AudioFlinger.cpp revision 44cda3a4e7ca3de0db9cb49145def3803b03ebb4
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <utils/Trace.h> 31#include <binder/Parcel.h> 32#include <binder/IPCThreadState.h> 33#include <utils/String16.h> 34#include <utils/threads.h> 35#include <utils/Atomic.h> 36 37#include <cutils/bitops.h> 38#include <cutils/properties.h> 39#include <cutils/compiler.h> 40 41#undef ADD_BATTERY_DATA 42 43#ifdef ADD_BATTERY_DATA 44#include <media/IMediaPlayerService.h> 45#include <media/IMediaDeathNotifier.h> 46#endif 47 48#include <private/media/AudioTrackShared.h> 49#include <private/media/AudioEffectShared.h> 50 51#include <system/audio.h> 52#include <hardware/audio.h> 53 54#include "AudioMixer.h" 55#include "AudioFlinger.h" 56#include "ServiceUtilities.h" 57 58#include <media/EffectsFactoryApi.h> 59#include <audio_effects/effect_visualizer.h> 60#include <audio_effects/effect_ns.h> 61#include <audio_effects/effect_aec.h> 62 63#include <audio_utils/primitives.h> 64 65#include <powermanager/PowerManager.h> 66 67// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 68#ifdef DEBUG_CPU_USAGE 69#include <cpustats/CentralTendencyStatistics.h> 70#include <cpustats/ThreadCpuUsage.h> 71#endif 72 73#include <common_time/cc_helper.h> 74#include <common_time/local_clock.h> 75 76#include "FastMixer.h" 77 78// NBAIO implementations 79#include "AudioStreamOutSink.h" 80#include "MonoPipe.h" 81#include "MonoPipeReader.h" 82#include "Pipe.h" 83#include "PipeReader.h" 84#include "SourceAudioBufferProvider.h" 85 86#include "SchedulingPolicyService.h" 87 88// ---------------------------------------------------------------------------- 89 90// Note: the following macro is used for extremely verbose logging message. In 91// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 92// 0; but one side effect of this is to turn all LOGV's as well. Some messages 93// are so verbose that we want to suppress them even when we have ALOG_ASSERT 94// turned on. Do not uncomment the #def below unless you really know what you 95// are doing and want to see all of the extremely verbose messages. 96//#define VERY_VERY_VERBOSE_LOGGING 97#ifdef VERY_VERY_VERBOSE_LOGGING 98#define ALOGVV ALOGV 99#else 100#define ALOGVV(a...) do { } while(0) 101#endif 102 103namespace android { 104 105static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 106static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 107 108static const float MAX_GAIN = 4096.0f; 109static const uint32_t MAX_GAIN_INT = 0x1000; 110 111// retry counts for buffer fill timeout 112// 50 * ~20msecs = 1 second 113static const int8_t kMaxTrackRetries = 50; 114static const int8_t kMaxTrackStartupRetries = 50; 115// allow less retry attempts on direct output thread. 116// direct outputs can be a scarce resource in audio hardware and should 117// be released as quickly as possible. 118static const int8_t kMaxTrackRetriesDirect = 2; 119 120static const int kDumpLockRetries = 50; 121static const int kDumpLockSleepUs = 20000; 122 123// don't warn about blocked writes or record buffer overflows more often than this 124static const nsecs_t kWarningThrottleNs = seconds(5); 125 126// RecordThread loop sleep time upon application overrun or audio HAL read error 127static const int kRecordThreadSleepUs = 5000; 128 129// maximum time to wait for setParameters to complete 130static const nsecs_t kSetParametersTimeoutNs = seconds(2); 131 132// minimum sleep time for the mixer thread loop when tracks are active but in underrun 133static const uint32_t kMinThreadSleepTimeUs = 5000; 134// maximum divider applied to the active sleep time in the mixer thread loop 135static const uint32_t kMaxThreadSleepTimeShift = 2; 136 137// minimum normal mix buffer size, expressed in milliseconds rather than frames 138static const uint32_t kMinNormalMixBufferSizeMs = 20; 139// maximum normal mix buffer size 140static const uint32_t kMaxNormalMixBufferSizeMs = 24; 141 142nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 143 144// Whether to use fast mixer 145static const enum { 146 FastMixer_Never, // never initialize or use: for debugging only 147 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 148 // normal mixer multiplier is 1 149 FastMixer_Static, // initialize if needed, then use all the time if initialized, 150 // multiplier is calculated based on min & max normal mixer buffer size 151 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 152 // multiplier is calculated based on min & max normal mixer buffer size 153 // FIXME for FastMixer_Dynamic: 154 // Supporting this option will require fixing HALs that can't handle large writes. 155 // For example, one HAL implementation returns an error from a large write, 156 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 157 // We could either fix the HAL implementations, or provide a wrapper that breaks 158 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 159} kUseFastMixer = FastMixer_Static; 160 161static uint32_t gScreenState; // incremented by 2 when screen state changes, bit 0 == 1 means "off" 162 // AudioFlinger::setParameters() updates, other threads read w/o lock 163 164// Priorities for requestPriority 165static const int kPriorityAudioApp = 2; 166static const int kPriorityFastMixer = 3; 167 168// ---------------------------------------------------------------------------- 169 170#ifdef ADD_BATTERY_DATA 171// To collect the amplifier usage 172static void addBatteryData(uint32_t params) { 173 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 174 if (service == NULL) { 175 // it already logged 176 return; 177 } 178 179 service->addBatteryData(params); 180} 181#endif 182 183static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 184{ 185 const hw_module_t *mod; 186 int rc; 187 188 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 189 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 190 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 191 if (rc) { 192 goto out; 193 } 194 rc = audio_hw_device_open(mod, dev); 195 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 196 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 197 if (rc) { 198 goto out; 199 } 200 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 201 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 202 rc = BAD_VALUE; 203 goto out; 204 } 205 return 0; 206 207out: 208 *dev = NULL; 209 return rc; 210} 211 212// ---------------------------------------------------------------------------- 213 214AudioFlinger::AudioFlinger() 215 : BnAudioFlinger(), 216 mPrimaryHardwareDev(NULL), 217 mHardwareStatus(AUDIO_HW_IDLE), 218 mMasterVolume(1.0f), 219 mMasterVolumeSW(1.0f), 220 mMasterVolumeSupportLvl(MVS_NONE), 221 mMasterMute(false), 222 mNextUniqueId(1), 223 mMode(AUDIO_MODE_INVALID), 224 mBtNrecIsOff(false) 225{ 226} 227 228void AudioFlinger::onFirstRef() 229{ 230 int rc = 0; 231 232 Mutex::Autolock _l(mLock); 233 234 /* TODO: move all this work into an Init() function */ 235 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 236 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 237 uint32_t int_val; 238 if (1 == sscanf(val_str, "%u", &int_val)) { 239 mStandbyTimeInNsecs = milliseconds(int_val); 240 ALOGI("Using %u mSec as standby time.", int_val); 241 } else { 242 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 243 ALOGI("Using default %u mSec as standby time.", 244 