 0) { 4652 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4653 sp<AudioFlinger> af = mClient->audioFlinger(); 4654 4655 Mutex::Autolock _l(af->mLock); 4656 4657 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4658 4659 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4660 Mutex::Autolock _dl(playbackThread->mLock); 4661 Mutex::Autolock _sl(srcThread->mLock); 4662 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4663 if (chain == 0) { 4664 return INVALID_OPERATION; 4665 } 4666 4667 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4668 if (effect == 0) { 4669 return INVALID_OPERATION; 4670 } 4671 srcThread->removeEffect_l(effect); 4672 playbackThread->addEffect_l(effect); 4673 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4674 if (effect->state() == EffectModule::ACTIVE || 4675 effect->state() == EffectModule::STOPPING) { 4676 effect->start(); 4677 } 4678 4679 sp<EffectChain> dstChain = effect->chain().promote(); 4680 if (dstChain == 0) { 4681 srcThread->addEffect_l(effect); 4682 return INVALID_OPERATION; 4683 } 4684 AudioSystem::unregisterEffect(effect->id()); 4685 AudioSystem::registerEffect(&effect->desc(), 4686 srcThread->id(), 4687 dstChain->strategy(), 4688 AUDIO_SESSION_OUTPUT_MIX, 4689 effect->id()); 4690 } 4691 status = playbackThread->attachAuxEffect(this, EffectId); 4692 } 4693 return status; 4694} 4695 4696void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4697{ 4698 mAuxEffectId = EffectId; 4699 mAuxBuffer = buffer; 4700} 4701 4702bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4703 size_t audioHalFrames) 4704{ 4705 // a track is considered presented when the total number of frames written to audio HAL 4706 // corresponds to the number of frames written when presentationComplete() is called for the 4707 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4708 if (mPresentationCompleteFrames == 0) { 4709 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4710 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4711 mPresentationCompleteFrames, audioHalFrames); 4712 } 4713 if (framesWritten >= mPresentationCompleteFrames) { 4714 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4715 mSessionId, framesWritten); 4716 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4717 return true; 4718 } 4719 return false; 4720} 4721 4722void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4723{ 4724 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4725 if (mSyncEvents[i]->type() == type) { 4726 mSyncEvents[i]->trigger(); 4727 mSyncEvents.removeAt(i); 4728 i--; 4729 } 4730 } 4731} 4732 4733// implement VolumeBufferProvider interface 4734 4735uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4736{ 4737 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4738 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4739 uint32_t vlr = mCblk->getVolumeLR(); 4740 uint32_t vl = vlr & 0xFFFF; 4741 uint32_t vr = vlr >> 16; 4742 // track volumes come from shared memory, so can't be trusted and must be clamped 4743 if (vl > MAX_GAIN_INT) { 4744 vl = MAX_GAIN_INT; 4745 } 4746 if (vr > MAX_GAIN_INT) { 4747 vr = MAX_GAIN_INT; 4748 } 4749 // now apply the cached master volume and stream type volume; 4750 // this is trusted but lacks any synchronization or barrier so may be stale 4751 float v = mCachedVolume; 4752 vl *= v; 4753 vr *= v; 4754 // re-combine into U4.16 4755 vlr = (vr << 16) | (vl & 0xFFFF); 4756 // FIXME look at mute, pause, and stop flags 4757 return vlr; 4758} 4759 4760status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4761{ 4762 if (mState == TERMINATED || mState == PAUSED || 4763 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4764 (mState == STOPPED)))) { 4765 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4766 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4767 event->cancel(); 4768 return INVALID_OPERATION; 4769 } 4770 TrackBase::setSyncEvent(event); 4771 return NO_ERROR; 4772} 4773 4774// timed audio tracks 4775 4776sp<AudioFlinger::PlaybackThread::TimedTrack> 4777AudioFlinger::PlaybackThread::TimedTrack::create( 4778 PlaybackThread *thread, 4779 const sp<Client>& client, 4780 audio_stream_type_t streamType, 4781 uint32_t sampleRate, 4782 audio_format_t format, 4783 audio_channel_mask_t channelMask, 4784 int frameCount, 4785 const sp<IMemory>& sharedBuffer, 4786 int sessionId) { 4787 if (!client->reserveTimedTrack()) 4788 return 0; 4789 4790 return new TimedTrack( 4791 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4792 sharedBuffer, sessionId); 4793} 4794 4795AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4796 PlaybackThread *thread, 4797 const sp<Client>& client, 4798 audio_stream_type_t streamType, 4799 uint32_t sampleRate, 4800 audio_format_t format, 4801 audio_channel_mask_t channelMask, 4802 int frameCount, 4803 const sp<IMemory>& sharedBuffer, 4804 int sessionId) 4805 : Track(thread, client, streamType, sampleRate, format, channelMask, 4806 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4807 mQueueHeadInFlight(false), 4808 mTrimQueueHeadOnRelease(false), 4809 mFramesPendingInQueue(0), 4810 mTimedSilenceBuffer(NULL), 4811 mTimedSilenceBufferSize(0), 4812 mTimedAudioOutputOnTime(false), 4813 mMediaTimeTransformValid(false) 4814{ 4815 LocalClock lc; 4816 mLocalTimeFreq = lc.getLocalFreq(); 4817 4818 mLocalTimeToSampleTransform.a_zero = 0; 4819 mLocalTimeToSampleTransform.b_zero = 0; 4820 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4821 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4822 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4823 &mLocalTimeToSampleTransform.a_to_b_denom); 4824 4825 mMediaTimeToSampleTransform.a_zero = 0; 4826 mMediaTimeToSampleTransform.b_zero = 0; 4827 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4828 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4829 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4830 &mMediaTimeToSampleTransform.a_to_b_denom); 4831} 4832 4833AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4834 mClient->releaseTimedTrack(); 4835 delete [] mTimedSilenceBuffer; 4836} 4837 4838status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4839 size_t size, sp<IMemory>* buffer) { 4840 4841 Mutex::Autolock _l(mTimedBufferQueueLock); 4842 4843 trimTimedBufferQueue_l(); 4844 4845 // lazily initialize the shared memory heap for timed buffers 4846 if (mTimedMemoryDealer == NULL) { 4847 const int kTimedBufferHeapSize = 512 << 10; 4848 4849 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4850 "AudioFlingerTimed"); 4851 if (mTimedMemoryDealer == NULL) 4852 return NO_MEMORY; 4853 } 4854 4855 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4856 if (newBuffer == NULL) { 4857 newBuffer = mTimedMemoryDealer->allocate(size); 4858 if (newBuffer == NULL) 4859 return NO_MEMORY; 4860 } 4861 4862 *buffer = newBuffer; 4863 return NO_ERROR; 4864} 4865 4866// caller must hold mTimedBufferQueueLock 4867void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4868 int64_t mediaTimeNow; 4869 { 4870 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4871 if (!mMediaTimeTransformValid) 4872 return; 4873 4874 int64_t targetTimeNow; 4875 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4876 ? mCCHelper.getCommonTime(&targetTimeNow) 4877 : mCCHelper.getLocalTime(&targetTimeNow); 4878 4879 if (OK != res) 4880 return; 4881 4882 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4883 &mediaTimeNow)) { 4884 return; 4885 } 4886 } 4887 4888 size_t trimEnd; 4889 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4890 int64_t bufEnd; 4891 4892 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4893 // We have a next buffer. Just use its PTS as the PTS of the frame 4894 // following the last frame in this buffer. If the stream is sparse 4895 // (ie, there are deliberate gaps left in the stream which should be 4896 // filled with silence by the TimedAudioTrack), then this can result 4897 // in one extra buffer being left un-trimmed when it could have 4898 // been. In general, this is not typical, and we would rather 4899 // optimized away the TS calculation below for the more common case 4900 // where PTSes are contiguous. 4901 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4902 } else { 4903 // We have no next buffer. Compute the PTS of the frame following 4904 // the last frame in this buffer by computing the duration of of 4905 // this frame in media time units and adding it to the PTS of the 4906 // buffer. 4907 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4908 / mCblk->frameSize; 4909 4910 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4911 &bufEnd)) { 4912 ALOGE("Failed to convert frame count of %lld to media time" 4913 " duration" " (scale factor %d/%u) in %s", 4914 frameCount, 4915 mMediaTimeToSampleTransform.a_to_b_numer, 4916 mMediaTimeToSampleTransform.a_to_b_denom, 4917 __PRETTY_FUNCTION__); 4918 break; 4919 } 4920 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4921 } 4922 4923 if (bufEnd > mediaTimeNow) 4924 break; 4925 4926 // Is the buffer we want to use in the middle of a mix operation right 4927 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4928 // from the mixer which should be coming back shortly. 4929 if (!trimEnd && mQueueHeadInFlight) { 4930 mTrimQueueHeadOnRelease = true; 4931 } 4932 } 4933 4934 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4935 if (trimStart < trimEnd) { 4936 // Update the bookkeeping for framesReady() 4937 for (size_t i = trimStart; i < trimEnd; ++i) { 4938 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4939 } 4940 4941 // Now actually remove the buffers from the queue. 4942 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4943 } 4944} 4945 4946void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4947 const char* logTag) { 4948 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4949 "%s called (reason \"%s\"), but timed buffer queue has no" 4950 " elements to trim.", __FUNCTION__, logTag); 4951 4952 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4953 mTimedBufferQueue.removeAt(0); 4954} 4955 4956void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4957 const TimedBuffer& buf, 4958 const char* logTag) { 4959 uint32_t bufBytes = buf.buffer()->size(); 4960 uint32_t consumedAlready = buf.position(); 4961 4962 ALOG_ASSERT(consumedAlready <= bufBytes, 4963 "Bad bookkeeping while updating frames pending. Timed buffer is" 4964 " only %u bytes long, but claims to have consumed %u" 4965 " bytes. (update reason: \"%s\")", 4966 bufBytes, consumedAlready, logTag); 4967 4968 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4969 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4970 "Bad bookkeeping while updating frames pending. Should have at" 4971 " least %u queued frames, but we think we have only %u. (update" 4972 " reason: \"%s\")", 4973 bufFrames, mFramesPendingInQueue, logTag); 4974 4975 mFramesPendingInQueue -= bufFrames; 4976} 4977 4978status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4979 const sp<IMemory>& buffer, int64_t pts) { 4980 4981 { 4982 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4983 if (!mMediaTimeTransformValid) 4984 return INVALID_OPERATION; 4985 } 4986 4987 Mutex::Autolock _l(mTimedBufferQueueLock); 4988 4989 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4990 mFramesPendingInQueue += bufFrames; 4991 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4992 4993 return NO_ERROR; 4994} 4995 4996status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4997 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4998 4999 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 5000 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 5001 target); 5002 5003 if (!(target == TimedAudioTrack::LOCAL_TIME || 5004 target == TimedAudioTrack::COMMON_TIME)) { 5005 return BAD_VALUE; 5006 } 5007 5008 Mutex::Autolock lock(mMediaTimeTransformLock); 5009 mMediaTimeTransform = xform; 5010 mMediaTimeTransformTarget = target; 5011 mMediaTimeTransformValid = true; 5012 5013 return NO_ERROR; 5014} 5015 5016#define min(a, b) ((a) < (b) ? (a) : (b)) 5017 5018// implementation of getNextBuffer for tracks whose buffers have timestamps 5019status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5020 AudioBufferProvider::Buffer* buffer, int64_t pts) 5021{ 5022 if (pts == AudioBufferProvider::kInvalidPTS) { 5023 buffer->raw = NULL; 5024 buffer->frameCount = 0; 5025 mTimedAudioOutputOnTime = false; 5026 return INVALID_OPERATION; 5027 } 5028 5029 Mutex::Autolock _l(mTimedBufferQueueLock); 5030 5031 ALOG_ASSERT(!mQueueHeadInFlight, 5032 "getNextBuffer called without releaseBuffer!"); 5033 5034 while (true) { 5035 5036 // if we have no timed buffers, then fail 5037 if (mTimedBufferQueue.isEmpty()) { 5038 buffer->raw = NULL; 5039 buffer->frameCount = 0; 5040 return NOT_ENOUGH_DATA; 5041 } 5042 5043 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5044 5045 // calculate the PTS of the head of the timed buffer queue expressed in 5046 // local time 5047 int64_t headLocalPTS; 5048 { 5049 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5050 5051 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5052 5053 if (mMediaTimeTransform.a_to_b_denom == 0) { 5054 // the transform represents a pause, so yield silence 5055 timedYieldSilence_l(buffer->frameCount, buffer); 5056 return NO_ERROR; 5057 } 5058 5059 int64_t transformedPTS; 5060 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5061 &transformedPTS)) { 5062 // the transform failed. this shouldn't happen, but if it does 5063 // then just drop this buffer 5064 ALOGW("timedGetNextBuffer transform failed"); 5065 buffer->raw = NULL; 5066 buffer->frameCount = 0; 5067 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5068 return NO_ERROR; 5069 } 5070 5071 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5072 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5073 &headLocalPTS)) { 5074 buffer->raw = NULL; 5075 buffer->frameCount = 0; 5076 return INVALID_OPERATION; 5077 } 5078 } else { 5079 headLocalPTS = transformedPTS; 5080 } 5081 } 5082 5083 // adjust the head buffer's PTS to reflect the portion of the head buffer 5084 // that has already been consumed 5085 int64_t effectivePTS = headLocalPTS + 5086 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5087 5088 // Calculate the delta in samples between the head of the input buffer 5089 // queue and the start of the next output buffer that will be written. 5090 // If the transformation fails because of over or underflow, it means 5091 // that the sample's position in the output stream is so far out of 5092 // whack that it should just be dropped. 5093 int64_t sampleDelta; 5094 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5095 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5096 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5097 " mix"); 5098 continue; 5099 } 5100 if (!mLocalTimeToSampleTransform.doForwardTransform( 5101 (effectivePTS - pts) << 32, &sampleDelta)) { 5102 ALOGV("*** too late during sample rate transform: dropped buffer"); 5103 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5104 continue; 5105 } 5106 5107 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5108 " sampleDelta=[%d.%08x]", 5109 head.pts(), head.position(), pts, 5110 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5111 + (sampleDelta >> 32)), 5112 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5113 5114 // if the delta between the ideal placement for the next input sample and 5115 // the current output position is within this threshold, then we will 5116 // concatenate the next input samples to the previous output 5117 const int64_t kSampleContinuityThreshold = 5118 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5119 5120 // if this is the first buffer of audio that we're emitting from this track 5121 // then it should be almost exactly on time. 5122 const int64_t kSampleStartupThreshold = 1LL << 32; 5123 5124 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5125 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5126 // the next input is close enough to being on time, so concatenate it 5127 // with the last output 5128 timedYieldSamples_l(buffer); 5129 5130 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5131 head.position(), buffer->frameCount); 5132 return NO_ERROR; 5133 } 5134 5135 // Looks like our output is not on time. Reset our on timed status. 5136 // Next time we mix samples from our input queue, then should be within 5137 // the StartupThreshold. 5138 mTimedAudioOutputOnTime = false; 5139 if (sampleDelta > 0) { 5140 // the gap between the current output position and the proper start of 5141 // the next input sample is too big, so fill it with silence 5142 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5143 5144 timedYieldSilence_l(framesUntilNextInput, buffer); 5145 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5146 return NO_ERROR; 5147 } else { 5148 // the next input sample is late 5149 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5150 size_t onTimeSamplePosition = 5151 head.position() + lateFrames * mCblk->frameSize; 5152 5153 if (onTimeSamplePosition > head.buffer()->size()) { 5154 // all the remaining samples in the head are too late, so 5155 // drop it and move on 5156 ALOGV("*** too late: dropped buffer"); 5157 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5158 continue; 5159 } else { 5160 // skip over the late samples 5161 head.setPosition(onTimeSamplePosition); 5162 5163 // yield the available samples 5164 timedYieldSamples_l(buffer); 5165 5166 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5167 return NO_ERROR; 5168 } 5169 } 5170 } 5171} 5172 5173// Yield samples from the timed buffer queue head up to the given output 5174// buffer's capacity. 5175// 5176// Caller must hold mTimedBufferQueueLock 5177void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5178 AudioBufferProvider::Buffer* buffer) { 5179 5180 const TimedBuffer& head = mTimedBufferQueue[0]; 5181 5182 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5183 head.position()); 5184 5185 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5186 mCblk->frameSize); 5187 size_t framesRequested = buffer->frameCount; 5188 buffer->frameCount = min(framesLeftInHead, framesRequested); 5189 5190 mQueueHeadInFlight = true; 5191 mTimedAudioOutputOnTime = true; 5192} 5193 5194// Yield samples of silence up to the given output buffer's capacity 5195// 5196// Caller must hold mTimedBufferQueueLock 5197void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5198 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5199 5200 // lazily allocate a buffer filled with silence 5201 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5202 delete [] mTimedSilenceBuffer; 5203 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5204 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5205 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5206 } 5207 5208 buffer->raw = mTimedSilenceBuffer; 5209 size_t framesRequested = buffer->frameCount; 5210 buffer->frameCount = min(numFrames, framesRequested); 5211 5212 mTimedAudioOutputOnTime = false; 5213} 5214 5215// AudioBufferProvider interface 5216void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5217 AudioBufferProvider::Buffer* buffer) { 5218 5219 Mutex::Autolock _l(mTimedBufferQueueLock); 5220 5221 // If the buffer which was just released is part of the buffer at the head 5222 // of the queue, be sure to update the amt of the buffer which has been 5223 // consumed. If the buffer being returned is not part of the head of the 5224 // queue, its either because the buffer is part of the silence buffer, or 5225 // because the head of the timed queue was trimmed after the mixer called 5226 // getNextBuffer but before the mixer called releaseBuffer. 5227 if (buffer->raw == mTimedSilenceBuffer) { 5228 ALOG_ASSERT(!mQueueHeadInFlight, 5229 "Queue head in flight during release of silence buffer!"); 5230 goto done; 5231 } 5232 5233 ALOG_ASSERT(mQueueHeadInFlight, 5234 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5235 " head in flight."); 5236 5237 if (mTimedBufferQueue.size()) { 5238 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5239 5240 void* start = head.buffer()->pointer(); 5241 void* end = reinterpret_cast<void*>( 5242 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5243 + head.buffer()->size()); 5244 5245 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5246 "released buffer not within the head of the timed buffer" 5247 " queue; qHead = [%p, %p], released buffer = %p", 5248 start, end, buffer->raw); 5249 5250 head.setPosition(head.position() + 5251 (buffer->frameCount * mCblk->frameSize)); 5252 mQueueHeadInFlight = false; 5253 5254 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5255 "Bad bookkeeping during releaseBuffer! Should have at" 5256 " least %u queued frames, but we think we have only %u", 5257 buffer->frameCount, mFramesPendingInQueue); 5258 5259 mFramesPendingInQueue -= buffer->frameCount; 5260 5261 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5262 || mTrimQueueHeadOnRelease) { 5263 trimTimedBufferQueueHead_l("releaseBuffer"); 5264 mTrimQueueHeadOnRelease = false; 5265 } 5266 } else { 5267 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5268 " buffers in the timed buffer queue"); 5269 } 5270 5271done: 5272 buffer->raw = 0; 5273 buffer->frameCount = 0; 5274} 5275 5276size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5277 Mutex::Autolock _l(mTimedBufferQueueLock); 5278 return mFramesPendingInQueue; 5279} 5280 5281AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5282 : mPTS(0), mPosition(0) {} 5283 5284AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5285 const sp<IMemory>& buffer, int64_t pts) 5286 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5287 5288// ---------------------------------------------------------------------------- 5289 5290// RecordTrack constructor must be called with AudioFlinger::mLock held 5291AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5292 RecordThread *thread, 5293 const sp<Client>& client, 5294 uint32_t sampleRate, 5295 audio_format_t format, 5296 audio_channel_mask_t channelMask, 5297 int frameCount, 5298 int sessionId) 5299 : TrackBase(thread, client, sampleRate, format, 5300 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5301 mOverflow(false) 5302{ 5303 if (mCblk != NULL) { 5304 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5305 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5306 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5307 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5308 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5309 } else { 5310 mCblk->frameSize = sizeof(int8_t); 5311 } 5312 } 5313} 5314 5315AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5316{ 5317 sp<ThreadBase> thread = mThread.promote(); 5318 if (thread != 0) { 5319 AudioSystem::releaseInput(thread->id()); 5320 } 5321} 5322 5323// AudioBufferProvider interface 5324status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5325{ 5326 audio_track_cblk_t* cblk = this->cblk(); 5327 uint32_t framesAvail; 5328 uint32_t framesReq = buffer->frameCount; 5329 5330 // Check if last stepServer failed, try to step now 5331 if (mStepServerFailed) { 5332 if (!step()) goto getNextBuffer_exit; 5333 ALOGV("stepServer recovered"); 5334 mStepServerFailed = false; 5335 } 5336 5337 framesAvail = cblk->framesAvailable_l(); 5338 5339 if (CC_LIKELY(framesAvail)) { 5340 uint32_t s = cblk->server; 5341 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5342 5343 if (framesReq > framesAvail) { 5344 framesReq = framesAvail; 5345 } 5346 if (framesReq > bufferEnd - s) { 5347 framesReq = bufferEnd - s; 5348 } 5349 5350 buffer->raw = getBuffer(s, framesReq); 5351 buffer->frameCount = framesReq; 5352 return NO_ERROR; 5353 } 5354 5355getNextBuffer_exit: 5356 buffer->raw = NULL; 5357 buffer->frameCount = 0; 5358 return NOT_ENOUGH_DATA; 5359} 5360 5361status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5362 int triggerSession) 5363{ 5364 sp<ThreadBase> thread = mThread.promote(); 5365 if (thread != 0) { 5366 RecordThread *recordThread = (RecordThread *)thread.get(); 5367 return recordThread->start(this, event, triggerSession); 5368 } else { 5369 return BAD_VALUE; 5370 } 5371} 5372 5373void AudioFlinger::RecordThread::RecordTrack::stop() 5374{ 5375 sp<ThreadBase> thread = mThread.promote(); 5376 if (thread != 0) { 5377 RecordThread *recordThread = (RecordThread *)thread.get(); 5378 recordThread->stop(this); 5379 TrackBase::reset(); 5380 // Force overrun condition to avoid false overrun callback until first data is 5381 // read from buffer 5382 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5383 } 5384} 5385 5386void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5387{ 5388 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5389 (mClient == 0) ? getpid_cached : mClient->pid(), 5390 mFormat, 5391 mChannelMask, 5392 mSessionId, 5393 mFrameCount, 5394 mState, 5395 mCblk->sampleRate, 5396 mCblk->server, 5397 mCblk->user); 5398} 5399 5400 5401// ---------------------------------------------------------------------------- 5402 5403AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5404 PlaybackThread *playbackThread, 5405 DuplicatingThread *sourceThread, 5406 uint32_t sampleRate, 5407 audio_format_t format, 5408 audio_channel_mask_t channelMask, 5409 int frameCount) 5410 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5411 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5412 mActive(false), mSourceThread(sourceThread) 5413{ 5414 5415 if (mCblk != NULL) { 5416 mCblk->flags |= CBLK_DIRECTION_OUT; 5417 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5418 mOutBuffer.frameCount = 0; 5419 playbackThread->mTracks.add(this); 5420 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5421 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5422 mCblk, mBuffer, mCblk->buffers, 5423 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5424 } else { 5425 ALOGW("Error creating output track on thread %p", playbackThread); 5426 } 5427} 5428 5429AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5430{ 5431 clearBufferQueue(); 5432} 5433 5434status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5435 int triggerSession) 5436{ 5437 status_t status = Track::start(event, triggerSession); 5438 if (status != NO_ERROR) { 5439 return status; 5440 } 5441 5442 mActive = true; 5443 mRetryCount = 127; 5444 return status; 5445} 5446 5447void AudioFlinger::PlaybackThread::OutputTrack::stop() 5448{ 5449 Track::stop(); 5450 clearBufferQueue(); 5451 mOutBuffer.frameCount = 0; 5452 mActive = false; 5453} 5454 5455bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5456{ 5457 Buffer *pInBuffer; 5458 Buffer inBuffer; 5459 uint32_t channelCount = mChannelCount; 5460 bool outputBufferFull = false; 5461 inBuffer.frameCount = frames; 5462 inBuffer.i16 = data; 5463 5464 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5465 5466 if (!mActive && frames != 0) { 5467 start(); 5468 sp<ThreadBase> thread = mThread.promote(); 5469 if (thread != 0) { 5470 MixerThread *mixerThread = (MixerThread *)thread.get(); 5471 if (mCblk->frameCount > frames){ 5472 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5473 uint32_t startFrames = (mCblk->frameCount - frames); 5474 pInBuffer = new Buffer; 5475 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5476 pInBuffer->frameCount = startFrames; 5477 pInBuffer->i16 = pInBuffer->mBuffer; 5478 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5479 mBufferQueue.add(pInBuffer); 5480 } else { 5481 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5482 } 5483 } 5484 } 5485 } 5486 5487 while (waitTimeLeftMs) { 5488 // First write pending buffers, then new data 5489 if (mBufferQueue.size()) { 5490 pInBuffer = mBufferQueue.itemAt(0); 5491 } else { 5492 pInBuffer = &inBuffer; 5493 } 5494 5495 if (pInBuffer->frameCount == 0) { 5496 break; 5497 } 5498 5499 if (mOutBuffer.frameCount == 0) { 5500 mOutBuffer.frameCount = pInBuffer->frameCount; 5501 nsecs_t startTime = systemTime(); 5502 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5503 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5504 outputBufferFull = true; 5505 break; 5506 } 5507 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5508 if (waitTimeLeftMs >= waitTimeMs) { 5509 waitTimeLeftMs -= waitTimeMs; 5510 } else { 5511 waitTimeLeftMs = 0; 5512 } 5513 } 5514 5515 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5516 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5517 mCblk->stepUser(outFrames); 5518 pInBuffer->frameCount -= outFrames; 5519 pInBuffer->i16 += outFrames * channelCount; 5520 mOutBuffer.frameCount -= outFrames; 5521 mOutBuffer.i16 += outFrames * channelCount; 5522 5523 if (pInBuffer->frameCount == 0) { 5524 if (mBufferQueue.size()) { 5525 mBufferQueue.removeAt(0); 5526 delete [] pInBuffer->mBuffer; 5527 delete pInBuffer; 5528 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5529 } else { 5530 break; 5531 } 5532 } 5533 } 5534 5535 // If we could not write all frames, allocate a buffer and queue it for next time. 5536 if (inBuffer.frameCount) { 5537 sp<ThreadBase> thread = mThread.promote(); 5538 if (thread != 0 && !thread->standby()) { 5539 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5540 pInBuffer = new Buffer; 5541 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5542 pInBuffer->frameCount = inBuffer.frameCount; 5543 pInBuffer->i16 = pInBuffer->mBuffer; 5544 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5545 mBufferQueue.add(pInBuffer); 5546 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5547 } else { 5548 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5549 } 5550 } 5551 } 5552 5553 // Calling write() with a 0 length buffer, means that no more data will be written: 5554 // If no more buffers are pending, fill output track buffer to make sure it is started 5555 // by output mixer. 5556 if (frames == 0 && mBufferQueue.size() == 0) { 5557 if (mCblk->user < mCblk->frameCount) { 5558 frames = mCblk->frameCount - mCblk->user; 5559 pInBuffer = new Buffer; 5560 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5561 pInBuffer->frameCount = frames; 5562 pInBuffer->i16 = pInBuffer->mBuffer; 5563 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5564 mBufferQueue.add(pInBuffer); 5565 } else if (mActive) { 5566 stop(); 5567 } 5568 } 5569 5570 return outputBufferFull; 5571} 5572 5573status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5574{ 5575 int active; 5576 status_t result; 5577 audio_track_cblk_t* cblk = mCblk; 5578 uint32_t framesReq = buffer->frameCount; 5579 5580// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5581 buffer->frameCount = 0; 5582 5583 uint32_t framesAvail = cblk->framesAvailable(); 5584 5585 5586 if (framesAvail == 0) { 5587 Mutex::Autolock _l(cblk->lock); 5588 goto start_loop_here; 5589 while (framesAvail == 0) { 5590 active = mActive; 5591 if (CC_UNLIKELY(!active)) { 5592 ALOGV("Not active and NO_MORE_BUFFERS"); 5593 return NO_MORE_BUFFERS; 5594 } 5595 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5596 if (result != NO_ERROR) { 5597 return NO_MORE_BUFFERS; 5598 } 5599 // read the server count again 5600 start_loop_here: 5601 framesAvail = cblk->framesAvailable_l(); 5602 } 5603 } 5604 5605// if (framesAvail < framesReq) { 5606// return NO_MORE_BUFFERS; 5607// } 5608 5609 if (framesReq > framesAvail) { 5610 framesReq = framesAvail; 5611 } 5612 5613 uint32_t u = cblk->user; 5614 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5615 5616 if (framesReq > bufferEnd - u) { 5617 framesReq = bufferEnd - u; 5618 } 5619 5620 buffer->frameCount = framesReq; 5621 buffer->raw = (void *)cblk->buffer(u); 5622 return NO_ERROR; 5623} 5624 5625 5626void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5627{ 5628 size_t size = mBufferQueue.size(); 5629 5630 for (size_t i = 0; i < size; i++) { 5631 Buffer *pBuffer = mBufferQueue.itemAt(i); 5632 delete [] pBuffer->mBuffer; 5633 delete pBuffer; 5634 } 5635 mBufferQueue.clear(); 5636} 5637 5638// ---------------------------------------------------------------------------- 5639 5640AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5641 : RefBase(), 5642 mAudioFlinger(audioFlinger), 5643 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5644 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5645 mPid(pid), 5646 mTimedTrackCount(0) 5647{ 5648 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5649} 5650 5651// Client destructor must be called with AudioFlinger::mLock held 5652AudioFlinger::Client::~Client() 5653{ 5654 mAudioFlinger->removeClient_l(mPid); 5655} 5656 5657sp<MemoryDealer> AudioFlinger::Client::heap() const 5658{ 5659 return mMemoryDealer; 5660} 5661 5662// Reserve one of the limited slots for a timed audio track associated 5663// with this client 5664bool AudioFlinger::Client::reserveTimedTrack() 5665{ 5666 const int kMaxTimedTracksPerClient = 4; 5667 5668 Mutex::Autolock _l(mTimedTrackLock); 5669 5670 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5671 ALOGW("can not create timed track - pid %d has exceeded the limit", 5672 mPid); 5673 return false; 5674 } 5675 5676 mTimedTrackCount++; 5677 return true; 5678} 5679 5680// Release a slot for a timed audio track 5681void AudioFlinger::Client::releaseTimedTrack() 5682{ 5683 Mutex::Autolock _l(mTimedTrackLock); 5684 mTimedTrackCount--; 5685} 5686 5687// ---------------------------------------------------------------------------- 5688 5689AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5690 const sp<IAudioFlingerClient>& client, 5691 pid_t pid) 5692 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5693{ 5694} 5695 5696AudioFlinger::NotificationClient::~NotificationClient() 5697{ 5698} 5699 5700void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5701{ 5702 sp<NotificationClient> keep(this); 5703 mAudioFlinger->removeNotificationClient(mPid); 5704} 5705 5706// ---------------------------------------------------------------------------- 5707 5708AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5709 : BnAudioTrack(), 5710 mTrack(track) 5711{ 5712} 5713 5714AudioFlinger::TrackHandle::~TrackHandle() { 5715 // just stop the track on deletion, associated resources 5716 // will be freed from the main thread once all pending buffers have 5717 // been played. Unless it's not in the active track list, in which 5718 // case we free everything now... 5719 mTrack->destroy(); 5720} 5721 5722sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5723 return mTrack->getCblk(); 5724} 5725 5726status_t AudioFlinger::TrackHandle::start() { 5727 return mTrack->start(); 5728} 5729 5730void AudioFlinger::TrackHandle::stop() { 5731 mTrack->stop(); 5732} 5733 5734void AudioFlinger::TrackHandle::flush() { 5735 mTrack->flush(); 5736} 5737 5738void AudioFlinger::TrackHandle::mute(bool e) { 5739 mTrack->mute(e); 5740} 5741 5742void AudioFlinger::TrackHandle::pause() { 5743 mTrack->pause(); 5744} 5745 5746status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5747{ 5748 return mTrack->attachAuxEffect(EffectId); 5749} 5750 5751status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5752 sp<IMemory>* buffer) { 5753 if (!mTrack->isTimedTrack()) 5754 return INVALID_OPERATION; 5755 5756 PlaybackThread::TimedTrack* tt = 5757 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5758 return tt->allocateTimedBuffer(size, buffer); 5759} 5760 5761status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5762 int64_t pts) { 5763 if (!mTrack->isTimedTrack()) 5764 return INVALID_OPERATION; 5765 5766 PlaybackThread::TimedTrack* tt = 5767 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5768 return tt->queueTimedBuffer(buffer, pts); 5769} 5770 5771status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5772 const LinearTransform& xform, int target) { 5773 5774 if (!mTrack->isTimedTrack()) 5775 return INVALID_OPERATION; 5776 5777 PlaybackThread::TimedTrack* tt = 5778 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5779 return tt->setMediaTimeTransform( 5780 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5781} 5782 5783status_t AudioFlinger::TrackHandle::onTransact( 5784 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5785{ 5786 return BnAudioTrack::onTransact(code, data, reply, flags); 5787} 5788 5789// ---------------------------------------------------------------------------- 5790 5791sp<IAudioRecord> AudioFlinger::openRecord( 5792 pid_t pid, 5793 audio_io_handle_t input, 5794 uint32_t sampleRate, 5795 audio_format_t format, 5796 audio_channel_mask_t channelMask, 5797 int frameCount, 5798 IAudioFlinger::track_flags_t flags, 5799 pid_t tid, 5800 int *sessionId, 5801 status_t *status) 5802{ 5803 sp<RecordThread::RecordTrack> recordTrack; 5804 sp<RecordHandle> recordHandle; 5805 sp<Client> client; 5806 status_t lStatus; 5807 RecordThread *thread; 5808 size_t inFrameCount; 5809 int lSessionId; 5810 5811 // check calling permissions 5812 if (!recordingAllowed()) { 5813 lStatus = PERMISSION_DENIED; 5814 goto Exit; 5815 } 5816 5817 // add client to list 5818 { // scope for mLock 5819 Mutex::Autolock _l(mLock); 5820 thread = checkRecordThread_l(input); 5821 if (thread == NULL) { 5822 lStatus = BAD_VALUE; 5823 goto Exit; 5824 } 5825 5826 client = registerPid_l(pid); 5827 5828 // If no audio session id is