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 245 } 246 } 247 248 mMode = AUDIO_MODE_NORMAL; 249} 250 251AudioFlinger::~AudioFlinger() 252{ 253 while (!mRecordThreads.isEmpty()) { 254 // closeInput() will remove first entry from mRecordThreads 255 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 256 } 257 while (!mPlaybackThreads.isEmpty()) { 258 // closeOutput() will remove first entry from mPlaybackThreads 259 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 260 } 261 262 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 263 // no mHardwareLock needed, as there are no other references to this 264 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 265 delete mAudioHwDevs.valueAt(i); 266 } 267} 268 269static const char * const audio_interfaces[] = { 270 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 271 AUDIO_HARDWARE_MODULE_ID_A2DP, 272 AUDIO_HARDWARE_MODULE_ID_USB, 273}; 274#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 275 276audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, audio_devices_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 } else { 286 // check a match for the requested module handle 287 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 288 if (audioHwdevice != NULL) { 289 return audioHwdevice->hwDevice(); 290 } 291 } 292 // then try to find a module supporting the requested device. 293 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 294 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 295 if ((dev->get_supported_devices(dev) & devices) == devices) 296 return dev; 297 } 298 299 return NULL; 300} 301 302void AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 303{ 304 const size_t SIZE = 256; 305 char buffer[SIZE]; 306 String8 result; 307 308 result.append("Clients:\n"); 309 for (size_t i = 0; i < mClients.size(); ++i) { 310 sp<Client> client = mClients.valueAt(i).promote(); 311 if (client != 0) { 312 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 313 result.append(buffer); 314 } 315 } 316 317 result.append("Global session refs:\n"); 318 result.append(" session pid count\n"); 319 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 320 AudioSessionRef *r = mAudioSessionRefs[i]; 321 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 322 result.append(buffer); 323 } 324 write(fd, result.string(), result.size()); 325} 326 327 328void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 329{ 330 const size_t SIZE = 256; 331 char buffer[SIZE]; 332 String8 result; 333 hardware_call_state hardwareStatus = mHardwareStatus; 334 335 snprintf(buffer, SIZE, "Hardware status: %d\n" 336 "Standby Time mSec: %u\n", 337 hardwareStatus, 338 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 339 result.append(buffer); 340 write(fd, result.string(), result.size()); 341} 342 343void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 344{ 345 const size_t SIZE = 256; 346 char buffer[SIZE]; 347 String8 result; 348 snprintf(buffer, SIZE, "Permission Denial: " 349 "can't dump AudioFlinger from pid=%d, uid=%d\n", 350 IPCThreadState::self()->getCallingPid(), 351 IPCThreadState::self()->getCallingUid()); 352 result.append(buffer); 353 write(fd, result.string(), result.size()); 354} 355 356static bool tryLock(Mutex& mutex) 357{ 358 bool locked = false; 359 for (int i = 0; i < kDumpLockRetries; ++i) { 360 if (mutex.tryLock() == NO_ERROR) { 361 locked = true; 362 break; 363 } 364 usleep(kDumpLockSleepUs); 365 } 366 return locked; 367} 368 369status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 370{ 371 if (!dumpAllowed()) { 372 dumpPermissionDenial(fd, args); 373 } else { 374 // get state of hardware lock 375 bool hardwareLocked = tryLock(mHardwareLock); 376 if (!hardwareLocked) { 377 String8 result(kHardwareLockedString); 378 write(fd, result.string(), result.size()); 379 } else { 380 mHardwareLock.unlock(); 381 } 382 383 bool locked = tryLock(mLock); 384 385 // failed to lock - AudioFlinger is probably deadlocked 386 if (!locked) { 387 String8 result(kDeadlockedString); 388 write(fd, result.string(), result.size()); 389 } 390 391 dumpClients(fd, args); 392 dumpInternals(fd, args); 393 394 // dump playback threads 395 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 396 mPlaybackThreads.valueAt(i)->dump(fd, args); 397 } 398 399 // dump record threads 400 for (size_t i = 0; i < mRecordThreads.size(); i++) { 401 mRecordThreads.valueAt(i)->dump(fd, args); 402 } 403 404 // dump all hardware devs 405 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 406 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 407 dev->dump(dev, fd); 408 } 409 if (locked) mLock.unlock(); 410 } 411 return NO_ERROR; 412} 413 414sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 415{ 416 // If pid is already in the mClients wp<> map, then use that entry 417 // (for which promote() is always != 0), otherwise create a new entry and Client. 418 sp<Client> client = mClients.valueFor(pid).promote(); 419 if (client == 0) { 420 client = new Client(this, pid); 421 mClients.add(pid, client); 422 } 423 424 return client; 425} 426 427// IAudioFlinger interface 428 429 430sp<IAudioTrack> AudioFlinger::createTrack( 431 pid_t pid, 432 audio_stream_type_t streamType, 433 uint32_t sampleRate, 434 audio_format_t format, 435 audio_channel_mask_t channelMask, 436 int frameCount, 437 IAudioFlinger::track_flags_t flags, 438 const sp<IMemory>& sharedBuffer, 439 audio_io_handle_t output, 440 pid_t tid, 441 int *sessionId, 442 status_t *status) 443{ 444 sp<PlaybackThread::Track> track; 445 sp<TrackHandle> trackHandle; 446 sp<Client> client; 447 status_t lStatus; 448 int lSessionId; 449 450 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 451 // but if someone uses binder directly they could bypass that and cause us to crash 452 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 453 ALOGE("createTrack() invalid stream type %d", streamType); 454 lStatus = BAD_VALUE; 455 goto Exit; 456 } 457 458 { 459 Mutex::Autolock _l(mLock); 460 PlaybackThread *thread = checkPlaybackThread_l(output); 461 PlaybackThread *effectThread = NULL; 462 if (thread == NULL) { 463 ALOGE("unknown output thread"); 464 lStatus = BAD_VALUE; 465 goto Exit; 466 } 467 468 client = registerPid_l(pid); 469 470 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 471 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 472 // check if an effect chain with the same session ID is present on another 473 // output thread and move it here. 474 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 475 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 476 if (mPlaybackThreads.keyAt(i) != output) { 477 uint32_t sessions = t->hasAudioSession(*sessionId); 478 if (sessions & PlaybackThread::EFFECT_SESSION) { 479 effectThread = t.get(); 480 break; 481 } 482 } 483 } 484 lSessionId = *sessionId; 485 } else { 486 // if no audio session id is provided, create one here 487 lSessionId = nextUniqueId(); 488 if (sessionId != NULL) { 489 *sessionId = lSessionId; 490 } 491 } 492 ALOGV("createTrack() lSessionId: %d", lSessionId); 493 494 track = thread->createTrack_l(client, streamType, sampleRate, format, 495 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 496 497 // move effect chain to this output thread if an effect on same session was waiting 498 // for a track to be created 499 if (lStatus == NO_ERROR && effectThread != NULL) { 500 Mutex::Autolock _dl(thread->mLock); 501 Mutex::Autolock _sl(effectThread->mLock); 502 moveEffectChain_l(lSessionId, effectThread, thread, true); 503 } 504 505 // Look for sync events awaiting for a session to be used. 506 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 507 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 508 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 509 if (lStatus == NO_ERROR) { 510 track->setSyncEvent(mPendingSyncEvents[i]); 511 } else { 512 mPendingSyncEvents[i]->cancel(); 513 } 514 mPendingSyncEvents.removeAt(i); 515 i--; 516 } 517 } 518 } 519 } 520 if (lStatus == NO_ERROR) { 521 trackHandle = new TrackHandle(track); 522 } else { 523 // remove local strong reference to Client before deleting the Track so that the Client 524 // destructor is called by the TrackBase destructor with mLock held 525 client.clear(); 526 track.clear(); 527 } 528 529Exit: 530 if (status != NULL) { 531 *status = lStatus; 532 } 533 return trackHandle; 534} 535 536uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 537{ 538 Mutex::Autolock _l(mLock); 539 PlaybackThread *thread = checkPlaybackThread_l(output); 540 if (thread == NULL) { 541 ALOGW("sampleRate() unknown thread %d", output); 542 return 0; 543 } 544 return thread->sampleRate(); 545} 546 547int AudioFlinger::channelCount(audio_io_handle_t output) const 548{ 549 Mutex::Autolock _l(mLock); 550 PlaybackThread *thread = checkPlaybackThread_l(output); 551 if (thread == NULL) { 552 ALOGW("channelCount() unknown thread %d", output); 553 return 0; 554 } 555 return thread->channelCount(); 556} 557 558audio_format_t AudioFlinger::format(audio_io_handle_t output) const 559{ 560 Mutex::Autolock _l(mLock); 561 PlaybackThread *thread = checkPlaybackThread_l(output); 562 if (thread == NULL) { 563 ALOGW("format() unknown thread %d", output); 564 return AUDIO_FORMAT_INVALID; 565 } 566 return thread->format(); 567} 568 569size_t AudioFlinger::frameCount(audio_io_handle_t output) const 570{ 571 Mutex::Autolock _l(mLock); 572 PlaybackThread *thread = checkPlaybackThread_l(output); 573 if (thread == NULL) { 574 ALOGW("frameCount() unknown thread %d", output); 575 return 0; 576 } 577 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 578 // should examine all callers and fix them to handle smaller counts 579 return thread->frameCount(); 580} 581 582uint32_t AudioFlinger::latency(audio_io_handle_t output) const 583{ 584 Mutex::Autolock _l(mLock); 585 PlaybackThread *thread = checkPlaybackThread_l(output); 586 if (thread == NULL) { 587 ALOGW("latency() unknown thread %d", output); 588 return 0; 589 } 590 return thread->latency(); 591} 592 593status_t AudioFlinger::setMasterVolume(float value) 594{ 595 status_t ret = initCheck(); 596 if (ret != NO_ERROR) { 597 return ret; 598 } 599 600 // check calling permissions 601 if (!settingsAllowed()) { 602 return PERMISSION_DENIED; 603 } 604 605 float swmv = value; 606 607 Mutex::Autolock _l(mLock); 608 609 // when hw supports master volume, don't scale in sw mixer 610 if (MVS_NONE != mMasterVolumeSupportLvl) { 611 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 612 AutoMutex lock(mHardwareLock); 613 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 614 615 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 616 if (NULL != dev->set_master_volume) { 617 dev->set_master_volume(dev, value); 618 } 619 mHardwareStatus = AUDIO_HW_IDLE; 620 } 621 622 swmv = 1.0; 623 } 624 625 mMasterVolume = value; 626 mMasterVolumeSW = swmv; 627 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 628 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 629 630 return NO_ERROR; 631} 632 633status_t AudioFlinger::setMode(audio_mode_t mode) 634{ 635 status_t ret = initCheck(); 636 if (ret != NO_ERROR) { 637 return ret; 638 } 639 640 // check calling permissions 641 if (!settingsAllowed()) { 642 return PERMISSION_DENIED; 643 } 644 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 645 ALOGW("Illegal value: setMode(%d)", mode); 646 return BAD_VALUE; 647 } 648 649 { // scope for the lock 650 AutoMutex lock(mHardwareLock); 651 mHardwareStatus = AUDIO_HW_SET_MODE; 652 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 653 mHardwareStatus = AUDIO_HW_IDLE; 654 } 655 656 if (NO_ERROR == ret) { 657 Mutex::Autolock _l(mLock); 658 mMode = mode; 659 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 660 mPlaybackThreads.valueAt(i)->setMode(mode); 661 } 662 663 return ret; 664} 665 666status_t AudioFlinger::setMicMute(bool state) 667{ 668 status_t ret = initCheck(); 669 if (ret != NO_ERROR) { 670 return ret; 671 } 672 673 // check calling permissions 674 if (!settingsAllowed()) { 675 return PERMISSION_DENIED; 676 } 677 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 680 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return ret; 683} 684 685bool AudioFlinger::getMicMute() const 686{ 687 status_t ret = initCheck(); 688 if (ret != NO_ERROR) { 689 return false; 690 } 691 692 bool state = AUDIO_MODE_INVALID; 693 AutoMutex lock(mHardwareLock); 694 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 695 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 696 mHardwareStatus = AUDIO_HW_IDLE; 697 return state; 698} 699 700status_t AudioFlinger::setMasterMute(bool muted) 701{ 702 // check calling permissions 703 if (!settingsAllowed()) { 704 return PERMISSION_DENIED; 705 } 706 707 Mutex::Autolock _l(mLock); 708 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 709 mMasterMute = muted; 710 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 711 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 712 713 return NO_ERROR; 714} 715 716float AudioFlinger::masterVolume() const 717{ 718 Mutex::Autolock _l(mLock); 719 return masterVolume_l(); 720} 721 722float AudioFlinger::masterVolumeSW() const 723{ 724 Mutex::Autolock _l(mLock); 725 return masterVolumeSW_l(); 726} 727 728bool AudioFlinger::masterMute() const 729{ 730 Mutex::Autolock _l(mLock); 731 return masterMute_l(); 732} 733 734float AudioFlinger::masterVolume_l() const 735{ 736 if (MVS_FULL == mMasterVolumeSupportLvl) { 737 float ret_val; 738 AutoMutex lock(mHardwareLock); 739 740 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 741 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 742 (NULL != mPrimaryHardwareDev->get_master_volume), 743 "can't get master volume"); 744 745 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 746 mHardwareStatus = AUDIO_HW_IDLE; 747 return ret_val; 748 } 749 750 return mMasterVolume; 751} 752 753status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 754 audio_io_handle_t output) 755{ 756 // check calling permissions 757 if (!settingsAllowed()) { 758 return PERMISSION_DENIED; 759 } 760 761 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 762 ALOGE("setStreamVolume() invalid stream %d", stream); 763 return BAD_VALUE; 764 } 765 766 AutoMutex lock(mLock); 767 PlaybackThread *thread = NULL; 768 if (output) { 769 thread = checkPlaybackThread_l(output); 770 if (thread == NULL) { 771 return BAD_VALUE; 772 } 773 } 774 775 mStreamTypes[stream].volume = value; 776 777 if (thread == NULL) { 778 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 779 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 780 } 781 } else { 782 thread->setStreamVolume(stream, value); 783 } 784 785 return NO_ERROR; 786} 787 788status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 789{ 790 // check calling permissions 791 if (!settingsAllowed()) { 792 return PERMISSION_DENIED; 793 } 794 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 796 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 797 ALOGE("setStreamMute() invalid stream %d", stream); 798 return BAD_VALUE; 799 } 800 801 AutoMutex lock(mLock); 802 mStreamTypes[stream].mute = muted; 803 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 804 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 805 806 return NO_ERROR; 807} 808 809float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 810{ 811 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 812 return 0.0f; 813 } 814 815 AutoMutex lock(mLock); 816 float volume; 817 if (output) { 818 PlaybackThread *thread = checkPlaybackThread_l(output); 819 if (thread == NULL) { 820 return 0.0f; 821 } 822 volume = thread->streamVolume(stream); 823 } else { 824 volume = streamVolume_l(stream); 825 } 826 827 return volume; 828} 829 830bool AudioFlinger::streamMute(audio_stream_type_t stream) const 831{ 832 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 833 return true; 834 } 835 836 AutoMutex lock(mLock); 837 return streamMute_l(stream); 838} 839 840status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 841{ 842 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 843 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 844 // check calling permissions 845 if (!settingsAllowed()) { 846 return PERMISSION_DENIED; 847 } 848 849 // ioHandle == 0 means the parameters are global to the audio hardware interface 850 if (ioHandle == 0) { 851 Mutex::Autolock _l(mLock); 852 status_t final_result = NO_ERROR; 853 { 854 AutoMutex lock(mHardwareLock); 855 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 856 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 857 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 858 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 859 final_result = result ?: final_result; 860 } 861 mHardwareStatus = AUDIO_HW_IDLE; 862 } 863 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 864 AudioParameter param = AudioParameter(keyValuePairs); 865 String8 value; 866 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 867 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 868 if (mBtNrecIsOff != btNrecIsOff) { 869 for (size_t i = 0; i < mRecordThreads.size(); i++) { 870 sp<RecordThread> thread = mRecordThreads.valueAt(i); 871 audio_devices_t device = thread->device() & AUDIO_DEVICE_IN_ALL; 872 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 873 // collect all of the thread's session IDs 874 KeyedVector<int, bool> ids = thread->sessionIds(); 875 // suspend effects associated with those session IDs 876 for (size_t j = 0; j < ids.size(); ++j) { 877 int sessionId = ids.keyAt(j); 878 thread->setEffectSuspended(FX_IID_AEC, 879 suspend, 880 sessionId); 881 thread->setEffectSuspended(FX_IID_NS, 882 suspend, 883 sessionId); 884 } 885 } 886 mBtNrecIsOff = btNrecIsOff; 887 } 888 } 889 String8 screenState; 890 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 891 bool isOff = screenState == "off"; 892 if (isOff != (gScreenState & 1)) { 893 gScreenState = ((gScreenState & ~1) + 2) | isOff; 894 } 895 } 896 return final_result; 897 } 898 899 // hold a strong ref on thread in case closeOutput() or closeInput() is called 900 // and the thread is exited once the lock is released 901 sp<ThreadBase> thread; 902 { 903 Mutex::Autolock _l(mLock); 904 thread = checkPlaybackThread_l(ioHandle); 905 if (thread == 0) { 906 thread = checkRecordThread_l(ioHandle); 907 } else if (thread == primaryPlaybackThread_l()) { 908 // indicate output device change to all input threads for pre processing 909 AudioParameter param = AudioParameter(keyValuePairs); 910 int value; 911 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 912 (value != 0)) { 913 for (size_t i = 0; i < mRecordThreads.size(); i++) { 914 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 915 } 916 } 917 } 918 } 919 if (thread != 0) { 920 return thread->setParameters(keyValuePairs); 921 } 922 return BAD_VALUE; 923} 924 925String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 926{ 927// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 928// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 929 930 Mutex::Autolock _l(mLock); 931 932 if (ioHandle == 0) { 933 String8 out_s8; 934 935 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 936 char *s; 937 { 938 AutoMutex lock(mHardwareLock); 939 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 940 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 941 s = dev->get_parameters(dev, keys.string()); 942 mHardwareStatus = AUDIO_HW_IDLE; 943 } 944 out_s8 += String8(s ? s : ""); 945 free(s); 946 } 947 return out_s8; 948 } 949 950 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 951 if (playbackThread != NULL) { 952 return playbackThread->getParameters(keys); 953 } 954 RecordThread *recordThread = checkRecordThread_l(ioHandle); 955 if (recordThread != NULL) { 956 return recordThread->getParameters(keys); 957 } 958 return String8(""); 959} 960 961size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 962 audio_channel_mask_t channelMask) const 963{ 964 status_t ret = initCheck(); 965 if (ret != NO_ERROR) { 966 return 0; 967 } 968 969 AutoMutex lock(mHardwareLock); 970 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 971 struct audio_config config = { 972 sample_rate: sampleRate, 973 channel_mask: channelMask, 974 format: format, 975 }; 976 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 977 mHardwareStatus = AUDIO_HW_IDLE; 978 return size; 979} 980 981unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 982{ 983 Mutex::Autolock _l(mLock); 984 985 RecordThread *recordThread = checkRecordThread_l(ioHandle); 986 if (recordThread != NULL) { 987 return recordThread->getInputFramesLost(); 988 } 989 return 0; 990} 991 992status_t AudioFlinger::setVoiceVolume(float value) 993{ 994 status_t ret = initCheck(); 995 if (ret != NO_ERROR) { 996 return ret; 997 } 998 999 // check calling permissions 1000 if (!settingsAllowed()) { 1001 return PERMISSION_DENIED; 1002 } 1003 1004 AutoMutex lock(mHardwareLock); 1005 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1006 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1007 mHardwareStatus = AUDIO_HW_IDLE; 1008 1009 return ret; 1010} 1011 1012status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1013 audio_io_handle_t output) const 1014{ 1015 status_t status; 1016 1017 Mutex::Autolock _l(mLock); 1018 1019 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1020 if (playbackThread != NULL) { 1021 return playbackThread->getRenderPosition(halFrames, dspFrames); 1022 } 1023 1024 return BAD_VALUE; 1025} 1026 1027void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1028{ 1029 1030 Mutex::Autolock _l(mLock); 1031 1032 pid_t pid = IPCThreadState::self()->getCallingPid(); 1033 if (mNotificationClients.indexOfKey(pid) < 0) { 1034 sp<NotificationClient> notificationClient = new NotificationClient(this, 1035 client, 1036 pid); 1037 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1038 1039 mNotificationClients.add(pid, notificationClient); 1040 1041 sp<IBinder> binder = client->asBinder(); 1042 binder->linkToDeath(notificationClient); 1043 1044 // the config change is always sent from playback or record threads to avoid