provided, create one here 5829 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5830 lSessionId = *sessionId; 5831 } else { 5832 lSessionId = nextUniqueId(); 5833 if (sessionId != NULL) { 5834 *sessionId = lSessionId; 5835 } 5836 } 5837 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5838 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5839 frameCount, lSessionId, flags, tid, &lStatus); 5840 } 5841 if (lStatus != NO_ERROR) { 5842 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5843 // destructor is called by the TrackBase destructor with mLock held 5844 client.clear(); 5845 recordTrack.clear(); 5846 goto Exit; 5847 } 5848 5849 // return to handle to client 5850 recordHandle = new RecordHandle(recordTrack); 5851 lStatus = NO_ERROR; 5852 5853Exit: 5854 if (status) { 5855 *status = lStatus; 5856 } 5857 return recordHandle; 5858} 5859 5860// ---------------------------------------------------------------------------- 5861 5862AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5863 : BnAudioRecord(), 5864 mRecordTrack(recordTrack) 5865{ 5866} 5867 5868AudioFlinger::RecordHandle::~RecordHandle() { 5869 stop_nonvirtual(); 5870} 5871 5872sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5873 return mRecordTrack->getCblk(); 5874} 5875 5876status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5877 ALOGV("RecordHandle::start()"); 5878 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5879} 5880 5881void AudioFlinger::RecordHandle::stop() { 5882 stop_nonvirtual(); 5883} 5884 5885void AudioFlinger::RecordHandle::stop_nonvirtual() { 5886 ALOGV("RecordHandle::stop()"); 5887 mRecordTrack->stop(); 5888} 5889 5890status_t AudioFlinger::RecordHandle::onTransact( 5891 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5892{ 5893 return BnAudioRecord::onTransact(code, data, reply, flags); 5894} 5895 5896// ---------------------------------------------------------------------------- 5897 5898AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5899 AudioStreamIn *input, 5900 uint32_t sampleRate, 5901 audio_channel_mask_t channelMask, 5902 audio_io_handle_t id, 5903 uint32_t device) : 5904 ThreadBase(audioFlinger, id, device, RECORD), 5905 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5906 // mRsmpInIndex and mInputBytes set by readInputParameters() 5907 mReqChannelCount(popcount(channelMask)), 5908 mReqSampleRate(sampleRate) 5909 // mBytesRead is only meaningful while active, and so is cleared in start() 5910 // (but might be better to also clear here for dump?) 5911{ 5912 snprintf(mName, kNameLength, "AudioIn_%X", id); 5913 5914 readInputParameters(); 5915} 5916 5917 5918AudioFlinger::RecordThread::~RecordThread() 5919{ 5920 delete[] mRsmpInBuffer; 5921 delete mResampler; 5922 delete[] mRsmpOutBuffer; 5923} 5924 5925void AudioFlinger::RecordThread::onFirstRef() 5926{ 5927 run(mName, PRIORITY_URGENT_AUDIO); 5928} 5929 5930status_t AudioFlinger::RecordThread::readyToRun() 5931{ 5932 status_t status = initCheck(); 5933 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5934 return status; 5935} 5936 5937bool AudioFlinger::RecordThread::threadLoop() 5938{ 5939 AudioBufferProvider::Buffer buffer; 5940 sp<RecordTrack> activeTrack; 5941 Vector< sp<EffectChain> > effectChains; 5942 5943 nsecs_t lastWarning = 0; 5944 5945 acquireWakeLock(); 5946 5947 // start recording 5948 while (!exitPending()) { 5949 5950 processConfigEvents(); 5951 5952 { // scope for mLock 5953 Mutex::Autolock _l(mLock); 5954 checkForNewParameters_l(); 5955 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5956 if (!mStandby) { 5957 mInput->stream->common.standby(&mInput->stream->common); 5958 mStandby = true; 5959 } 5960 5961 if (exitPending()) break; 5962 5963 releaseWakeLock_l(); 5964 ALOGV("RecordThread: loop stopping"); 5965 // go to sleep 5966 mWaitWorkCV.wait(mLock); 5967 ALOGV("RecordThread: loop starting"); 5968 acquireWakeLock_l(); 5969 continue; 5970 } 5971 if (mActiveTrack != 0) { 5972 if (mActiveTrack->mState == TrackBase::PAUSING) { 5973 if (!mStandby) { 5974 mInput->stream->common.standby(&mInput->stream->common); 5975 mStandby = true; 5976 } 5977 mActiveTrack.clear(); 5978 mStartStopCond.broadcast(); 5979 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5980 if (mReqChannelCount != mActiveTrack->channelCount()) { 5981 mActiveTrack.clear(); 5982 mStartStopCond.broadcast(); 5983 } else if (mBytesRead != 0) { 5984 // record start succeeds only if first read from audio input 5985 // succeeds 5986 if (mBytesRead > 0) { 5987 mActiveTrack->mState = TrackBase::ACTIVE; 5988 } else { 5989 mActiveTrack.clear(); 5990 } 5991 mStartStopCond.broadcast(); 5992 } 5993 mStandby = false; 5994 } 5995 } 5996 lockEffectChains_l(effectChains); 5997 } 5998 5999 if (mActiveTrack != 0) { 6000 if (mActiveTrack->mState != TrackBase::ACTIVE && 6001 mActiveTrack->mState != TrackBase::RESUMING) { 6002 unlockEffectChains(effectChains); 6003 usleep(kRecordThreadSleepUs); 6004 continue; 6005 } 6006 for (size_t i = 0; i < effectChains.size(); i ++) { 6007 effectChains[i]->process_l(); 6008 } 6009 6010 buffer.frameCount = mFrameCount; 6011 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6012 size_t framesOut = buffer.frameCount; 6013 if (mResampler == NULL) { 6014 // no resampling 6015 while (framesOut) { 6016 size_t framesIn = mFrameCount - mRsmpInIndex; 6017 if (framesIn) { 6018 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6019 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6020 if (framesIn > framesOut) 6021 framesIn = framesOut; 6022 mRsmpInIndex += framesIn; 6023 framesOut -= framesIn; 6024 if ((int)mChannelCount == mReqChannelCount || 6025 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6026 memcpy(dst, src, framesIn * mFrameSize); 6027 } else { 6028 int16_t *src16 = (int16_t *)src; 6029 int16_t *dst16 = (int16_t *)dst; 6030 if (mChannelCount == 1) { 6031 while (framesIn--) { 6032 *dst16++ = *src16; 6033 *dst16++ = *src16++; 6034 } 6035 } else { 6036 while (framesIn--) { 6037 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 6038 src16 += 2; 6039 } 6040 } 6041 } 6042 } 6043 if (framesOut && mFrameCount == mRsmpInIndex) { 6044 if (framesOut == mFrameCount && 6045 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6046 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6047 framesOut = 0; 6048 } else { 6049 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6050 mRsmpInIndex = 0; 6051 } 6052 if (mBytesRead < 0) { 6053 ALOGE("Error reading audio input"); 6054 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6055 // Force input into standby so that it tries to 6056 // recover at next read attempt 6057 mInput->stream->common.standby(&mInput->stream->common); 6058 usleep(kRecordThreadSleepUs); 6059 } 6060 mRsmpInIndex = mFrameCount; 6061 framesOut = 0; 6062 buffer.frameCount = 0; 6063 } 6064 } 6065 } 6066 } else { 6067 // resampling 6068 6069 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6070 // alter output frame count as if we were expecting stereo samples 6071 if (mChannelCount == 1 && mReqChannelCount == 1) { 6072 framesOut >>= 1; 6073 } 6074 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6075 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6076 // are 32 bit aligned which should be always true. 6077 if (mChannelCount == 2 && mReqChannelCount == 1) { 6078 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6079 // the resampler always outputs stereo samples: do post stereo to mono conversion 6080 int16_t *src = (int16_t *)mRsmpOutBuffer; 6081 int16_t *dst = buffer.i16; 6082 while (framesOut--) { 6083 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 6084 src += 2; 6085 } 6086 } else { 6087 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6088 } 6089 6090 } 6091 if (mFramestoDrop == 0) { 6092 mActiveTrack->releaseBuffer(&buffer); 6093 } else { 6094 if (mFramestoDrop > 0) { 6095 mFramestoDrop -= buffer.frameCount; 6096 if (mFramestoDrop <= 0) { 6097 clearSyncStartEvent(); 6098 } 6099 } else { 6100 mFramestoDrop += buffer.frameCount; 6101 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6102 mSyncStartEvent->isCancelled()) { 6103 ALOGW("Synced record %s, session %d, trigger session %d", 6104 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6105 mActiveTrack->sessionId(), 6106 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6107 clearSyncStartEvent(); 6108 } 6109 } 6110 } 6111 mActiveTrack->clearOverflow(); 6112 } 6113 // client isn't retrieving buffers fast enough 6114 else { 6115 if (!mActiveTrack->setOverflow()) { 6116 nsecs_t now = systemTime(); 6117 if ((now - lastWarning) > kWarningThrottleNs) { 6118 ALOGW("RecordThread: buffer overflow"); 6119 lastWarning = now; 6120 } 6121 } 6122 // Release the processor for a while before asking for a new buffer. 6123 // This will give the application more chance to read from the buffer and 6124 // clear the overflow. 6125 usleep(kRecordThreadSleepUs); 6126 } 6127 } 6128 // enable changes in effect chain 6129 unlockEffectChains(effectChains); 6130 effectChains.clear(); 6131 } 6132 6133 if (!mStandby) { 6134 mInput->stream->common.standby(&mInput->stream->common); 6135 } 6136 mActiveTrack.clear(); 6137 6138 mStartStopCond.broadcast(); 6139 6140 releaseWakeLock(); 6141 6142 ALOGV("RecordThread %p exiting", this); 6143 return false; 6144} 6145 6146 6147sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6148 const sp<AudioFlinger::Client>& client, 6149 uint32_t sampleRate, 6150 audio_format_t format, 6151 audio_channel_mask_t channelMask, 6152 int frameCount, 6153 int sessionId, 6154 IAudioFlinger::track_flags_t flags, 6155 pid_t tid, 6156 status_t *status) 6157{ 6158 sp<RecordTrack> track; 6159 status_t lStatus; 6160 6161 lStatus = initCheck(); 6162 if (lStatus != NO_ERROR) { 6163 ALOGE("Audio driver not initialized."); 6164 goto Exit; 6165 } 6166 6167 // FIXME use flags and tid similar to createTrack_l() 6168 6169 { // scope for mLock 6170 Mutex::Autolock _l(mLock); 6171 6172 track = new RecordTrack(this, client, sampleRate, 6173 format, channelMask, frameCount, sessionId); 6174 6175 if (track->getCblk() == 0) { 6176 lStatus = NO_MEMORY; 6177 goto Exit; 6178 } 6179 6180 mTrack = track.get(); 6181 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6182 bool suspend = audio_is_bluetooth_sco_device( 6183 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6184 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6185 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6186 } 6187 lStatus = NO_ERROR; 6188 6189Exit: 6190 if (status) { 6191 *status = lStatus; 6192 } 6193 return track; 6194} 6195 6196status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6197 AudioSystem::sync_event_t event, 6198 int triggerSession) 6199{ 6200 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6201 sp<ThreadBase> strongMe = this; 6202 status_t status = NO_ERROR; 6203 6204 if (event == AudioSystem::SYNC_EVENT_NONE) { 6205 clearSyncStartEvent(); 6206 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6207 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6208 triggerSession, 6209 recordTrack->sessionId(), 6210 syncStartEventCallback, 6211 this); 6212 // Sync event can be cancelled by the trigger session if the track is not in a 6213 // compatible state in which case we start record immediately 6214 if (mSyncStartEvent->isCancelled()) { 6215 clearSyncStartEvent(); 6216 } else { 6217 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6218 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6219 } 6220 } 6221 6222 { 6223 AutoMutex lock(mLock); 6224 if (mActiveTrack != 0) { 6225 if (recordTrack != mActiveTrack.get()) { 6226 status = -EBUSY; 6227 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6228 mActiveTrack->mState = TrackBase::ACTIVE; 6229 } 6230 return status; 6231 } 6232 6233 recordTrack->mState = TrackBase::IDLE; 6234 mActiveTrack = recordTrack; 6235 mLock.unlock(); 6236 status_t status = AudioSystem::startInput(mId); 6237 mLock.lock(); 6238 if (status != NO_ERROR) { 6239 mActiveTrack.clear(); 6240 clearSyncStartEvent(); 6241 return status; 6242 } 6243 mRsmpInIndex = mFrameCount; 6244 mBytesRead = 0; 6245 if (mResampler != NULL) { 6246 mResampler->reset(); 6247 } 6248 mActiveTrack->mState = TrackBase::RESUMING; 6249 // signal thread to start 6250 ALOGV("Signal record thread"); 6251 mWaitWorkCV.signal(); 6252 // do not wait for mStartStopCond if exiting 6253 if (exitPending()) { 6254 mActiveTrack.clear(); 6255 status = INVALID_OPERATION; 6256 goto startError; 6257 } 6258 mStartStopCond.wait(mLock); 6259 if (mActiveTrack == 0) { 6260 ALOGV("Record failed to start"); 6261 status = BAD_VALUE; 6262 goto startError; 6263 } 6264 ALOGV("Record started OK"); 6265 return status; 6266 } 6267startError: 6268 AudioSystem::stopInput(mId); 6269 clearSyncStartEvent(); 6270 return status; 6271} 6272 6273void AudioFlinger::RecordThread::clearSyncStartEvent() 6274{ 6275 if (mSyncStartEvent != 0) { 6276 mSyncStartEvent->cancel(); 6277 } 6278 mSyncStartEvent.clear(); 6279 mFramestoDrop = 0; 6280} 6281 6282void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6283{ 6284 sp<SyncEvent> strongEvent = event.promote(); 6285 6286 if (strongEvent != 0) { 6287 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6288 me->handleSyncStartEvent(strongEvent); 6289 } 6290} 6291 6292void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6293{ 6294 if (event == mSyncStartEvent) { 6295 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6296 // from audio HAL 6297 mFramestoDrop = mFrameCount * 2; 6298 } 6299} 6300 6301void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6302 ALOGV("RecordThread::stop"); 6303 sp<ThreadBase> strongMe = this; 6304 { 6305 AutoMutex lock(mLock); 6306 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6307 mActiveTrack->mState = TrackBase::PAUSING; 6308 // do not wait for mStartStopCond if exiting 6309 if (exitPending()) { 6310 return; 6311 } 6312 mStartStopCond.wait(mLock); 6313 // if we have been restarted, recordTrack == mActiveTrack.get() here 6314 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6315 mLock.unlock(); 6316 AudioSystem::stopInput(mId); 6317 mLock.lock(); 6318 ALOGV("Record stopped OK"); 6319 } 6320 } 6321 } 6322} 6323 6324bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6325{ 6326 return false; 6327} 6328 6329status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6330{ 6331 if (!isValidSyncEvent(event)) { 6332 return BAD_VALUE; 6333 } 6334 6335 Mutex::Autolock _l(mLock); 6336 6337 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6338 mTrack->setSyncEvent(event); 6339 return NO_ERROR; 6340 } 6341 return NAME_NOT_FOUND; 6342} 6343 6344status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6345{ 6346 const size_t SIZE = 256; 6347 char buffer[SIZE]; 6348 String8 result; 6349 6350 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6351 result.append(buffer); 6352 6353 if (mActiveTrack != 0) { 6354 result.append("Active Track:\n"); 6355 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6356 mActiveTrack->dump(buffer, SIZE); 6357 result.append(buffer); 6358 6359 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6360 result.append(buffer); 6361 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6362 result.append(buffer); 6363 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6364 result.append(buffer); 6365 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6366 result.append(buffer); 6367 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6368 result.append(buffer); 6369 6370 6371 } else { 6372 result.append("No record client\n"); 6373 } 6374 write(fd, result.string(), result.size()); 6375 6376 dumpBase(fd, args); 6377 dumpEffectChains(fd, args); 6378 6379 return NO_ERROR; 6380} 6381 6382// AudioBufferProvider interface 6383status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6384{ 6385 size_t framesReq = buffer->frameCount; 6386 size_t framesReady = mFrameCount - mRsmpInIndex; 6387 int channelCount; 6388 6389 if (framesReady == 0) { 6390 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6391 if (mBytesRead < 0) { 6392 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6393 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6394 // Force input into standby so that it tries to 6395 // recover at next read attempt 6396 mInput->stream->common.standby(&mInput->stream->common); 6397 usleep(kRecordThreadSleepUs); 6398 } 6399 buffer->raw = NULL; 6400 buffer->frameCount = 0; 6401 return NOT_ENOUGH_DATA; 6402 } 6403 mRsmpInIndex = 0; 6404 framesReady = mFrameCount; 6405 } 6406 6407 if (framesReq > framesReady) { 6408 framesReq = framesReady; 6409 } 6410 6411 if (mChannelCount == 1 && mReqChannelCount == 2) { 6412 channelCount = 1; 6413 } else { 6414 channelCount = 2; 6415 } 6416 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6417 buffer->frameCount = framesReq; 6418 return NO_ERROR; 6419} 6420 6421// AudioBufferProvider interface 6422void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6423{ 6424 mRsmpInIndex += buffer->frameCount; 6425 buffer->frameCount = 0; 6426} 6427 6428bool AudioFlinger::RecordThread::checkForNewParameters_l() 6429{ 6430 bool reconfig = false; 6431 6432 while (!mNewParameters.isEmpty()) { 6433 status_t status = NO_ERROR; 6434 String8 keyValuePair = mNewParameters[0]; 6435 AudioParameter param = AudioParameter(keyValuePair); 6436 int value; 6437 audio_format_t reqFormat = mFormat; 6438 int reqSamplingRate = mReqSampleRate; 6439 int reqChannelCount = mReqChannelCount; 6440 6441 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6442 reqSamplingRate = value; 6443 reconfig = true; 6444 } 6445 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6446 reqFormat = (audio_format_t) value; 6447 reconfig = true; 6448 } 6449 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6450 reqChannelCount = popcount(value); 6451 reconfig = true; 6452 } 6453 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6454 // do not accept frame count changes if tracks are open as the track buffer 6455 // size depends on frame count and correct behavior would not be guaranteed 6456 // if frame count is changed after track creation 6457 if (mActiveTrack != 0) { 6458 status = INVALID_OPERATION; 6459 } else { 6460 reconfig = true; 6461 } 6462 } 6463 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6464 // forward device change to effects that have requested to be 6465 // aware of attached audio device. 6466 for (size_t i = 0; i < mEffectChains.size(); i++) { 6467 mEffectChains[i]->setDevice_l(value); 6468 } 6469 // store input device and output device but do not forward output device to audio HAL. 6470 // Note that status is ignored by the caller for output device 6471 // (see AudioFlinger::setParameters() 6472 uint32_t /*audio_devices_t*/ newDevice = mDevice; 6473 if (value & AUDIO_DEVICE_OUT_ALL) { 6474 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6475 status = BAD_VALUE; 6476 } else { 6477 newDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6478 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6479 if (mTrack != NULL) { 6480 bool suspend = audio_is_bluetooth_sco_device( 6481 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6482 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6483 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6484 } 6485 } 6486 newDevice |= value; 6487 mDevice = (audio_devices_t) newDevice; // since mDevice is read by other threads, only write to it once 6488 } 6489 if (status == NO_ERROR) { 6490 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6491 if (status == INVALID_OPERATION) { 6492 mInput->stream->common.standby(&mInput->stream->common); 6493 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6494 keyValuePair.string()); 6495 } 6496 if (reconfig) { 6497 if (status == BAD_VALUE && 6498 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6499 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6500 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6501 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6502 (reqChannelCount <= FCC_2)) { 6503 status = NO_ERROR; 6504 } 6505 if (status == NO_ERROR) { 6506 readInputParameters(); 6507 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6508 } 6509 } 6510 } 6511 6512 mNewParameters.removeAt(0); 6513 6514 mParamStatus = status; 6515 mParamCond.signal(); 6516 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6517 // already timed out waiting for the status and will never signal the condition. 6518 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6519 } 6520 return reconfig; 6521} 6522 6523String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6524{ 6525 char *s; 6526 String8 out_s8 = String8(); 6527 6528 Mutex::Autolock _l(mLock); 6529 if (initCheck() != NO_ERROR) { 6530 return out_s8; 6531 } 6532 6533 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6534 out_s8 = String8(s); 6535 free(s); 6536 return out_s8; 6537} 6538 6539void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6540 AudioSystem::OutputDescriptor desc; 6541 void *param2 = NULL; 6542 6543 switch (event) { 6544 case AudioSystem::INPUT_OPENED: 6545 case AudioSystem::INPUT_CONFIG_CHANGED: 6546 desc.channels = mChannelMask; 6547 desc.samplingRate = mSampleRate; 6548 desc.format = mFormat; 6549 desc.frameCount = mFrameCount; 6550 desc.latency = 0; 6551 param2 = &desc; 6552 break; 6553 6554 case AudioSystem::INPUT_CLOSED: 6555 default: 6556 break; 6557 } 6558 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6559} 6560 6561void AudioFlinger::RecordThread::readInputParameters() 6562{ 6563 delete mRsmpInBuffer; 6564 // mRsmpInBuffer is always assigned a new[] below 6565 delete mRsmpOutBuffer; 6566 mRsmpOutBuffer = NULL; 6567 delete mResampler; 6568 mResampler = NULL; 6569 6570 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6571 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6572 mChannelCount = (uint16_t)popcount(mChannelMask); 6573 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6574 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6575 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6576 mFrameCount = mInputBytes / mFrameSize; 6577 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6578 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6579 6580 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6581 { 6582 int channelCount; 6583 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6584 // stereo to mono post process as the resampler always outputs stereo. 6585 if (mChannelCount == 1 && mReqChannelCount == 2) { 6586 channelCount = 1; 6587 } else { 6588 channelCount = 2; 6589 } 6590 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6591 mResampler->setSampleRate(mSampleRate); 6592 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6593 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6594 6595 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6596 if (mChannelCount == 1 && mReqChannelCount == 1) { 6597 mFrameCount >>= 1; 6598 } 6599 6600 } 6601 mRsmpInIndex = mFrameCount; 6602} 6603 6604unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6605{ 6606 Mutex::Autolock _l(mLock); 6607 if (initCheck() != NO_ERROR) { 6608 return 0; 6609 } 6610 6611 return mInput->stream->get_input_frames_lost(mInput->stream); 6612} 6613 6614uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6615{ 6616 Mutex::Autolock _l(mLock); 6617 uint32_t result = 0; 6618 if (getEffectChain_l(sessionId) != 0) { 6619 result = EFFECT_SESSION; 6620 } 6621 6622 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6623 result |= TRACK_SESSION; 6624 } 6625 6626 return result; 6627} 6628 6629AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6630{ 6631 Mutex::Autolock _l(mLock); 6632 return mTrack; 6633} 6634 6635AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6636{ 6637 Mutex::Autolock _l(mLock); 6638 AudioStreamIn *input = mInput; 6639 mInput = NULL; 6640 return input; 6641} 6642 6643// this method must always be called either with ThreadBase mLock held or inside the thread loop 6644audio_stream_t* AudioFlinger::RecordThread::stream() const 6645{ 6646 if (mInput == NULL) { 6647 return NULL; 6648 } 6649 return &mInput->stream->common; 6650} 6651 6652 6653// ---------------------------------------------------------------------------- 6654 6655audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6656{ 6657 if (!settingsAllowed()) { 6658 return 0; 6659 } 6660 Mutex::Autolock _l(mLock); 6661 return loadHwModule_l(name); 6662} 6663 6664// loadHwModule_l() must be called with AudioFlinger::mLock held 6665audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6666{ 6667 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6668 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6669 ALOGW("loadHwModule() module %s already loaded", name); 6670 return mAudioHwDevs.keyAt(i); 6671 } 6672 } 6673 6674 audio_hw_device_t *dev; 6675 6676 int rc = load_audio_interface(name, &dev); 6677 if (rc) { 6678 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6679 return 0; 6680 } 6681 6682 mHardwareStatus = AUDIO_HW_INIT; 6683 rc = dev->init_check(dev); 6684 mHardwareStatus = AUDIO_HW_IDLE; 6685 if (rc) { 6686 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6687 return 0; 6688 } 6689 6690 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6691 (NULL != dev->set_master_volume)) { 6692 AutoMutex lock(mHardwareLock); 6693 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6694 dev->set_master_volume(dev, mMasterVolume); 6695 mHardwareStatus = AUDIO_HW_IDLE; 6696 } 6697 6698 audio_module_handle_t handle = nextUniqueId(); 6699 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6700 6701 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6702 name, dev->common.module->name, dev->common.module->id, handle); 6703 6704 return handle; 6705 6706} 6707 6708audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6709 audio_devices_t *pDevices, 6710 uint32_t *pSamplingRate, 6711 audio_format_t *pFormat, 6712 audio_channel_mask_t *pChannelMask, 6713 uint32_t *pLatencyMs, 6714 audio_output_flags_t flags) 6715{ 6716 status_t status; 6717 PlaybackThread *thread = NULL; 6718 struct audio_config config = { 6719 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6720 channel_mask: pChannelMask ? *pChannelMask : 0, 6721 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6722 }; 6723 audio_stream_out_t *outStream = NULL; 6724 audio_hw_device_t *outHwDev; 6725 6726 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6727 module, 6728 (pDevices != NULL) ? (int)*pDevices : 0, 6729 config.sample_rate, 6730 config.format, 6731 config.channel_mask, 6732 flags); 6733 6734 if (pDevices == NULL || *pDevices == 0) { 6735 return 0; 6736 } 6737 6738 Mutex::Autolock _l(mLock); 6739 6740 outHwDev = findSuitableHwDev_l(module, *pDevices); 6741 if (outHwDev == NULL) 6742 return 0; 6743 6744 audio_io_handle_t id = nextUniqueId(); 6745 6746 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6747 6748 status = outHwDev->open_output_stream(outHwDev, 6749 id, 6750 *pDevices, 6751 (audio_output_flags_t)flags, 6752 &config, 6753 &outStream); 6754 6755 mHardwareStatus = AUDIO_HW_IDLE; 6756 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6757 outStream, 6758 config.sample_rate, 6759 config.format, 6760 config.channel_mask, 6761 status); 6762 6763 if (status == NO_ERROR && outStream != NULL) { 6764 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6765 6766 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6767 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6768 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6769 thread = new DirectOutputThread(this, output, id, *pDevices); 6770 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6771 } else { 6772 thread = new MixerThread(this, output, id, *pDevices); 6773 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6774 } 6775 mPlaybackThreads.add(id, thread); 6776 6777 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6778 if (pFormat != NULL) *pFormat = config.format; 6779 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6780 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6781 6782 // notify client processes of the new output creation 6783 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6784 6785 // the first primary output opened designates the primary hw device 6786 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6787 ALOGI("Using module %d has the primary audio interface", module); 6788 mPrimaryHardwareDev = outHwDev; 6789 6790 AutoMutex lock(mHardwareLock); 6791 mHardwareStatus = AUDIO_HW_SET_MODE; 6792 outHwDev->set_mode(outHwDev, mMode); 6793 6794 // Determine the level of master volume support the primary audio HAL has, 6795 // and set the initial master volume at the same time. 6796 float initialVolume = 1.0; 6797 mMasterVolumeSupportLvl = MVS_NONE; 6798 6799 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6800 if ((NULL != outHwDev->get_master_volume) && 6801 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6802 mMasterVolumeSupportLvl = MVS_FULL; 6803 } else { 6804 mMasterVolumeSupportLvl = MVS_SETONLY; 6805 initialVolume = 1.0; 6806 } 6807 6808 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6809 if ((NULL == outHwDev->set_master_volume) || 6810 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6811 mMasterVolumeSupportLvl = MVS_NONE; 6812 } 6813 // now that we have a primary device, initialize master volume on other devices 6814 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6815 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6816 6817 if ((dev != mPrimaryHardwareDev) && 6818 (NULL != dev->set_master_volume)) { 6819 dev->set_master_volume(dev, initialVolume); 6820 } 6821 } 6822 mHardwareStatus = AUDIO_HW_IDLE; 6823 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6824 ? initialVolume 6825 : 1.0; 6826 mMasterVolume = initialVolume; 6827 } 6828 return id; 6829 } 6830 6831 return 0; 6832} 6833 6834audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6835 audio_io_handle_t output2) 6836{ 6837 Mutex::Autolock _l(mLock); 6838 MixerThread *thread1 = checkMixerThread_l(output1); 6839 MixerThread *thread2 = checkMixerThread_l(output2); 6840 6841 if (thread1 == NULL || thread2 == NULL) { 6842 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6843 return 0; 6844 } 6845 6846 audio_io_handle_t id = nextUniqueId(); 6847 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6848 thread->addOutputTrack(thread2); 6849 mPlaybackThreads.add(id, thread); 6850 // notify client processes of the new output creation 6851 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6852 return id; 6853} 6854 6855status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6856{ 6857 return closeOutput_nonvirtual(output); 6858} 6859 6860status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 6861{ 6862 // keep strong reference on the playback thread so that 6863 // it is not destroyed while exit() is executed 6864 sp<PlaybackThread> thread; 6865 { 6866 Mutex::Autolock _l(mLock); 6867 thread = checkPlaybackThread_l(output); 6868 if (thread == NULL) { 6869 return BAD_VALUE; 6870 } 6871 6872 ALOGV("closeOutput() %d", output); 6873 6874 if (thread->type() == ThreadBase::MIXER) { 6875 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6876 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6877 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6878 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6879 } 6880 } 6881 } 6882 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6883 mPlaybackThreads.removeItem(output); 6884 } 6885 thread->exit(); 6886 // The thread entity (active unit of execution) is no longer running here, 6887 // but the ThreadBase container still exists. 6888 6889 if (thread->type() != ThreadBase::DUPLICATING) { 6890 AudioStreamOut *out = thread->clearOutput(); 6891 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6892 // from now on thread->mOutput is NULL 6893 out->hwDev->close_output_stream(out->hwDev, out->stream); 6894 delete out; 6895 } 6896 return NO_ERROR; 6897} 6898 6899status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6900{ 6901 Mutex::Autolock _l(mLock); 6902 PlaybackThread *thread = checkPlaybackThread_l(output); 6903 6904 if (thread == NULL) { 6905 return BAD_VALUE; 6906 } 6907 6908 ALOGV("suspendOutput() %d", output); 6909 thread->suspend(); 6910 6911 return NO_ERROR; 6912} 6913 6914status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6915{ 6916 Mutex::Autolock _l(mLock); 6917 PlaybackThread *thread = checkPlaybackThread_l(output); 6918 6919 if (thread == NULL) { 6920 return BAD_VALUE; 6921 } 6922 6923 ALOGV("restoreOutput() %d", output); 6924 6925 thread->restore(); 6926 6927 return NO_ERROR; 6928} 6929 6930audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6931 audio_devices_t *pDevices, 6932 uint32_t *pSamplingRate, 6933 audio_format_t *pFormat, 6934 audio_channel_mask_t *pChannelMask) 6935{ 6936 status_t status; 6937 RecordThread *thread = NULL; 6938 struct audio_config config = { 6939 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6940 channel_mask: pChannelMask ? *pChannelMask : 0, 6941 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6942 }; 6943 uint32_t reqSamplingRate = config.sample_rate; 6944 audio_format_t reqFormat = config.format; 6945 audio_channel_mask_t reqChannels = config.channel_mask; 6946 audio_stream_in_t *inStream = NULL; 6947 audio_hw_device_t *inHwDev; 6948 6949 if (pDevices == NULL || *pDevices == 0) { 6950 return 0; 6951 } 6952 6953 Mutex::Autolock _l(mLock); 6954 6955 inHwDev = findSuitableHwDev_l(module, *pDevices); 6956 if (inHwDev == NULL) 6957 return 0; 6958 6959 audio_io_handle_t id = nextUniqueId(); 6960 6961 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6962 &inStream); 6963 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6964 inStream, 6965 config.sample_rate, 6966 config.format, 6967 config.channel_mask, 6968 status); 6969 6970 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6971 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6972 // or stereo to mono conversions on 16 bit PCM inputs. 6973 if (status == BAD_VALUE && 6974 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6975 (config.sample_rate <= 2 * reqSamplingRate) && 6976 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6977 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 6978 inStream = NULL; 6979 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6980 } 6981 6982 if (status == NO_ERROR && inStream != NULL) { 6983 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6984 6985 // Start record thread 6986 // RecorThread require both input and output device indication to forward to audio 6987 // pre processing modules 6988 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6989 thread = new RecordThread(this, 6990 input, 6991 reqSamplingRate, 6992 reqChannels, 6993 id, 6994 device); 6995 mRecordThreads.add(id, thread); 6996 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6997 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6998 if (pFormat != NULL) *pFormat = config.format; 6999 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7000 7001 input->stream->common.standby(&input->stream->common); 7002 7003 // notify client processes of the new input creation 7004 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7005 return id; 7006 } 7007 7008 return 0; 7009} 7010 7011status_t AudioFlinger::closeInput(audio_io_handle_t input) 7012{ 7013 return closeInput_nonvirtual(input); 7014} 7015 7016status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7017{ 7018 // keep strong reference on the record thread so that 7019 // it is not destroyed while exit() is executed 7020 sp<RecordThread> thread; 7021 { 7022 Mutex::Autolock _l(mLock); 7023 thread = checkRecordThread_l(input); 7024 if (thread == 0) { 7025 return BAD_VALUE; 7026 } 7027 7028 ALOGV("closeInput() %d", input); 7029 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7030 mRecordThreads.removeItem(input); 7031 } 7032 thread->exit(); 7033 // The thread entity (active unit of execution) is no longer running here, 7034 // but the ThreadBase container still exists. 