deadlock 1045 // with AudioSystem::gLock 1046 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1047 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1048 } 1049 1050 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1051 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1052 } 1053 } 1054} 1055 1056void AudioFlinger::removeNotificationClient(pid_t pid) 1057{ 1058 Mutex::Autolock _l(mLock); 1059 1060 mNotificationClients.removeItem(pid); 1061 1062 ALOGV("%d died, releasing its sessions", pid); 1063 size_t num = mAudioSessionRefs.size(); 1064 bool removed = false; 1065 for (size_t i = 0; i< num; ) { 1066 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1067 ALOGV(" pid %d @ %d", ref->mPid, i); 1068 if (ref->mPid == pid) { 1069 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1070 mAudioSessionRefs.removeAt(i); 1071 delete ref; 1072 removed = true; 1073 num--; 1074 } else { 1075 i++; 1076 } 1077 } 1078 if (removed) { 1079 purgeStaleEffects_l(); 1080 } 1081} 1082 1083// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1084void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1085{ 1086 size_t size = mNotificationClients.size(); 1087 for (size_t i = 0; i < size; i++) { 1088 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1089 param2); 1090 } 1091} 1092 1093// removeClient_l() must be called with AudioFlinger::mLock held 1094void AudioFlinger::removeClient_l(pid_t pid) 1095{ 1096 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1097 mClients.removeItem(pid); 1098} 1099 1100// getEffectThread_l() must be called with AudioFlinger::mLock held 1101sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1102{ 1103 sp<PlaybackThread> thread; 1104 1105 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1106 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1107 ALOG_ASSERT(thread == 0); 1108 thread = mPlaybackThreads.valueAt(i); 1109 } 1110 } 1111 1112 return thread; 1113} 1114 1115// ---------------------------------------------------------------------------- 1116 1117AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1118 audio_devices_t device, type_t type) 1119 : Thread(false), 1120 mType(type), 1121 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1122 // mChannelMask 1123 mChannelCount(0), 1124 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1125 mParamStatus(NO_ERROR), 1126 mStandby(false), mDevice(device), mId(id), 1127 mDeathRecipient(new PMDeathRecipient(this)) 1128{ 1129} 1130 1131AudioFlinger::ThreadBase::~ThreadBase() 1132{ 1133 mParamCond.broadcast(); 1134 // do not lock the mutex in destructor 1135 releaseWakeLock_l(); 1136 if (mPowerManager != 0) { 1137 sp<IBinder> binder = mPowerManager->asBinder(); 1138 binder->unlinkToDeath(mDeathRecipient); 1139 } 1140} 1141 1142void AudioFlinger::ThreadBase::exit() 1143{ 1144 ALOGV("ThreadBase::exit"); 1145 { 1146 // This lock prevents the following race in thread (uniprocessor for illustration): 1147 // if (!exitPending()) { 1148 // // context switch from here to exit() 1149 // // exit() calls requestExit(), what exitPending() observes 1150 // // exit() calls signal(), which is dropped since no waiters 1151 // // context switch back from exit() to here 1152 // mWaitWorkCV.wait(...); 1153 // // now thread is hung 1154 // } 1155 AutoMutex lock(mLock); 1156 requestExit(); 1157 mWaitWorkCV.signal(); 1158 } 1159 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1160 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1161 requestExitAndWait(); 1162} 1163 1164status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1165{ 1166 status_t status; 1167 1168 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1169 Mutex::Autolock _l(mLock); 1170 1171 mNewParameters.add(keyValuePairs); 1172 mWaitWorkCV.signal(); 1173 // wait condition with timeout in case the thread loop has exited 1174 // before the request could be processed 1175 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1176 status = mParamStatus; 1177 mWaitWorkCV.signal(); 1178 } else { 1179 status = TIMED_OUT; 1180 } 1181 return status; 1182} 1183 1184void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1185{ 1186 Mutex::Autolock _l(mLock); 1187 sendConfigEvent_l(event, param); 1188} 1189 1190// sendConfigEvent_l() must be called with ThreadBase::mLock held 1191void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1192{ 1193 ConfigEvent configEvent; 1194 configEvent.mEvent = event; 1195 configEvent.mParam = param; 1196 mConfigEvents.add(configEvent); 1197 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1198 mWaitWorkCV.signal(); 1199} 1200 1201void AudioFlinger::ThreadBase::processConfigEvents() 1202{ 1203 mLock.lock(); 1204 while (!mConfigEvents.isEmpty()) { 1205 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1206 ConfigEvent configEvent = mConfigEvents[0]; 1207 mConfigEvents.removeAt(0); 1208 // release mLock before locking AudioFlinger mLock: lock order is always 1209 // AudioFlinger then ThreadBase to avoid cross deadlock 1210 mLock.unlock(); 1211 mAudioFlinger->mLock.lock(); 1212 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1213 mAudioFlinger->mLock.unlock(); 1214 mLock.lock(); 1215 } 1216 mLock.unlock(); 1217} 1218 1219void AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1220{ 1221 const size_t SIZE = 256; 1222 char buffer[SIZE]; 1223 String8 result; 1224 1225 bool locked = tryLock(mLock); 1226 if (!locked) { 1227 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1228 write(fd, buffer, strlen(buffer)); 1229 } 1230 1231 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1238 result.append(buffer); 1239 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1240 result.append(buffer); 1241 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1242 result.append(buffer); 1243 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1244 result.append(buffer); 1245 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1246 result.append(buffer); 1247 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1248 result.append(buffer); 1249 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1250 result.append(buffer); 1251 1252 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1253 result.append(buffer); 1254 result.append(" Index Command"); 1255 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1256 snprintf(buffer, SIZE, "\n %02d ", i); 1257 result.append(buffer); 1258 result.append(mNewParameters[i]); 1259 } 1260 1261 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1262 result.append(buffer); 1263 snprintf(buffer, SIZE, " Index event param\n"); 1264 result.append(buffer); 1265 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1266 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1267 result.append(buffer); 1268 } 1269 result.append("\n"); 1270 1271 write(fd, result.string(), result.size()); 1272 1273 if (locked) { 1274 mLock.unlock(); 1275 } 1276} 1277 1278void AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1279{ 1280 const size_t SIZE = 256; 1281 char buffer[SIZE]; 1282 String8 result; 1283 1284 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1285 write(fd, buffer, strlen(buffer)); 1286 1287 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1288 sp<EffectChain> chain = mEffectChains[i]; 1289 if (chain != 0) { 1290 chain->dump(fd, args); 1291 } 1292 } 1293} 1294 1295void AudioFlinger::ThreadBase::acquireWakeLock() 1296{ 1297 Mutex::Autolock _l(mLock); 1298 acquireWakeLock_l(); 1299} 1300 1301void AudioFlinger::ThreadBase::acquireWakeLock_l() 1302{ 1303 if (mPowerManager == 0) { 1304 // use checkService() to avoid blocking if power service is not up yet 1305 sp<IBinder> binder = 1306 defaultServiceManager()->checkService(String16("power")); 1307 if (binder == 0) { 1308 ALOGW("Thread %s cannot connect to the power manager service", mName); 1309 } else { 1310 mPowerManager = interface_cast<IPowerManager>(binder); 1311 binder->linkToDeath(mDeathRecipient); 1312 } 1313 } 1314 if (mPowerManager != 0) { 1315 sp<IBinder> binder = new BBinder(); 1316 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1317 binder, 1318 String16(mName)); 1319 if (status == NO_ERROR) { 1320 mWakeLockToken = binder; 1321 } 1322 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1323 } 1324} 1325 1326void AudioFlinger::ThreadBase::releaseWakeLock() 1327{ 1328 Mutex::Autolock _l(mLock); 1329 releaseWakeLock_l(); 1330} 1331 1332void AudioFlinger::ThreadBase::releaseWakeLock_l() 1333{ 1334 if (mWakeLockToken != 0) { 1335 ALOGV("releaseWakeLock_l() %s", mName); 1336 if (mPowerManager != 0) { 1337 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1338 } 1339 mWakeLockToken.clear(); 1340 } 1341} 1342 1343void AudioFlinger::ThreadBase::clearPowerManager() 1344{ 1345 Mutex::Autolock _l(mLock); 1346 releaseWakeLock_l(); 1347 mPowerManager.clear(); 1348} 1349 1350void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1351{ 1352 sp<ThreadBase> thread = mThread.promote(); 1353 if (thread != 0) { 1354 thread->clearPowerManager(); 1355 } 1356 ALOGW("power manager service died !!!"); 1357} 1358 1359void AudioFlinger::ThreadBase::setEffectSuspended( 1360 const effect_uuid_t *type, bool suspend, int sessionId) 1361{ 1362 Mutex::Autolock _l(mLock); 1363 setEffectSuspended_l(type, suspend, sessionId); 1364} 1365 1366void AudioFlinger::ThreadBase::setEffectSuspended_l( 1367 const effect_uuid_t *type, bool suspend, int sessionId) 1368{ 1369 sp<EffectChain> chain = getEffectChain_l(sessionId); 1370 if (chain != 0) { 1371 if (type != NULL) { 1372 chain->setEffectSuspended_l(type, suspend); 1373 } else { 1374 chain->setEffectSuspendedAll_l(suspend); 1375 } 1376 } 1377 1378 updateSuspendedSessions_l(type, suspend, sessionId); 1379} 1380 1381void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1382{ 1383 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1384 if (index < 0) { 1385 return; 1386 } 1387 1388 const KeyedVector <int, sp<SuspendedSessionDesc> >& sessionEffects = 1389 mSuspendedSessions.valueAt(index); 1390 1391 for (size_t i = 0; i < sessionEffects.size(); i++) { 1392 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1393 for (int j = 0; j < desc->mRefCount; j++) { 1394 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1395 chain->setEffectSuspendedAll_l(true); 1396 } else { 1397 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1398 desc->mType.timeLow); 1399 chain->setEffectSuspended_l(&desc->mType, true); 1400 } 1401 } 1402 } 1403} 1404 1405void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1406 bool suspend, 1407 int sessionId) 1408{ 1409 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1410 1411 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1412 1413 if (suspend) { 1414 if (index >= 0) { 1415 sessionEffects = mSuspendedSessions.valueAt(index); 1416 } else { 1417 mSuspendedSessions.add(sessionId, sessionEffects); 1418 } 1419 } else { 1420 if (index < 0) { 1421 return; 1422 } 1423 sessionEffects = mSuspendedSessions.valueAt(index); 1424 } 1425 1426 1427 int key = EffectChain::kKeyForSuspendAll; 1428 if (type != NULL) { 1429 key = type->timeLow; 1430 } 1431 index = sessionEffects.indexOfKey(key); 1432 1433 sp<SuspendedSessionDesc> desc; 1434 if (suspend) { 1435 if (index >= 0) { 1436 desc = sessionEffects.valueAt(index); 1437 } else { 1438 desc = new SuspendedSessionDesc(); 1439 if (type != NULL) { 1440 desc->mType = *type; 1441 } 1442 sessionEffects.add(key, desc); 1443 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1444 } 1445 desc->mRefCount++; 1446 } else { 1447 if (index < 0) { 1448 return; 1449 } 1450 desc = sessionEffects.valueAt(index); 1451 if (--desc->mRefCount == 0) { 1452 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1453 sessionEffects.removeItemsAt(index); 1454 if (sessionEffects.isEmpty()) { 1455 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1456 sessionId); 1457 mSuspendedSessions.removeItem(sessionId); 1458 } 1459 } 1460 } 1461 if (!sessionEffects.isEmpty()) { 1462 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1463 } 1464} 1465 1466void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1467 bool enabled, 1468 int sessionId) 1469{ 1470 Mutex::Autolock _l(mLock); 1471 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1472} 1473 1474void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1475 bool enabled, 1476 int sessionId) 1477{ 1478 if (mType != RECORD) { 1479 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1480 // another session. This gives the priority to well behaved effect control panels 1481 // and applications not using global effects. 1482 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1483 // global effects 1484 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1485 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1486 } 1487 } 1488 1489 sp<EffectChain> chain = getEffectChain_l(sessionId); 1490 if (chain != 0) { 1491 chain->checkSuspendOnEffectEnabled(effect, enabled); 1492 } 1493} 1494 1495// ---------------------------------------------------------------------------- 1496 1497AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1498 AudioStreamOut* output, 1499 audio_io_handle_t id, 1500 audio_devices_t device, 1501 type_t type) 1502 : ThreadBase(audioFlinger, id, device, type), 1503 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1504 // Assumes constructor is called by AudioFlinger with it's mLock held, 1505 // but it would be safer to explicitly pass initial masterMute as parameter 1506 mMasterMute(audioFlinger->masterMute_l()), 1507 // mStreamTypes[] initialized in constructor body 1508 mOutput(output), 1509 // Assumes constructor is called by AudioFlinger with it's mLock held, 1510 // but it would be safer to explicitly pass initial masterVolume as parameter 1511 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1512 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1513 mMixerStatus(MIXER_IDLE), 1514 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1515 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1516 mScreenState(gScreenState), 1517 // index 0 is reserved for normal mixer's submix 1518 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1519{ 1520 snprintf(mName, kNameLength, "AudioOut_%X", id); 1521 1522 readOutputParameters(); 1523 1524 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1525 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1526 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1527 stream = (audio_stream_type_t) (stream + 1)) { 1528 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1529 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1530 } 1531 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1532 // because mAudioFlinger doesn't have one to copy from 1533} 1534 1535AudioFlinger::PlaybackThread::~PlaybackThread() 1536{ 1537 delete [] mMixBuffer; 1538} 1539 1540void AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1541{ 1542 dumpInternals(fd, args); 1543 dumpTracks(fd, args); 1544 dumpEffectChains(fd, args); 1545} 1546 1547void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1548{ 1549 const size_t SIZE = 256; 1550 char buffer[SIZE]; 1551 String8 result; 1552 1553 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1554 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1555 const stream_type_t *st = &mStreamTypes[i]; 1556 if (i > 0) { 1557 result.appendFormat(", "); 1558 } 1559 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1560 if (st->mute) { 1561 result.append("M"); 1562 } 1563 } 1564 result.append("\n"); 1565 write(fd, result.string(), result.length()); 1566 result.clear(); 1567 1568 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1569 result.append(buffer); 1570 Track::appendDumpHeader(result); 1571 for (size_t i = 0; i < mTracks.size(); ++i) { 1572 sp<Track> track = mTracks[i]; 1573 if (track != 0) { 1574 track->dump(buffer, SIZE); 1575 result.append(buffer); 1576 } 1577 } 1578 1579 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1580 result.append(buffer); 1581 Track::appendDumpHeader(result); 1582 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1583 sp<Track> track = mActiveTracks[i].promote(); 1584 if (track != 0) { 1585 track->dump(buffer, SIZE); 1586 result.append(buffer); 1587 } 1588 } 1589 write(fd, result.string(), result.size()); 1590 1591 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1592 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1593 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1594 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1595} 1596 1597void AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1598{ 1599 const size_t SIZE = 256; 1600 char buffer[SIZE]; 1601 String8 result; 1602 1603 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1604 result.append(buffer); 1605 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1606 result.append(buffer); 1607 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1608 result.append(buffer); 1609 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1610 result.append(buffer); 1611 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1612 result.append(buffer); 1613 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1614 result.append(buffer); 1615 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1616 result.append(buffer); 1617 write(fd, result.string(), result.size()); 1618 fdprintf(fd, "Fast track availMask=%#x\n", mFastTrackAvailMask); 1619 1620 dumpBase(fd, args); 1621} 1622 1623// Thread virtuals 1624status_t AudioFlinger::PlaybackThread::readyToRun() 1625{ 1626 status_t status = initCheck(); 1627 if (status == NO_ERROR) { 1628 ALOGI("AudioFlinger's thread %p ready to run", this); 1629 } else { 1630 ALOGE("No working audio driver found."); 1631 } 1632 return status; 1633} 1634 1635void AudioFlinger::PlaybackThread::onFirstRef() 1636{ 1637 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1638} 1639 1640// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1641sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1642 const sp<AudioFlinger::Client>& client, 1643 audio_stream_type_t streamType, 1644 uint32_t sampleRate, 1645 audio_format_t format, 1646 audio_channel_mask_t channelMask, 1647 int frameCount, 1648 const sp<IMemory>& sharedBuffer, 1649 int sessionId, 1650 IAudioFlinger::track_flags_t flags, 1651 pid_t tid, 1652 status_t *status) 1653{ 1654 sp<Track> track; 1655 status_t lStatus; 1656 1657 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1658 1659 // client expresses a preference for FAST, but we get the final say 1660 if (flags & IAudioFlinger::TRACK_FAST) { 1661 if ( 1662 // not timed 1663 (!isTimed) && 1664 // either of these use cases: 1665 ( 1666 // use case 1: shared buffer with any frame count 1667 ( 1668 (sharedBuffer != 0) 1669 ) || 1670 // use case 2: callback handler and frame count is default or at least as large as HAL 1671 ( 1672 (tid != -1) && 1673 ((frameCount == 0) || 1674 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1675 ) 1676 ) && 1677 // PCM data 1678 audio_is_linear_pcm(format) && 1679 // mono or stereo 1680 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1681 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1682#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1683 // hardware sample rate 1684 (sampleRate == mSampleRate) && 1685#endif 1686 // normal mixer has an associated fast mixer 1687 hasFastMixer() && 1688 // there are sufficient fast track slots available 1689 (mFastTrackAvailMask != 0) 1690 // FIXME test that MixerThread for this fast track has a capable output HAL 1691 // FIXME add a permission test also? 1692 ) { 1693 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1694 if (frameCount == 0) { 1695 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1696 } 1697 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1698 frameCount, mFrameCount); 1699 } else { 1700 ALOGV("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1701 "mFrameCount=%d format=%d isLinear=%d channelMask=%#x sampleRate=%d mSampleRate=%d " 1702 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1703 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1704 audio_is_linear_pcm(format), 1705 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1706 flags &= ~IAudioFlinger::TRACK_FAST; 1707 // For compatibility with AudioTrack calculation, buffer depth is forced 1708 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1709 // This is probably too conservative, but legacy application code may depend on it. 1710 // If you change this calculation, also review the start threshold which is related. 1711 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1712 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1713 if (minBufCount < 2) { 1714 minBufCount = 2; 1715 } 1716 int minFrameCount = mNormalFrameCount * minBufCount; 1717 if (frameCount < minFrameCount) { 1718 frameCount = minFrameCount; 1719 } 1720 } 1721 } 1722 1723 if (mType == DIRECT) { 1724 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1725 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1726 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1727 "for output %p with format %d", 1728 sampleRate, format, channelMask, mOutput, mFormat); 1729 lStatus = BAD_VALUE; 1730 goto Exit; 1731 } 1732 } 1733 } else { 1734 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1735 if (sampleRate > mSampleRate*2) { 1736 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1737 lStatus = BAD_VALUE; 1738 goto Exit; 1739 } 1740 } 1741 1742 lStatus = initCheck(); 1743 if (lStatus != NO_ERROR) { 1744 ALOGE("Audio driver not initialized."); 1745 goto Exit; 1746 } 1747 1748 { // scope for mLock 1749 Mutex::Autolock _l(mLock); 1750 1751 // all tracks in same audio session must share the same routing strategy otherwise 1752 // conflicts will happen when tracks are moved from one output to another by audio policy 1753 // manager 1754 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1755 for (size_t i = 0; i < mTracks.size(); ++i) { 1756 sp<Track> t = mTracks[i]; 1757 if (t != 0 && !t->isOutputTrack()) { 1758 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1759 if (sessionId == t->sessionId() && strategy != actual) { 1760 