7035 7036 AudioStreamIn *in = thread->clearInput(); 7037 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7038 // from now on thread->mInput is NULL 7039 in->hwDev->close_input_stream(in->hwDev, in->stream); 7040 delete in; 7041 7042 return NO_ERROR; 7043} 7044 7045status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7046{ 7047 Mutex::Autolock _l(mLock); 7048 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7049 7050 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7051 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7052 thread->invalidateTracks(stream); 7053 } 7054 7055 return NO_ERROR; 7056} 7057 7058 7059int AudioFlinger::newAudioSessionId() 7060{ 7061 return nextUniqueId(); 7062} 7063 7064void AudioFlinger::acquireAudioSessionId(int audioSession) 7065{ 7066 Mutex::Autolock _l(mLock); 7067 pid_t caller = IPCThreadState::self()->getCallingPid(); 7068 ALOGV("acquiring %d from %d", audioSession, caller); 7069 size_t num = mAudioSessionRefs.size(); 7070 for (size_t i = 0; i< num; i++) { 7071 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7072 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7073 ref->mCnt++; 7074 ALOGV(" incremented refcount to %d", ref->mCnt); 7075 return; 7076 } 7077 } 7078 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7079 ALOGV(" added new entry for %d", audioSession); 7080} 7081 7082void AudioFlinger::releaseAudioSessionId(int audioSession) 7083{ 7084 Mutex::Autolock _l(mLock); 7085 pid_t caller = IPCThreadState::self()->getCallingPid(); 7086 ALOGV("releasing %d from %d", audioSession, caller); 7087 size_t num = mAudioSessionRefs.size(); 7088 for (size_t i = 0; i< num; i++) { 7089 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7090 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7091 ref->mCnt--; 7092 ALOGV(" decremented refcount to %d", ref->mCnt); 7093 if (ref->mCnt == 0) { 7094 mAudioSessionRefs.removeAt(i); 7095 delete ref; 7096 purgeStaleEffects_l(); 7097 } 7098 return; 7099 } 7100 } 7101 ALOGW("session id %d not found for pid %d", audioSession, caller); 7102} 7103 7104void AudioFlinger::purgeStaleEffects_l() { 7105 7106 ALOGV("purging stale effects"); 7107 7108 Vector< sp<EffectChain> > chains; 7109 7110 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7111 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7112 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7113 sp<EffectChain> ec = t->mEffectChains[j]; 7114 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7115 chains.push(ec); 7116 } 7117 } 7118 } 7119 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7120 sp<RecordThread> t = mRecordThreads.valueAt(i); 7121 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7122 sp<EffectChain> ec = t->mEffectChains[j]; 7123 chains.push(ec); 7124 } 7125 } 7126 7127 for (size_t i = 0; i < chains.size(); i++) { 7128 sp<EffectChain> ec = chains[i]; 7129 int sessionid = ec->sessionId(); 7130 sp<ThreadBase> t = ec->mThread.promote(); 7131 if (t == 0) { 7132 continue; 7133 } 7134 size_t numsessionrefs = mAudioSessionRefs.size(); 7135 bool found = false; 7136 for (size_t k = 0; k < numsessionrefs; k++) { 7137 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7138 if (ref->mSessionid == sessionid) { 7139 ALOGV(" session %d still exists for %d with %d refs", 7140 sessionid, ref->mPid, ref->mCnt); 7141 found = true; 7142 break; 7143 } 7144 } 7145 if (!found) { 7146 Mutex::Autolock _l (t->mLock); 7147 // remove all effects from the chain 7148 while (ec->mEffects.size()) { 7149 sp<EffectModule> effect = ec->mEffects[0]; 7150 effect->unPin(); 7151 t->removeEffect_l(effect); 7152 if (effect->purgeHandles()) { 7153 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7154 } 7155 AudioSystem::unregisterEffect(effect->id()); 7156 } 7157 } 7158 } 7159 return; 7160} 7161 7162// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7163AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7164{ 7165 return mPlaybackThreads.valueFor(output).get(); 7166} 7167 7168// checkMixerThread_l() must be called with AudioFlinger::mLock held 7169AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7170{ 7171 PlaybackThread *thread = checkPlaybackThread_l(output); 7172 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7173} 7174 7175// checkRecordThread_l() must be called with AudioFlinger::mLock held 7176AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7177{ 7178 return mRecordThreads.valueFor(input).get(); 7179} 7180 7181uint32_t AudioFlinger::nextUniqueId() 7182{ 7183 return android_atomic_inc(&mNextUniqueId); 7184} 7185 7186AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7187{ 7188 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7189 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7190 AudioStreamOut *output = thread->getOutput(); 7191 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7192 return thread; 7193 } 7194 } 7195 return NULL; 7196} 7197 7198uint32_t AudioFlinger::primaryOutputDevice_l() const 7199{ 7200 PlaybackThread *thread = primaryPlaybackThread_l(); 7201 7202 if (thread == NULL) { 7203 return 0; 7204 } 7205 7206 return thread->device(); 7207} 7208 7209sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7210 int triggerSession, 7211 int listenerSession, 7212 sync_event_callback_t callBack, 7213 void *cookie) 7214{ 7215 Mutex::Autolock _l(mLock); 7216 7217 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7218 status_t playStatus = NAME_NOT_FOUND; 7219 status_t recStatus = NAME_NOT_FOUND; 7220 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7221 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7222 if (playStatus == NO_ERROR) { 7223 return event; 7224 } 7225 } 7226 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7227 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7228 if (recStatus == NO_ERROR) { 7229 return event; 7230 } 7231 } 7232 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7233 mPendingSyncEvents.add(event); 7234 } else { 7235 ALOGV("createSyncEvent() invalid event %d", event->type()); 7236 event.clear(); 7237 } 7238 return event; 7239} 7240 7241// ---------------------------------------------------------------------------- 7242// Effect management 7243// ---------------------------------------------------------------------------- 7244 7245 7246status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7247{ 7248 Mutex::Autolock _l(mLock); 7249 return EffectQueryNumberEffects(numEffects); 7250} 7251 7252status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7253{ 7254 Mutex::Autolock _l(mLock); 7255 return EffectQueryEffect(index, descriptor); 7256} 7257 7258status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7259 effect_descriptor_t *descriptor) const 7260{ 7261 Mutex::Autolock _l(mLock); 7262 return EffectGetDescriptor(pUuid, descriptor); 7263} 7264 7265 7266sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7267 effect_descriptor_t *pDesc, 7268 const sp<IEffectClient>& effectClient, 7269 int32_t priority, 7270 audio_io_handle_t io, 7271 int sessionId, 7272 status_t *status, 7273 int *id, 7274 int *enabled) 7275{ 7276 status_t lStatus = NO_ERROR; 7277 sp<EffectHandle> handle; 7278 effect_descriptor_t desc; 7279 7280 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7281 pid, effectClient.get(), priority, sessionId, io); 7282 7283 if (pDesc == NULL) { 7284 lStatus = BAD_VALUE; 7285 goto Exit; 7286 } 7287 7288 // check audio settings permission for global effects 7289 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7290 lStatus = PERMISSION_DENIED; 7291 goto Exit; 7292 } 7293 7294 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7295 // that can only be created by audio policy manager (running in same process) 7296 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7297 lStatus = PERMISSION_DENIED; 7298 goto Exit; 7299 } 7300 7301 if (io == 0) { 7302 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7303 // output must be specified by AudioPolicyManager when using session 7304 // AUDIO_SESSION_OUTPUT_STAGE 7305 lStatus = BAD_VALUE; 7306 goto Exit; 7307 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7308 // if the output returned by getOutputForEffect() is removed before we lock the 7309 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7310 // and we will exit safely 7311 io = AudioSystem::getOutputForEffect(&desc); 7312 } 7313 } 7314 7315 { 7316 Mutex::Autolock _l(mLock); 7317 7318 7319 if (!EffectIsNullUuid(&pDesc->uuid)) { 7320 // if uuid is specified, request effect descriptor 7321 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7322 if (lStatus < 0) { 7323 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7324 goto Exit; 7325 } 7326 } else { 7327 // if uuid is not specified, look for an available implementation 7328 // of the required type in effect factory 7329 if (EffectIsNullUuid(&pDesc->type)) { 7330 ALOGW("createEffect() no effect type"); 7331 lStatus = BAD_VALUE; 7332 goto Exit; 7333 } 7334 uint32_t numEffects = 0; 7335 effect_descriptor_t d; 7336 d.flags = 0; // prevent compiler warning 7337 bool found = false; 7338 7339 lStatus = EffectQueryNumberEffects(&numEffects); 7340 if (lStatus < 0) { 7341 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7342 goto Exit; 7343 } 7344 for (uint32_t i = 0; i < numEffects; i++) { 7345 lStatus = EffectQueryEffect(i, &desc); 7346 if (lStatus < 0) { 7347 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7348 continue; 7349 } 7350 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7351 // If matching type found save effect descriptor. If the session is 7352 // 0 and the effect is not auxiliary, continue enumeration in case 7353 // an auxiliary version of this effect type is available 7354 found = true; 7355 d = desc; 7356 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7357 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7358 break; 7359 } 7360 } 7361 } 7362 if (!found) { 7363 lStatus = BAD_VALUE; 7364 ALOGW("createEffect() effect not found"); 7365 goto Exit; 7366 } 7367 // For same effect type, chose auxiliary version over insert version if 7368 // connect to output mix (Compliance to OpenSL ES) 7369 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7370 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7371 desc = d; 7372 } 7373 } 7374 7375 // Do not allow auxiliary effects on a session different from 0 (output mix) 7376 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7377 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7378 lStatus = INVALID_OPERATION; 7379 goto Exit; 7380 } 7381 7382 // check recording permission for visualizer 7383 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7384 !recordingAllowed()) { 7385 lStatus = PERMISSION_DENIED; 7386 goto Exit; 7387 } 7388 7389 // return effect descriptor 7390 *pDesc = desc; 7391 7392 // If output is not specified try to find a matching audio session ID in one of the 7393 // output threads. 7394 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7395 // because of code checking output when entering the function. 7396 // Note: io is never 0 when creating an effect on an input 7397 if (io == 0) { 7398 // look for the thread where the specified audio session is present 7399 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7400 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7401 io = mPlaybackThreads.keyAt(i); 7402 break; 7403 } 7404 } 7405 if (io == 0) { 7406 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7407 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7408 io = mRecordThreads.keyAt(i); 7409 break; 7410 } 7411 } 7412 } 7413 // If no output thread contains the requested session ID, default to 7414 // first output. The effect chain will be moved to the correct output 7415 // thread when a track with the same session ID is created 7416 if (io == 0 && mPlaybackThreads.size()) { 7417 io = mPlaybackThreads.keyAt(0); 7418 } 7419 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7420 } 7421 ThreadBase *thread = checkRecordThread_l(io); 7422 if (thread == NULL) { 7423 thread = checkPlaybackThread_l(io); 7424 if (thread == NULL) { 7425 ALOGE("createEffect() unknown output thread"); 7426 lStatus = BAD_VALUE; 7427 goto Exit; 7428 } 7429 } 7430 7431 sp<Client> client = registerPid_l(pid); 7432 7433 // create effect on selected output thread 7434 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7435 &desc, enabled, &lStatus); 7436 if (handle != 0 && id != NULL) { 7437 *id = handle->id(); 7438 } 7439 } 7440 7441Exit: 7442 if (status != NULL) { 7443 *status = lStatus; 7444 } 7445 return handle; 7446} 7447 7448status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7449 audio_io_handle_t dstOutput) 7450{ 7451 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7452 sessionId, srcOutput, dstOutput); 7453 Mutex::Autolock _l(mLock); 7454 if (srcOutput == dstOutput) { 7455 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7456 return NO_ERROR; 7457 } 7458 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7459 if (srcThread == NULL) { 7460 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7461 return BAD_VALUE; 7462 } 7463 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7464 if (dstThread == NULL) { 7465 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7466 return BAD_VALUE; 7467 } 7468 7469 Mutex::Autolock _dl(dstThread->mLock); 7470 Mutex::Autolock _sl(srcThread->mLock); 7471 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7472 7473 return NO_ERROR; 7474} 7475 7476// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7477status_t AudioFlinger::moveEffectChain_l(int sessionId, 7478 AudioFlinger::PlaybackThread *srcThread, 7479 AudioFlinger::PlaybackThread *dstThread, 7480 bool reRegister) 7481{ 7482 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7483 sessionId, srcThread, dstThread); 7484 7485 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7486 if (chain == 0) { 7487 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7488 sessionId, srcThread); 7489 return INVALID_OPERATION; 7490 } 7491 7492 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7493 // so that a new chain is created with correct parameters when first effect is added. This is 7494 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7495 // removed. 7496 srcThread->removeEffectChain_l(chain); 7497 7498 // transfer all effects one by one so that new effect chain is created on new thread with 7499 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7500 audio_io_handle_t dstOutput = dstThread->id(); 7501 sp<EffectChain> dstChain; 7502 uint32_t strategy = 0; // prevent compiler warning 7503 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7504 while (effect != 0) { 7505 srcThread->removeEffect_l(effect); 7506 dstThread->addEffect_l(effect); 7507 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7508 if (effect->state() == EffectModule::ACTIVE || 7509 effect->state() == EffectModule::STOPPING) { 7510 effect->start(); 7511 } 7512 // if the move request is not received from audio policy manager, the effect must be 7513 // re-registered with the new strategy and output 7514 if (dstChain == 0) { 7515 dstChain = effect->chain().promote(); 7516 if (dstChain == 0) { 7517 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7518 srcThread->addEffect_l(effect); 7519 return NO_INIT; 7520 } 7521 strategy = dstChain->strategy(); 7522 } 7523 if (reRegister) { 7524 AudioSystem::unregisterEffect(effect->id()); 7525 AudioSystem::registerEffect(&effect->desc(), 7526 dstOutput, 7527 strategy, 7528 sessionId, 7529 effect->id()); 7530 } 7531 effect = chain->getEffectFromId_l(0); 7532 } 7533 7534 return NO_ERROR; 7535} 7536 7537 7538// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7539sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7540 const sp<AudioFlinger::Client>& client, 7541 const sp<IEffectClient>& effectClient, 7542 int32_t priority, 7543 int sessionId, 7544 effect_descriptor_t *desc, 7545 int *enabled, 7546 status_t *status 7547 ) 7548{ 7549 sp<EffectModule> effect; 7550 sp<EffectHandle> handle; 7551 status_t lStatus; 7552 sp<EffectChain> chain; 7553 bool chainCreated = false; 7554 bool effectCreated = false; 7555 bool effectRegistered = false; 7556 7557 lStatus = initCheck(); 7558 if (lStatus != NO_ERROR) { 7559 ALOGW("createEffect_l() Audio driver not initialized."); 7560 goto Exit; 7561 } 7562 7563 // Do not allow effects with session ID 0 on direct output or duplicating threads 7564 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7565 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7566 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7567 desc->name, sessionId); 7568 lStatus = BAD_VALUE; 7569 goto Exit; 7570 } 7571 // Only Pre processor effects are allowed on input threads and only on input threads 7572 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7573 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7574 desc->name, desc->flags, mType); 7575 lStatus = BAD_VALUE; 7576 goto Exit; 7577 } 7578 7579 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7580 7581 { // scope for mLock 7582 Mutex::Autolock _l(mLock); 7583 7584 // check for existing effect chain with the requested audio session 7585 chain = getEffectChain_l(sessionId); 7586 if (chain == 0) { 7587 // create a new chain for this session 7588 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7589 chain = new EffectChain(this, sessionId); 7590 addEffectChain_l(chain); 7591 chain->setStrategy(getStrategyForSession_l(sessionId)); 7592 chainCreated = true; 7593 } else { 7594 effect = chain->getEffectFromDesc_l(desc); 7595 } 7596 7597 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7598 7599 if (effect == 0) { 7600 int id = mAudioFlinger->nextUniqueId(); 7601 // Check CPU and memory usage 7602 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7603 if (lStatus != NO_ERROR) { 7604 goto Exit; 7605 } 7606 effectRegistered = true; 7607 // create a new effect module if none present in the chain 7608 effect = new EffectModule(this, chain, desc, id, sessionId); 7609 lStatus = effect->status(); 7610 if (lStatus != NO_ERROR) { 7611 goto Exit; 7612 } 7613 lStatus = chain->addEffect_l(effect); 7614 if (lStatus != NO_ERROR) { 7615 goto Exit; 7616 } 7617 effectCreated = true; 7618 7619 effect->setDevice(mDevice); 7620 effect->setMode(mAudioFlinger->getMode()); 7621 } 7622 // create effect handle and connect it to effect module 7623 handle = new EffectHandle(effect, client, effectClient, priority); 7624 lStatus = effect->addHandle(handle.get()); 7625 if (enabled != NULL) { 7626 *enabled = (int)effect->isEnabled(); 7627 } 7628 } 7629 7630Exit: 7631 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7632 Mutex::Autolock _l(mLock); 7633 if (effectCreated) { 7634 chain->removeEffect_l(effect); 7635 } 7636 if (effectRegistered) { 7637 AudioSystem::unregisterEffect(effect->id()); 7638 } 7639 if (chainCreated) { 7640 removeEffectChain_l(chain); 7641 } 7642 handle.clear(); 7643 } 7644 7645 if (status != NULL) { 7646 *status = lStatus; 7647 } 7648 return handle; 7649} 7650 7651sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7652{ 7653 Mutex::Autolock _l(mLock); 7654 return getEffect_l(sessionId, effectId); 7655} 7656 7657sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7658{ 7659 sp<EffectChain> chain = getEffectChain_l(sessionId); 7660 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7661} 7662 7663// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7664// PlaybackThread::mLock held 7665status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7666{ 7667 // check for existing effect chain with the requested audio session 7668 int sessionId = effect->sessionId(); 7669 sp<EffectChain> chain = getEffectChain_l(sessionId); 7670 bool chainCreated = false; 7671 7672 if (chain == 0) { 7673 // create a new chain for this session 7674 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7675 chain = new EffectChain(this, sessionId); 7676 addEffectChain_l(chain); 7677 chain->setStrategy(getStrategyForSession_l(sessionId)); 7678 chainCreated = true; 7679 } 7680 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7681 7682 if (chain->getEffectFromId_l(effect->id()) != 0) { 7683 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7684 this, effect->desc().name, chain.get()); 7685 return BAD_VALUE; 7686 } 7687 7688 status_t status = chain->addEffect_l(effect); 7689 if (status != NO_ERROR) { 7690 if (chainCreated) { 7691 removeEffectChain_l(chain); 7692 } 7693 return status; 7694 } 7695 7696 effect->setDevice(mDevice); 7697 effect->setMode(mAudioFlinger->getMode()); 7698 return NO_ERROR; 7699} 7700 7701void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7702 7703 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7704 effect_descriptor_t desc = effect->desc(); 7705 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7706 detachAuxEffect_l(effect->id()); 7707 } 7708 7709 sp<EffectChain> chain = effect->chain().promote(); 7710 if (chain != 0) { 7711 // remove effect chain if removing last effect 7712 if (chain->removeEffect_l(effect) == 0) { 7713 removeEffectChain_l(chain); 7714 } 7715 } else { 7716 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7717 } 7718} 7719 7720void AudioFlinger::ThreadBase::lockEffectChains_l( 7721 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7722{ 7723 effectChains = mEffectChains; 7724 for (size_t i = 0; i < mEffectChains.size(); i++) { 7725 mEffectChains[i]->lock(); 7726 } 7727} 7728 7729void AudioFlinger::ThreadBase::unlockEffectChains( 7730 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7731{ 7732 for (size_t i = 0; i < effectChains.size(); i++) { 7733 effectChains[i]->unlock(); 7734 } 7735} 7736 7737sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7738{ 7739 Mutex::Autolock _l(mLock); 7740 return getEffectChain_l(sessionId); 7741} 7742 7743sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7744{ 7745 size_t size = mEffectChains.size(); 7746 for (size_t i = 0; i < size; i++) { 7747 if (mEffectChains[i]->sessionId() == sessionId) { 7748 return mEffectChains[i]; 7749 } 7750 } 7751 return 0; 7752} 7753 7754void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7755{ 7756 Mutex::Autolock _l(mLock); 7757 size_t size = mEffectChains.size(); 7758 for (size_t i = 0; i < size; i++) { 7759 mEffectChains[i]->setMode_l(mode); 7760 } 7761} 7762 7763void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7764 EffectHandle *handle, 7765 bool unpinIfLast) { 7766 7767 Mutex::Autolock _l(mLock); 7768 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7769 // delete the effect module if removing last handle on it 7770 if (effect->removeHandle(handle) == 0) { 7771 if (!effect->isPinned() || unpinIfLast) { 7772 removeEffect_l(effect); 7773 AudioSystem::unregisterEffect(effect->id()); 7774 } 7775 } 7776} 7777 7778status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7779{ 7780 int session = chain->sessionId(); 7781 int16_t *buffer = mMixBuffer; 7782 bool ownsBuffer = false; 7783 7784 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7785 if (session > 0) { 7786 // Only one effect chain can be present in direct output thread and it uses 7787 // the mix buffer as input 7788 if (mType != DIRECT) { 7789 size_t numSamples = mNormalFrameCount * mChannelCount; 7790 buffer = new int16_t[numSamples]; 7791 memset(buffer, 0, numSamples * sizeof(int16_t)); 7792 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7793 ownsBuffer = true; 7794 } 7795 7796 // Attach all tracks with same session ID to this chain. 7797 for (size_t i = 0; i < mTracks.size(); ++i) { 7798 sp<Track> track = mTracks[i]; 7799 if (session == track->sessionId()) { 7800 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7801 track->setMainBuffer(buffer); 7802 chain->incTrackCnt(); 7803 } 7804 } 7805 7806 // indicate all active tracks in the chain 7807 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7808 sp<Track> track = mActiveTracks[i].promote(); 7809 if (track == 0) continue; 7810 if (session == track->sessionId()) { 7811 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7812 chain->incActiveTrackCnt(); 7813 } 7814 } 7815 } 7816 7817 chain->setInBuffer(buffer, ownsBuffer); 7818 chain->setOutBuffer(mMixBuffer); 7819 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7820 // chains list in order to be processed last as it contains output stage effects 7821 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7822 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7823 // after track specific effects and before output stage 7824 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7825 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7826 // Effect chain for other sessions are inserted at beginning of effect 7827 // chains list to be processed before output mix effects. Relative order between other 7828 // sessions is not important 7829 size_t size = mEffectChains.size(); 7830 size_t i = 0; 7831 for (i = 0; i < size; i++) { 7832 if (mEffectChains[i]->sessionId() < session) break; 7833 } 7834 mEffectChains.insertAt(chain, i); 7835 checkSuspendOnAddEffectChain_l(chain); 7836 7837 return NO_ERROR; 7838} 7839 7840size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7841{ 7842 int session = chain->sessionId(); 7843 7844 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7845 7846 for (size_t i = 0; i < mEffectChains.size(); i++) { 7847 if (chain == mEffectChains[i]) { 7848 mEffectChains.removeAt(i); 7849 // detach all active tracks from the chain 7850 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7851 sp<Track> track = mActiveTracks[i].promote(); 7852 if (track == 0) continue; 7853 if (session == track->sessionId()) { 7854 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7855 chain.get(), session); 7856 chain->decActiveTrackCnt(); 7857 } 7858 } 7859 7860 // detach all tracks with same session ID from this chain 7861 for (size_t i = 0; i < mTracks.size(); ++i) { 7862 sp<Track> track = mTracks[i]; 7863 if (session == track->sessionId()) { 7864 track->setMainBuffer(mMixBuffer); 7865 chain->decTrackCnt(); 7866 } 7867 } 7868 break; 7869 } 7870 } 7871 return mEffectChains.size(); 7872} 7873 7874status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7875 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7876{ 7877 Mutex::Autolock _l(mLock); 7878 return attachAuxEffect_l(track, EffectId); 7879} 7880 7881status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7882 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7883{ 7884 status_t status = NO_ERROR; 7885 7886 if (EffectId == 0) { 7887 track->setAuxBuffer(0, NULL); 7888 } else { 7889 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7890 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7891 if (effect != 0) { 7892 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7893 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7894 } else { 7895 status = INVALID_OPERATION; 7896 } 7897 } else { 7898 status = BAD_VALUE; 7899 } 7900 } 7901 return status; 7902} 7903 7904void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7905{ 7906 for (size_t i = 0; i < mTracks.size(); ++i) { 7907 sp<Track> track = mTracks[i]; 7908 if (track->auxEffectId() == effectId) { 7909 attachAuxEffect_l(track, 0); 7910 } 7911 } 7912} 7913 7914status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7915{ 7916 // only one chain per input thread 7917 if (mEffectChains.size() != 0) { 7918 return INVALID_OPERATION; 7919 } 7920 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7921 7922 chain->setInBuffer(NULL); 7923 chain->setOutBuffer(NULL); 7924 7925 checkSuspendOnAddEffectChain_l(chain); 7926 7927 mEffectChains.add(chain); 7928 7929 return NO_ERROR; 7930} 7931 7932size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7933{ 7934 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7935 ALOGW_IF(mEffectChains.size() != 1, 7936 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7937 chain.get(), mEffectChains.size(), this); 7938 if (mEffectChains.size() == 1) { 7939 mEffectChains.removeAt(0); 7940 } 7941 return 0; 7942} 7943 7944// ---------------------------------------------------------------------------- 7945// EffectModule implementation 7946// ---------------------------------------------------------------------------- 7947 7948#undef LOG_TAG 7949#define LOG_TAG "AudioFlinger::EffectModule" 7950 7951AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7952 const wp<AudioFlinger::EffectChain>& chain, 7953 effect_descriptor_t *desc, 7954 int id, 7955 int sessionId) 7956 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 7957 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 7958 // mDescriptor is set below 7959 // mConfig is set by configure() and not used before then 7960 mEffectInterface(NULL), 7961 mStatus(NO_INIT), mState(IDLE), 7962 // mMaxDisableWaitCnt is set by configure() and not used before then 7963 // mDisableWaitCnt is set by process() and updateState() and not used before then 7964 mSuspended(false) 7965{ 7966 ALOGV("Constructor %p", this); 7967 int lStatus; 7968 if (thread == NULL) { 7969 return; 7970 } 7971 7972 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7973 7974 // create effect engine from effect factory 7975 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7976 7977 if (mStatus != NO_ERROR) { 7978 return; 7979 } 7980 lStatus = init(); 7981 if (lStatus < 0) { 7982 mStatus = lStatus; 7983 goto Error; 7984 } 7985 7986 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7987 return; 7988Error: 7989 EffectRelease(mEffectInterface); 7990 mEffectInterface = NULL; 7991 ALOGV("Constructor Error %d", mStatus); 7992} 7993 7994AudioFlinger::EffectModule::~EffectModule() 7995{ 7996 ALOGV("Destructor %p", this); 7997 if (mEffectInterface != NULL) { 7998 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7999 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8000 sp<ThreadBase> thread = mThread.promote(); 8001 if (thread != 0) { 8002 audio_stream_t *stream = thread->stream(); 8003 if (stream != NULL) { 8004 stream->remove_audio_effect(stream, mEffectInterface); 8005 } 8006 } 8007 } 8008 // release effect engine 8009 EffectRelease(mEffectInterface); 8010 } 8011} 8012 8013status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8014{ 8015 status_t status; 8016 8017 Mutex::Autolock _l(mLock); 8018 int priority = handle->priority(); 8019 size_t size = mHandles.size(); 8020 EffectHandle *controlHandle = NULL; 8021 size_t i; 8022 for (i = 0; i < size; i++) { 8023 EffectHandle *h = mHandles[i]; 8024 if (h == NULL || h->destroyed_l()) continue; 8025 // first non destroyed handle is considered in control 8026 if (controlHandle == NULL) 8027 controlHandle = h; 8028 if (h->priority() <= priority) break; 8029 } 8030 // if inserted in first place, move effect control from previous owner to this handle 8031 if (i == 0) { 8032 bool enabled = false; 8033 if (controlHandle != NULL) { 8034 enabled = controlHandle->enabled(); 8035 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8036 } 8037 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8038 status = NO_ERROR; 8039 } else { 8040 status = ALREADY_EXISTS; 8041 } 8042 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8043 mHandles.insertAt(handle, i); 8044 return status; 8045} 8046 8047size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8048{ 8049 Mutex::Autolock _l(mLock); 8050 size_t size = mHandles.size(); 8051 size_t i; 8052 for (i = 0; i < size; i++) { 8053 if (mHandles[i] == handle) break; 8054 } 8055 if (i == size) { 8056 return size; 8057 } 8058 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8059 8060 mHandles.removeAt(i); 8061 // if removed from first place, move effect control from this handle to next in line 8062 if (i == 0) { 8063 EffectHandle *h = controlHandle_l(); 8064 if (h != NULL) { 8065 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8066 } 8067 } 8068 8069 // Prevent calls to process() and other functions on effect interface from now on. 8070 // The effect engine will be released by the destructor when the last strong reference on 8071 // this object is released which can happen after next process is called. 8072 if (mHandles.size() == 0 && !mPinned) { 8073 mState = DESTROYED; 8074 } 8075 8076 return size; 8077} 8078 8079// must be called with EffectModule::mLock held 8080AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8081{ 8082 // the first valid handle in the list has control over the module 8083 for (size_t i = 0; i < mHandles.size(); i++) { 8084 EffectHandle *h = mHandles[i]; 8085 if (h != NULL && !h->destroyed_l()) { 8086 return h; 8087 } 8088 } 8089 8090 return NULL; 8091} 8092 8093size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8094{ 8095 ALOGV("disconnect() %p handle %p", this, handle); 8096 // keep a strong reference on this EffectModule to avoid calling the 8097 // destructor before we exit 8098 sp<EffectModule> keep(this); 8099 { 8100 sp<ThreadBase> thread = mThread.promote(); 8101 if (thread != 0) { 8102 thread->disconnectEffect(keep, handle, unpinIfLast); 8103 } 8104 } 8105 return mHandles.size(); 8106} 8107 8108void AudioFlinger::EffectModule::updateState() { 8109 Mutex::Autolock _l(mLock); 8110 8111 switch (mState) { 8112 case RESTART: 8113 reset_l(); 8114 // FALL THROUGH 8115 8116 case STARTING: 8117 // clear auxiliary effect input buffer for next accumulation 8118 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8119 memset(mConfig.inputCfg.buffer.raw, 8120 0, 8121 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8122 } 8123 start_l(); 8124 mState = ACTIVE; 8125 break; 8126 case STOPPING: 8127 stop_l(); 8128 mDisableWaitCnt = mMaxDisableWaitCnt; 8129 mState = STOPPED; 8130 break; 8131 case STOPPED: 8132 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8133 // turn off sequence. 8134 if (--mDisableWaitCnt == 0) { 8135 reset_l(); 8136 mState = IDLE; 8137 } 8138 break; 8139 default: //IDLE , ACTIVE, DESTROYED 8140 break; 8141 } 8142} 8143 8144void AudioFlinger::EffectModule::process() 8145{ 8146 Mutex::Autolock _l(mLock); 8147 8148 if (mState == DESTROYED || mEffectInterface == NULL || 8149 mConfig.inputCfg.buffer.raw == NULL || 8150 mConfig.outputCfg.buffer.raw == NULL) { 8151 return; 8152 } 8153 8154 if (isProcessEnabled()) { 8155 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8156 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8157 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8158 mConfig.inputCfg.buffer.s32, 8159 mConfig.inputCfg.buffer.frameCount/2); 8160 } 8161 8162 // do the actual processing in the effect engine 8163 int ret = (*mEffectInterface)->process(mEffectInterface, 8164 &mConfig.inputCfg.buffer, 8165 &mConfig.outputCfg.buffer); 8166 8167 // force transition to IDLE state when engine is ready 8168 if (mState == STOPPED && ret == -ENODATA) { 8169 mDisableWaitCnt = 1; 8170 } 8171 8172 // clear auxiliary effect input buffer for next accumulation 8173 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8174 memset(mConfig.inputCfg.buffer.raw, 0, 8175 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8176 } 8177 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8178 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8179 // If an insert effect is idle and input buffer is different from output buffer, 8180 // accumulate input onto output 8181 sp<EffectChain> chain = mChain.promote(); 8182 if (chain != 0 && chain->activeTrackCnt() != 0) { 8183 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8184 int16_t *in = mConfig.inputCfg.buffer.s16; 8185 int16_t *out = mConfig.outputCfg.buffer.s16; 8186 for (size_t i = 0; i < frameCnt; i++) { 8187 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8188 } 8189 } 8190 } 8191} 8192 8193void AudioFlinger::EffectModule::reset_l() 8194{ 8195 if (mEffectInterface == NULL) { 8196 return; 8197 } 8198 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8199} 8200 8201status_t AudioFlinger::EffectModule::configure() 8202{ 8203 if (mEffectInterface == NULL) { 8204 return NO_INIT; 8205 } 8206 8207 sp<ThreadBase> thread = mThread.promote(); 8208 if (thread == 0) { 8209 return DEAD_OBJECT; 8210 } 8211 8212 // TODO: handle configuration of effects replacing track process 8213 audio_channel_mask_t channelMask = thread->channelMask(); 8214 8215 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8216 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8217 } else { 8218 mConfig.inputCfg.channels = channelMask; 8219 } 8220 mConfig.outputCfg.channels = channelMask; 8221 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8222 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8223 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8224 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8225 mConfig.inputCfg.bufferProvider.cookie = NULL; 8226 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8227 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8228 mConfig.outputCfg.bufferProvider.cookie = NULL; 8229 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8230 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8231 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8232 // Insert effect: 8233 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8234 // always overwrites output buffer: input buffer == output buffer 8235 // - in other sessions: 8236 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8237 // other effect: overwrites output buffer: input buffer == output buffer 8238 // Auxiliary effect: 8239 // accumulates in output buffer: input buffer != output buffer 8240 // Therefore: accumulate <=> input buffer != output buffer 8241 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8242 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8243 } else { 8244 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8245 } 8246 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8247 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8248 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8249 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8250 8251 ALOGV("configure() %p thread %p buffer %p framecount %d", 8252 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8253 8254 status_t cmdStatus; 8255 uint32_t size = sizeof(int); 8256 status_t status = (*mEffectInterface)->command(mEffectInterface, 8257 EFFECT_CMD_SET_CONFIG, 8258 sizeof(effect_config_t), 8259 &mConfig, 8260 &size, 8261 &cmdStatus); 8262 if (status == 0) { 8263 status = cmdStatus; 8264 } 8265 8266 if (status == 0 && 8267 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8268 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8269 effect_param_t *p = (effect_param_t *)buf32; 8270 8271 p->psize = sizeof(uint32_t); 8272 p->vsize = sizeof(uint32_t); 8273 size = sizeof(int); 8274 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8275 8276 uint32_t latency = 0; 8277 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8278 if (pbt != NULL) { 8279 latency = pbt->latency_l(); 8280 } 8281 8282 *((int32_t *)p->data + 1)= latency; 8283 (*mEffectInterface)->command(mEffectInterface, 8284 EFFECT_CMD_SET_PARAM, 8285 sizeof(effect_param_t) + 8, 8286 &buf32, 8287 &size, 8288 &cmdStatus); 8289 } 8290 8291 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8292 (1000 * mConfig.outputCfg.buffer.frameCount); 8293 8294 return status; 8295} 8296 8297status_t AudioFlinger::EffectModule::init() 8298{ 8299 Mutex::Autolock _l(mLock); 8300 if (mEffectInterface == NULL) { 8301 return NO_INIT; 8302 } 8303 status_t cmdStatus; 8304 uint32_t size = sizeof(status_t); 8305 status_t status = (*mEffectInterface)->command(mEffectInterface, 8306 EFFECT_CMD_INIT, 8307 0, 8308 NULL, 8309 &size, 8310 &cmdStatus); 8311 if (status == 0) { 8312 status = cmdStatus; 8313 } 8314 return status; 8315} 8316 8317status_t AudioFlinger::EffectModule::start() 8318{ 8319 Mutex::Autolock _l(mLock); 8320 return start_l(); 8321} 8322 8323status_t AudioFlinger::EffectModule::start_l() 8324{ 8325 if (mEffectInterface == NULL) { 8326 return NO_INIT; 8327 } 8328 status_t cmdStatus; 8329 uint32_t size = sizeof(status_t); 8330 status_t status = (*mEffectInterface)->command(mEffectInterface, 8331 EFFECT_CMD_ENABLE, 8332 0, 8333 NULL, 8334 &size, 8335 &cmdStatus); 8336 if (status == 0) { 8337 status = cmdStatus; 8338 } 8339 if (status == 0 && 8340 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8341 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8342 sp<ThreadBase> thread = mThread.promote(); 8343 if (thread != 0) { 8344 audio_stream_t *stream = thread->stream(); 8345 if (stream != NULL) { 8346 stream->add_audio_effect(stream, mEffectInterface); 8347 } 8348 } 8349 } 8350 return status; 8351} 8352 8353status_t AudioFlinger::EffectModule::stop() 8354{ 8355 Mutex::Autolock _l(mLock); 8356 return stop_l(); 8357} 8358 8359status_t AudioFlinger::EffectModule::stop_l() 8360{ 8361 if (mEffectInterface == NULL) { 8362 return NO_INIT; 8363 } 8364 status_t cmdStatus; 8365 uint32_t size = sizeof(status_t); 8366 status_t status = (*mEffectInterface)->command(mEffectInterface, 8367 EFFECT_CMD_DISABLE, 8368 0, 8369 NULL, 8370 &size, 8371 &cmdStatus); 8372 if (status == 0) { 8373 status = cmdStatus; 8374 } 8375 if (status == 0 && 8376 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8377 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8378 sp<ThreadBase> thread = mThread.promote(); 8379 if (thread != 0) { 8380 audio_stream_t *stream = thread->stream(); 8381 if (stream != NULL) { 8382 stream->remove_audio_effect(stream, mEffectInterface); 8383 } 8384 } 8385 } 8386 return status; 8387} 8388 8389status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8390 uint32_t cmdSize, 8391 void *pCmdData, 8392 uint32_t *replySize, 8393 void *pReplyData) 8394{ 8395 Mutex::Autolock _l(mLock); 8396// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8397 8398 if (mState == DESTROYED || mEffectInterface == NULL) { 8399 return NO_INIT; 8400 } 8401 status_t status = (*mEffectInterface)->command(mEffectInterface, 8402 cmdCode, 8403 cmdSize, 8404 pCmdData, 8405 replySize, 8406 pReplyData); 8407 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8408 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8409 for (size_t i = 1; i < mHandles.size(); i++) { 8410 EffectHandle *h = mHandles[i]; 8411 if (h != NULL && !h->destroyed_l()) { 8412 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8413 } 8414 } 8415 } 8416 return status; 8417} 8418 8419status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8420{ 8421 Mutex::Autolock _l(mLock); 8422 return setEnabled_l(enabled); 8423} 8424 8425// must be called with EffectModule::mLock held 8426status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8427{ 8428 8429 ALOGV("setEnabled %p enabled %d", this, enabled); 8430 8431 if (enabled != isEnabled()) { 8432 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8433 if (enabled && status != NO_ERROR) { 8434 return status; 8435 } 8436 8437 switch (mState) { 8438 // going from disabled to enabled 8439 case IDLE: 8440 mState = STARTING; 8441 break; 8442 case STOPPED: 8443 mState = RESTART; 8444 break; 8445 case STOPPING: 8446 mState = ACTIVE; 8447 break; 8448 8449 // going from enabled to disabled 8450 case RESTART: 8451 mState = STOPPED; 8452 break; 8453 case STARTING: 8454 mState = IDLE; 8455 break; 8456 case ACTIVE: 8457 mState = STOPPING; 8458 break; 8459 case DESTROYED: 8460 return NO_ERROR; // simply ignore as we are being destroyed 8461 } 8462 for (size_t i = 1; i < mHandles.size(); i++) { 8463 EffectHandle *h = mHandles[i]; 8464 if (h != NULL && !h->destroyed_l()) { 8465 h->setEnabled(enabled); 8466 } 8467 } 8468 } 8469 return NO_ERROR; 8470} 8471 8472bool AudioFlinger::EffectModule::isEnabled() const 8473{ 8474 switch (mState) { 8475 case RESTART: 8476 case STARTING: 8477 case ACTIVE: 8478 return true; 8479 case IDLE: 8480 case STOPPING: 8481 case STOPPED: 8482 case DESTROYED: 8483 default: 8484 return false; 8485 } 8486} 8487 8488bool AudioFlinger::EffectModule::isProcessEnabled() const 8489{ 8490 switch (mState) { 8491 case RESTART: 8492 case ACTIVE: 8493 case STOPPING: 8494 case STOPPED: 8495 return true; 8496 case IDLE: 8497 case STARTING: 8498 case DESTROYED: 8499 default: 8500 return false; 8501 } 8502} 8503 8504status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8505{ 8506 Mutex::Autolock _l(mLock); 8507 status_t status = NO_ERROR; 8508 8509 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8510 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8511 if (isProcessEnabled() && 8512 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8513 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8514 status_t cmdStatus; 8515 uint32_t volume[2]; 8516 uint32_t *pVolume = NULL; 8517 uint32_t size = sizeof(volume); 8518 volume[0] = *left; 8519 volume[1] = *right; 8520 if (controller) { 8521 pVolume = volume; 8522 } 8523 status = (*mEffectInterface)->command(mEffectInterface, 8524 EFFECT_CMD_SET_VOLUME, 8525 size, 8526 volume, 8527 &size, 8528 pVolume); 8529 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8530 *left = volume[0]; 8531 *right = volume[1]; 8532 } 8533 } 8534 return status; 8535} 8536 8537status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8538{ 8539 Mutex::Autolock _l(mLock); 8540 status_t status = NO_ERROR; 8541 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8542 // audio pre processing modules on RecordThread can receive both output and 8543 // input device indication in the same call 8544 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8545 if (dev) { 8546 status_t cmdStatus; 8547 uint32_t size = sizeof(status_t); 8548 8549 status = (*mEffectInterface)->command(mEffectInterface, 8550 EFFECT_CMD_SET_DEVICE, 8551 sizeof(uint32_t), 8552 &dev, 8553 &size, 8554 &cmdStatus); 8555 if (status == NO_ERROR) { 8556 status = cmdStatus; 8557 } 8558 } 8559 dev = device & AUDIO_DEVICE_IN_ALL; 8560 if (dev) { 8561 status_t cmdStatus; 8562 uint32_t size = sizeof(status_t); 8563 8564 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8565 EFFECT_CMD_SET_INPUT_DEVICE, 8566 sizeof(uint32_t), 8567 &dev, 8568 &size, 8569 &cmdStatus); 8570 if (status2 == NO_ERROR) { 8571 status2 = cmdStatus; 8572 } 8573 if (status == NO_ERROR) { 8574 status = status2; 8575 } 8576 } 8577 } 8578 return status; 8579} 8580 8581status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8582{ 8583 Mutex::Autolock _l(mLock); 8584 status_t status = NO_ERROR; 8585 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8586 status_t cmdStatus; 8587 uint32_t size = sizeof(status_t); 8588 status = (*mEffectInterface)->command(mEffectInterface, 8589 EFFECT_CMD_SET_AUDIO_MODE, 8590 sizeof(audio_mode_t), 8591 &mode, 8592 &size, 8593 &cmdStatus); 8594 if (status == NO_ERROR) { 8595 status = cmdStatus; 8596 } 8597 } 8598 return status; 8599} 8600 8601void AudioFlinger::EffectModule::setSuspended(bool suspended) 8602{ 8603 Mutex::Autolock _l(mLock); 8604 mSuspended = suspended; 8605} 8606 8607bool AudioFlinger::EffectModule::suspended() const 8608{ 8609 Mutex::Autolock _l(mLock); 8610 return mSuspended; 8611} 8612 8613bool AudioFlinger::EffectModule::purgeHandles() 8614{ 8615 bool enabled = false; 8616 Mutex::Autolock _l(mLock); 8617 for (size_t i = 0; i < mHandles.size(); i++) { 8618 EffectHandle *handle = mHandles[i]; 8619 if (handle != NULL && !handle->destroyed_l()) { 8620 handle->effect().clear(); 8621 if (handle->hasControl()) { 8622 enabled = handle->enabled(); 8623 } 8624 } 8625 } 8626 return enabled; 8627} 8628 8629status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8630{ 8631 const size_t SIZE = 256; 8632 char buffer[SIZE]; 8633 String8 result; 8634 8635 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8636 result.append(buffer); 8637 8638 bool locked = tryLock(mLock); 8639 // failed to lock - AudioFlinger is probably deadlocked 8640 if (!locked) { 8641 result.append("\t\tCould not lock Fx mutex:\n"); 8642 } 8643 8644 result.append("\t\tSession Status State Engine:\n"); 8645 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8646 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8647 result.append(buffer); 8648 8649 result.append("\t\tDescriptor:\n"); 8650 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8651 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8652 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8653 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8654 result.append(buffer); 8655 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8656 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8657 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8658 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8659 result.append(buffer); 8660 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8661 mDescriptor.apiVersion, 8662 mDescriptor.flags); 8663 result.append(buffer); 8664 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8665 mDescriptor.name); 8666 result.append(buffer); 8667 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8668 mDescriptor.implementor); 8669 result.append(buffer); 8670 8671 result.append("\t\t- Input configuration:\n"); 8672 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8673 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8674 (uint32_t)mConfig.inputCfg.buffer.raw, 8675 mConfig.inputCfg.buffer.frameCount, 8676 mConfig.inputCfg.samplingRate, 8677 mConfig.inputCfg.channels, 8678 mConfig.inputCfg.format); 8679 result.append(buffer); 8680 8681 result.append("\t\t- Output configuration:\n"); 8682 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8683 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8684 (uint32_t)mConfig.outputCfg.buffer.raw, 8685 mConfig.outputCfg.buffer.frameCount, 8686 mConfig.outputCfg.samplingRate, 8687 mConfig.outputCfg.channels, 8688 mConfig.outputCfg.format); 8689 result.append(buffer); 8690 8691 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8692 result.append(buffer); 8693 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8694 for (size_t i = 0; i < mHandles.size(); ++i) { 8695 EffectHandle *handle = mHandles[i]; 8696 if (handle != NULL && !handle->destroyed_l()) { 8697 handle->dump(buffer, SIZE); 8698 result.append(buffer); 8699 } 8700 } 8701 8702 result.append("\n"); 8703 8704 write(fd, result.string(), result.length()); 8705 8706 if (locked) { 8707 mLock.unlock(); 8708 } 8709 8710 return NO_ERROR; 8711} 8712 8713// ---------------------------------------------------------------------------- 8714// EffectHandle implementation 8715// ---------------------------------------------------------------------------- 8716 8717#undef LOG_TAG 8718#define LOG_TAG "AudioFlinger::EffectHandle" 8719 8720AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8721 const sp<AudioFlinger::Client>& client, 8722 const sp<IEffectClient>& effectClient, 8723 int32_t priority) 8724 : BnEffect(), 8725 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8726 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8727{ 8728 ALOGV("constructor %p", this); 8729 8730 if (client == 0) { 8731 return; 8732 } 8733 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8734 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8735 if (mCblkMemory != 0) { 8736 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8737 8738 if (mCblk != NULL) { 8739 new(mCblk) effect_param_cblk_t(); 8740 mBuffer = (uint8_t *)mCblk + bufOffset; 8741 } 8742 } else { 8743 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8744 return; 8745 } 8746} 8747 8748AudioFlinger::EffectHandle::~EffectHandle() 8749{ 8750 ALOGV("Destructor %p", this); 8751 8752 if (mEffect == 0) { 8753 mDestroyed = true; 8754 return; 8755 } 8756 mEffect->lock(); 8757 mDestroyed = true; 8758 mEffect->unlock(); 8759 disconnect(false); 8760} 8761 8762status_t AudioFlinger::EffectHandle::enable() 8763{ 8764 ALOGV("enable %p", this); 8765 if (!mHasControl) return INVALID_OPERATION; 8766 if (mEffect == 0) return DEAD_OBJECT; 8767 8768 if (mEnabled) { 8769 return NO_ERROR; 8770 } 8771 8772 mEnabled = true; 8773 8774 sp<ThreadBase> thread = mEffect->thread().promote(); 8775 if (thread != 0) { 8776 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8777 } 8778 8779 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8780 if (mEffect->suspended()) { 8781 return NO_ERROR; 8782 } 8783 8784 status_t status = mEffect->setEnabled(true); 8785 if (status != NO_ERROR) { 8786 if (thread != 0) { 8787 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8788 } 8789 mEnabled = false; 8790 } 8791 return status; 8792} 8793 8794status_t AudioFlinger::EffectHandle::disable() 8795{ 8796 ALOGV("disable %p", this); 8797 if (!mHasControl) return INVALID_OPERATION; 8798 if (mEffect == 0) return DEAD_OBJECT; 8799 8800 if (!mEnabled) { 8801 return NO_ERROR; 8802 } 8803 mEnabled = false; 8804 8805 if (mEffect->suspended()) { 8806 return NO_ERROR; 8807 } 8808 8809 status_t status = mEffect->setEnabled(false); 8810 8811 sp<ThreadBase> thread = mEffect->thread().promote(); 8812 if (thread != 0) { 8813 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8814 } 8815 8816 return status; 8817} 8818 8819void AudioFlinger::EffectHandle::disconnect() 8820{ 8821 disconnect(true); 8822} 8823 8824void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8825{ 8826 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8827 if (mEffect == 0) { 8828 return; 8829 } 8830 // restore suspended effects if the disconnected handle was enabled and the last one. 8831 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8832 sp<ThreadBase> thread = mEffect->thread().promote(); 8833 if (thread != 0) { 8834 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8835 } 8836 } 8837 8838 // release sp on module => module destructor can be called now 8839 mEffect.clear(); 8840 if (mClient != 0) { 8841 if (mCblk != NULL) { 8842 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8843 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8844 } 8845 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8846 // Client destructor must run with AudioFlinger mutex locked 8847 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8848 mClient.clear(); 8849 } 8850} 8851 8852status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8853 uint32_t cmdSize, 8854 void *pCmdData, 8855 uint32_t *replySize, 8856 void *pReplyData) 8857{ 8858// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8859// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8860 8861 // only get parameter command is permitted for applications not controlling the effect 8862 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8863 return INVALID_OPERATION; 8864 } 8865 if (mEffect == 0) return DEAD_OBJECT; 8866 if (mClient == 0) return INVALID_OPERATION; 8867 8868 // handle commands that are not forwarded transparently to effect engine 8869 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8870 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8871 // no risk to block the whole media server process or mixer threads is we are stuck here 8872 Mutex::Autolock _l(mCblk->lock); 8873 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8874 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8875 mCblk->serverIndex = 0; 8876 mCblk->clientIndex = 0; 8877 return BAD_VALUE; 8878 } 8879 status_t status = NO_ERROR; 8880 while (mCblk->serverIndex < mCblk->clientIndex) { 8881 int reply; 8882 uint32_t rsize = sizeof(int); 8883 int *p = (int *)(mBuffer + mCblk->serverIndex); 8884 int size = *p++; 8885 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8886 ALOGW("command(): invalid parameter block size"); 8887 break; 8888 } 8889 effect_param_t *param = (effect_param_t *)p; 8890 if (param->psize == 0 || param->vsize == 0) { 8891 ALOGW("command(): null parameter or value size"); 8892 mCblk->serverIndex += size; 8893 continue; 8894 } 8895 uint32_t psize = sizeof(effect_param_t) + 8896 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8897 param->vsize; 8898 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8899 psize, 8900 p, 8901 &rsize, 8902 &reply); 8903 // stop at first error encountered 8904 if (ret != NO_ERROR) { 8905 status = ret; 8906 *(int *)pReplyData = reply; 8907 break; 8908 } else if (reply != NO_ERROR) { 8909 *(int *)pReplyData = reply; 8910 break; 8911 } 8912 mCblk->serverIndex += size; 8913 } 8914 mCblk->serverIndex = 0; 8915 mCblk->clientIndex = 0; 8916 return status; 8917 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8918 *(int *)pReplyData = NO_ERROR; 8919 return enable(); 8920 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8921 *(int *)pReplyData = NO_ERROR; 8922 return disable(); 8923 } 8924 8925 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8926} 8927 8928void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8929{ 8930 ALOGV("setControl %p control %d", this, hasControl); 8931 8932 mHasControl = hasControl; 8933 mEnabled = enabled; 8934 8935 if (signal && mEffectClient != 0) { 8936 mEffectClient->controlStatusChanged(hasControl); 8937 } 8938} 8939 8940void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8941 uint32_t cmdSize, 8942 void *pCmdData, 8943 uint32_t replySize, 8944 void *pReplyData) 8945{ 8946 if (mEffectClient != 0) { 8947 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8948 } 8949} 8950 8951 8952 8953void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8954{ 8955 if (mEffectClient != 0) { 8956 mEffectClient->enableStatusChanged(enabled); 8957 } 8958} 8959 8960status_t AudioFlinger::EffectHandle::onTransact( 8961 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8962{ 8963 return BnEffect::onTransact(code, data, reply, flags); 8964} 8965 8966 8967void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8968{ 8969 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8970 8971 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8972 (mClient == 0) ? getpid_cached : mClient->pid(), 8973 mPriority, 8974 mHasControl, 8975 !locked, 8976 mCblk ? mCblk->clientIndex : 0, 8977 mCblk ? mCblk->serverIndex : 0 8978 ); 8979 8980 if (locked) { 8981 mCblk->lock.unlock(); 8982 } 8983} 8984 8985#undef LOG_TAG 8986#define LOG_TAG "AudioFlinger::EffectChain" 8987 8988AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8989 int sessionId) 8990 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8991 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8992 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8993{ 8994 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8995 if (thread == NULL) { 8996 return; 8997 } 8998 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8999 thread->frameCount(); 9000} 9001 9002AudioFlinger::EffectChain::~EffectChain() 9003{ 9004 if (mOwnInBuffer) { 9005 delete mInBuffer; 9006 } 9007 9008} 9009 9010// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9011sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9012{ 9013 size_t size = mEffects.size(); 9014 9015 for (size_t i = 0; i < size; i++) { 9016 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9017 return mEffects[i]; 9018 } 9019 } 9020 return 0; 9021} 9022 9023// getEffectFromId_l() must be called with ThreadBase::mLock held 9024sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9025{ 9026 size_t size = mEffects.size(); 9027 9028 for (size_t i = 0; i < size; i++) { 9029 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9030 if (id == 0 || mEffects[i]->id() == id) { 9031 return mEffects[i]; 9032 } 9033 } 9034 return 0; 9035} 9036 9037// getEffectFromType_l() must be called with ThreadBase::mLock held 9038sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9039 const effect_uuid_t *type) 9040{ 9041 size_t size = mEffects.size(); 9042 9043 for (size_t i = 0; i < size; i++) { 9044 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9045 return mEffects[i]; 9046 } 9047 } 9048 return 0; 9049} 9050 9051void AudioFlinger::EffectChain::clearInputBuffer() 9052{ 9053 Mutex::Autolock _l(mLock); 9054 sp<ThreadBase> thread = mThread.promote(); 9055 if (thread == 0) { 9056 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9057 return; 9058 } 9059 clearInputBuffer_l(thread); 9060} 9061 9062// Must be called with EffectChain::mLock locked 9063void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9064{ 9065 size_t numSamples = thread->frameCount() * thread->channelCount(); 9066 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9067 9068} 9069 9070// Must be called with EffectChain::mLock locked 9071void AudioFlinger::EffectChain::process_l() 9072{ 9073 sp<ThreadBase> thread = mThread.promote(); 9074 if (thread == 0) { 9075 ALOGW("process_l(): cannot promote mixer thread"); 9076 return; 9077 } 9078 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9079 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9080 // always process effects unless no more tracks are on the session and the effect tail 9081 // has been rendered 9082 bool doProcess = true; 9083 if (!isGlobalSession) { 9084 bool tracksOnSession = (trackCnt() != 0); 9085 9086 if (!tracksOnSession && mTailBufferCount == 0) { 9087 doProcess = false; 9088 } 9089 9090 if (activeTrackCnt() == 0) { 9091 // if no track is active and the effect tail has not been rendered, 9092 // the input buffer must be cleared here as the mixer process will not do it 9093 if (tracksOnSession || mTailBufferCount > 0) { 9094 clearInputBuffer_l(thread); 9095 if (mTailBufferCount > 0) { 9096 mTailBufferCount--; 9097 } 9098 } 9099 } 9100 } 9101 9102 size_t size = mEffects.size(); 9103 if (doProcess) { 9104 for (size_t i = 0; i < size; i++) { 9105 mEffects[i]->process(); 9106 } 9107 } 9108 for (size_t i = 0; i < size; i++) { 9109 mEffects[i]->updateState(); 9110 } 9111} 9112 9113// addEffect_l() must be called with PlaybackThread::mLock held 9114status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9115{ 9116 effect_descriptor_t desc = effect->desc(); 9117 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9118 9119 Mutex::Autolock _l(mLock); 9120 effect->setChain(this); 9121 sp<ThreadBase> thread = mThread.promote(); 9122 if (thread == 0) { 9123 return NO_INIT; 9124 } 9125 effect->setThread(thread); 9126 9127 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9128 // Auxiliary effects are inserted at the beginning of mEffects vector as 9129 // they are processed first and accumulated in chain input buffer 9130 mEffects.insertAt(effect, 0); 9131 9132 // the input buffer for auxiliary effect contains mono samples in 9133 // 32 bit format. This is to avoid saturation in AudoMixer 9134 // accumulation stage. Saturation is done in EffectModule::process() before 9135 // calling the process in effect engine 9136 size_t numSamples = thread->frameCount(); 9137 int32_t *buffer = new int32_t[numSamples]; 9138 memset(buffer, 0, numSamples * sizeof(int32_t)); 9139 effect->setInBuffer((int16_t *)buffer); 9140 // auxiliary effects output samples to chain input buffer for further processing 9141 // by insert effects 9142 effect->setOutBuffer(mInBuffer); 9143 } else { 9144 // Insert effects are inserted at the end of mEffects vector as they are processed 9145 // after track and auxiliary effects. 9146 // Insert effect order as a function of indicated preference: 9147 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9148 // another effect is present 9149 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9150 // last effect claiming first position 9151 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9152 // first effect claiming last position 9153 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9154 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9155 // already present 9156 9157 size_t size = mEffects.size(); 9158 size_t idx_insert = size; 9159 ssize_t idx_insert_first = -1; 9160 ssize_t idx_insert_last = -1; 9161 9162 for (size_t i = 0; i < size; i++) { 9163 effect_descriptor_t d = mEffects[i]->desc(); 9164 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9165 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9166 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9167 // check invalid effect chaining combinations 9168 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9169 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9170 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9171 return INVALID_OPERATION; 9172 } 9173 // remember position of first insert effect and by default 9174 // select this as insert position for new effect 9175 if (idx_insert == size) { 9176 idx_insert = i; 9177 } 9178 // remember position of last insert effect claiming 9179 // first position 9180 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9181 idx_insert_first = i; 9182 } 9183 // remember position of first insert effect claiming 9184 // last position 9185 if (iPref == EFFECT_FLAG_INSERT_LAST && 9186 idx_insert_last == -1) { 9187 idx_insert_last = i; 9188 } 9189 } 9190 } 9191 9192 // modify idx_insert from first position if needed 9193 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9194 if (idx_insert_last != -1) { 9195 idx_insert = idx_insert_last; 9196 } else { 9197 idx_insert = size; 9198 } 9199 } else { 9200 if (idx_insert_first != -1) { 9201 idx_insert = idx_insert_first + 1; 9202 } 9203 } 9204 9205 // always read samples from chain input buffer 9206 effect->setInBuffer(mInBuffer); 9207 9208 // if last effect in the chain, output samples to chain 9209 // output buffer, otherwise to chain input buffer 9210 if (idx_insert == size) { 9211 if (idx_insert != 0) { 9212 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9213 mEffects[idx_insert-1]->configure(); 9214 } 9215 effect->setOutBuffer(mOutBuffer); 9216 } else { 9217 effect->setOutBuffer(mInBuffer); 9218 } 9219 mEffects.insertAt(effect, idx_insert); 9220 9221 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9222 } 9223 effect->configure(); 9224 return NO_ERROR; 9225} 9226 9227// removeEffect_l() must be called with PlaybackThread::mLock held 9228size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9229{ 9230 Mutex::Autolock _l(mLock); 9231 size_t size = mEffects.size(); 9232 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9233 9234 for (size_t i = 0; i < size; i++) { 9235 if (effect == mEffects[i]) { 9236 // calling stop here will remove pre-processing effect from the audio HAL. 9237 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9238 // the middle of a read from audio HAL 9239 if (mEffects[i]->state() == EffectModule::ACTIVE || 9240 mEffects[i]->state() == EffectModule::STOPPING) { 9241 mEffects[i]->stop(); 9242 } 9243 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9244 delete[] effect->inBuffer(); 9245 } else { 9246 if (i == size - 1 && i != 0) { 9247 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9248 mEffects[i - 1]->configure(); 9249 } 9250 } 9251 mEffects.removeAt(i); 9252 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9253 break; 9254 } 9255 } 9256 9257 return mEffects.size(); 9258} 9259 9260// setDevice_l() must be called with PlaybackThread::mLock held 9261void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9262{ 9263 size_t size = mEffects.size(); 9264 for (size_t i = 0; i < size; i++) { 9265 mEffects[i]->setDevice(device); 9266 } 9267} 9268 9269// setMode_l() must be called with PlaybackThread::mLock held 9270void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9271{ 9272 size_t size = mEffects.size(); 9273 for (size_t i = 0; i < size; i++) { 9274 mEffects[i]->setMode(mode); 9275 } 9276} 9277 9278// setVolume_l() must be called with PlaybackThread::mLock held 9279bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9280{ 9281 uint32_t newLeft = *left; 9282 uint32_t newRight = *right; 9283 bool hasControl = false; 9284 int ctrlIdx = -1; 9285 size_t size = mEffects.size(); 9286 9287 // first update volume controller 9288 for (size_t i = size; i > 0; i--) { 9289 if (mEffects[i - 1]->isProcessEnabled() && 9290 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9291 ctrlIdx = i - 1; 9292 hasControl = true; 9293 break; 9294 } 9295 } 9296 9297 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9298 if (hasControl) { 9299 *left = mNewLeftVolume; 9300 *right = mNewRightVolume; 9301 } 9302 return hasControl; 9303 } 9304 9305 mVolumeCtrlIdx = ctrlIdx; 9306 mLeftVolume = newLeft; 9307 mRightVolume = newRight; 9308 9309 // second get volume update from volume controller 9310 if (ctrlIdx >= 0) { 9311 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9312 mNewLeftVolume = newLeft; 9313 mNewRightVolume = newRight; 9314 } 9315 // then indicate volume to all other effects in chain. 9316 // Pass altered volume to effects before volume controller 9317 // and requested volume to effects after controller 9318 uint32_t lVol = newLeft; 9319 uint32_t rVol = newRight; 9320 9321 for (size_t i = 0; i < size; i++) { 9322 if ((int)i == ctrlIdx) continue; 9323 // this also works for ctrlIdx == -1 when there is no volume controller 9324 if ((int)i > ctrlIdx) { 9325 lVol = *left; 9326 rVol = *right; 9327 } 9328 mEffects[i]->setVolume(&lVol, &rVol, false); 9329 } 9330 *left = newLeft; 9331 *right = newRight; 9332 9333 return hasControl; 9334} 9335 9336status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9337{ 9338 const size_t SIZE = 256; 9339 char buffer[SIZE]; 9340 String8 result; 9341 9342 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9343 result.append(buffer); 9344 9345 bool locked = tryLock(mLock); 9346 // failed to lock - AudioFlinger is probably deadlocked 9347 if (!locked) { 9348 result.append("\tCould not lock mutex:\n"); 9349 } 9350 9351 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9352 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9353 mEffects.size(), 9354 (uint32_t)mInBuffer, 9355 (uint32_t)mOutBuffer, 9356 mActiveTrackCnt); 9357 result.append(buffer); 9358 write(fd, result.string(), result.size()); 9359 9360 for (size_t i = 0; i < mEffects.size(); ++i) { 9361 sp<EffectModule> effect = mEffects[i]; 9362 if (effect != 0) { 9363 effect->dump(fd, args); 9364 } 9365 } 9366 9367 if (locked) { 9368 mLock.unlock(); 9369 } 9370 9371 return NO_ERROR; 9372} 9373 9374// must be called with ThreadBase::mLock held 9375void AudioFlinger::EffectChain::setEffectSuspended_l( 9376 const effect_uuid_t *type, bool suspend) 9377{ 9378 sp<SuspendedEffectDesc> desc; 9379 // use effect type UUID timelow as key as there is no real risk of identical 9380 // timeLow fields among effect type UUIDs. 9381 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9382 if (suspend) { 9383 if (index >= 0) { 9384 desc = mSuspendedEffects.valueAt(index); 9385 } else { 9386 desc = new SuspendedEffectDesc(); 9387 desc->mType = *type; 9388 mSuspendedEffects.add(type->timeLow, desc); 9389 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9390 } 9391 if (desc->mRefCount++ == 0) { 9392 sp<EffectModule> effect = getEffectIfEnabled(type); 9393 if (effect != 0) { 9394 desc->mEffect = effect; 9395 effect->setSuspended(true); 9396 effect->setEnabled(false); 9397 } 9398 } 9399 } else { 9400 if (index < 0) { 9401 return; 9402 } 9403 desc = mSuspendedEffects.valueAt(index); 9404 if (desc->mRefCount <= 0) { 9405 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9406 desc->mRefCount = 1; 9407 } 9408 if (--desc->mRefCount == 0) { 9409 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9410 if (desc->mEffect != 0) { 9411 sp<EffectModule> effect = desc->mEffect.promote(); 9412 if (effect != 0) { 9413 effect->setSuspended(false); 9414 effect->lock(); 9415 EffectHandle *handle = effect->controlHandle_l(); 9416 if (handle != NULL && !handle->destroyed_l()) { 9417 effect->setEnabled_l(handle->enabled()); 9418 } 9419 effect->unlock(); 9420 } 9421 desc->mEffect.clear(); 9422 } 9423 mSuspendedEffects.removeItemsAt(index); 9424 } 9425 } 9426} 9427 9428// must be called with ThreadBase::mLock held 9429void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9430{ 9431 sp<SuspendedEffectDesc> desc; 9432 9433 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9434 if (suspend) { 9435 if (index >= 0) { 9436 desc = mSuspendedEffects.valueAt(index); 9437 } else { 9438 desc = new SuspendedEffectDesc(); 9439 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9440 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9441 } 9442 if (desc->mRefCount++ == 0) { 9443 Vector< sp<EffectModule> > effects; 9444 getSuspendEligibleEffects(effects); 9445 for (size_t i = 0; i < effects.size(); i++) { 9446 setEffectSuspended_l(&effects[i]->desc().type, true); 9447 } 9448 } 9449 } else { 9450 if (index < 0) { 9451 return; 9452 } 9453 desc = mSuspendedEffects.valueAt(index); 9454 if (desc->mRefCount <= 0) { 9455 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9456 desc->mRefCount = 1; 9457 } 9458 if (--desc->mRefCount == 0) { 9459 Vector<const effect_uuid_t *> types; 9460 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9461 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9462 continue; 9463 } 9464 types.add(&mSuspendedEffects.valueAt(i)->mType); 9465 } 9466 for (size_t i = 0; i < types.size(); i++) { 9467 setEffectSuspended_l(types[i], false); 9468 } 9469 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9470 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9471 } 9472 } 9473} 9474 9475 9476// The volume effect is used for automated tests only 9477#ifndef OPENSL_ES_H_ 9478static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9479 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9480const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9481#endif //OPENSL_ES_H_ 9482 9483bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9484{ 9485 // auxiliary effects and visualizer are never suspended on output mix 9486 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9487 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9488 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9489 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9490 return false; 9491 } 9492 return true; 9493} 9494 9495void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9496{ 9497 effects.clear(); 9498 for (size_t i = 0; i < mEffects.size(); i++) { 9499 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9500 effects.add(mEffects[i]); 9501 } 9502 } 9503} 9504 9505sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9506 const effect_uuid_t *type) 9507{ 9508 sp<EffectModule> effect = getEffectFromType_l(type); 9509 return effect != 0 && effect->isEnabled() ? effect : 0; 9510} 9511 9512void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9513 bool enabled) 9514{ 9515 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9516 if (enabled) { 9517 if (index < 0) { 9518 // if the effect is not suspend check if all effects are suspended 9519 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9520 if (index < 0) { 9521 return; 9522 } 9523 if (!isEffectEligibleForSuspend(effect->desc())) { 9524 return; 9525 } 9526 setEffectSuspended_l(&effect->desc().type, enabled); 9527 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9528 if (index < 0) { 9529 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9530 return; 9531 } 9532 } 9533 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9534 effect->desc().type.timeLow); 9535 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9536 // if effect is requested to suspended but was not yet enabled, supend it now. 9537 if (desc->mEffect == 0) { 9538 desc->mEffect = effect; 9539 effect->setEnabled(false); 9540 effect->setSuspended(true); 9541 } 9542 } else { 9543 if (index < 0) { 9544 return; 9545 } 9546 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9547 effect->desc().type.timeLow); 9548 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9549 desc->mEffect.clear(); 9550 effect->setSuspended(false); 9551 } 9552} 9553 9554#undef LOG_TAG 9555#define LOG_TAG "AudioFlinger" 9556 9557// ---------------------------------------------------------------------------- 9558 9559status_t AudioFlinger::onTransact( 9560 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9561{ 9562 return BnAudioFlinger::onTransact(code, data, reply, flags); 9563} 9564 9565}; // namespace android 9566