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1761 strategy, actual); 1762 lStatus = BAD_VALUE; 1763 goto Exit; 1764 } 1765 } 1766 } 1767 1768 if (!isTimed) { 1769 track = new Track(this, client, streamType, sampleRate, format, 1770 channelMask, frameCount, sharedBuffer, sessionId, flags); 1771 } else { 1772 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1773 channelMask, frameCount, sharedBuffer, sessionId); 1774 } 1775 if (track == 0 || track->getCblk() == NULL || track->name() < 0) { 1776 lStatus = NO_MEMORY; 1777 goto Exit; 1778 } 1779 mTracks.add(track); 1780 1781 sp<EffectChain> chain = getEffectChain_l(sessionId); 1782 if (chain != 0) { 1783 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1784 track->setMainBuffer(chain->inBuffer()); 1785 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1786 chain->incTrackCnt(); 1787 } 1788 } 1789 1790 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1791 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1792 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1793 // so ask activity manager to do this on our behalf 1794 int err = requestPriority(callingPid, tid, kPriorityAudioApp); 1795 if (err != 0) { 1796 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1797 kPriorityAudioApp, callingPid, tid, err); 1798 } 1799 } 1800 1801 lStatus = NO_ERROR; 1802 1803Exit: 1804 if (status) { 1805 *status = lStatus; 1806 } 1807 return track; 1808} 1809 1810uint32_t AudioFlinger::MixerThread::correctLatency(uint32_t latency) const 1811{ 1812 if (mFastMixer != NULL) { 1813 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 1814 latency += (pipe->getAvgFrames() * 1000) / mSampleRate; 1815 } 1816 return latency; 1817} 1818 1819uint32_t AudioFlinger::PlaybackThread::correctLatency(uint32_t latency) const 1820{ 1821 return latency; 1822} 1823 1824uint32_t AudioFlinger::PlaybackThread::latency() const 1825{ 1826 Mutex::Autolock _l(mLock); 1827 return latency_l(); 1828} 1829uint32_t AudioFlinger::PlaybackThread::latency_l() const 1830{ 1831 if (initCheck() == NO_ERROR) { 1832 return correctLatency(mOutput->stream->get_latency(mOutput->stream)); 1833 } else { 1834 return 0; 1835 } 1836} 1837 1838void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1839{ 1840 Mutex::Autolock _l(mLock); 1841 mMasterVolume = value; 1842} 1843 1844void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1845{ 1846 Mutex::Autolock _l(mLock); 1847 setMasterMute_l(muted); 1848} 1849 1850void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1851{ 1852 Mutex::Autolock _l(mLock); 1853 mStreamTypes[stream].volume = value; 1854} 1855 1856void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1857{ 1858 Mutex::Autolock _l(mLock); 1859 mStreamTypes[stream].mute = muted; 1860} 1861 1862float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1863{ 1864 Mutex::Autolock _l(mLock); 1865 return mStreamTypes[stream].volume; 1866} 1867 1868// addTrack_l() must be called with ThreadBase::mLock held 1869status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1870{ 1871 status_t status = ALREADY_EXISTS; 1872 1873 // set retry count for buffer fill 1874 track->mRetryCount = kMaxTrackStartupRetries; 1875 if (mActiveTracks.indexOf(track) < 0) { 1876 // the track is newly added, make sure it fills up all its 1877 // buffers before playing. This is to ensure the client will 1878 // effectively get the latency it requested. 1879 track->mFillingUpStatus = Track::FS_FILLING; 1880 track->mResetDone = false; 1881 track->mPresentationCompleteFrames = 0; 1882 mActiveTracks.add(track); 1883 if (track->mainBuffer() != mMixBuffer) { 1884 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1885 if (chain != 0) { 1886 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1887 chain->incActiveTrackCnt(); 1888 } 1889 } 1890 1891 status = NO_ERROR; 1892 } 1893 1894 ALOGV("mWaitWorkCV.broadcast"); 1895 mWaitWorkCV.broadcast(); 1896 1897 return status; 1898} 1899 1900// destroyTrack_l() must be called with ThreadBase::mLock held 1901void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1902{ 1903 track->mState = TrackBase::TERMINATED; 1904 // active tracks are removed by threadLoop() 1905 if (mActiveTracks.indexOf(track) < 0) { 1906 removeTrack_l(track); 1907 } 1908} 1909 1910void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1911{ 1912 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1913 mTracks.remove(track); 1914 deleteTrackName_l(track->name()); 1915 // redundant as track is about to be destroyed, for dumpsys only 1916 track->mName = -1; 1917 if (track->isFastTrack()) { 1918 int index = track->mFastIndex; 1919 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1920 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1921 mFastTrackAvailMask |= 1 << index; 1922 // redundant as track is about to be destroyed, for dumpsys only 1923 track->mFastIndex = -1; 1924 } 1925 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1926 if (chain != 0) { 1927 chain->decTrackCnt(); 1928 } 1929} 1930 1931String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1932{ 1933 String8 out_s8 = String8(""); 1934 char *s; 1935 1936 Mutex::Autolock _l(mLock); 1937 if (initCheck() != NO_ERROR) { 1938 return out_s8; 1939 } 1940 1941 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1942 out_s8 = String8(s); 1943 free(s); 1944 return out_s8; 1945} 1946 1947// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1948void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1949 AudioSystem::OutputDescriptor desc; 1950 void *param2 = NULL; 1951 1952 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1953 1954 switch (event) { 1955 case AudioSystem::OUTPUT_OPENED: 1956 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1957 desc.channels = mChannelMask; 1958 desc.samplingRate = mSampleRate; 1959 desc.format = mFormat; 1960 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1961 desc.latency = latency(); 1962 param2 = &desc; 1963 break; 1964 1965 case AudioSystem::STREAM_CONFIG_CHANGED: 1966 param2 = ¶m; 1967 case AudioSystem::OUTPUT_CLOSED: 1968 default: 1969 break; 1970 } 1971 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1972} 1973 1974void AudioFlinger::PlaybackThread::readOutputParameters() 1975{ 1976 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1977 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1978 mChannelCount = (uint16_t)popcount(mChannelMask); 1979 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1980 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1981 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1982 if (mFrameCount & 15) { 1983 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1984 mFrameCount); 1985 } 1986 1987 // Calculate size of normal mix buffer relative to the HAL output buffer size 1988 double multiplier = 1.0; 1989 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1990 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1991 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1992 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1993 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1994 maxNormalFrameCount = maxNormalFrameCount & ~15; 1995 if (maxNormalFrameCount < minNormalFrameCount) { 1996 maxNormalFrameCount = minNormalFrameCount; 1997 } 1998 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1999 if (multiplier <= 1.0) { 2000 multiplier = 1.0; 2001 } else if (multiplier <= 2.0) { 2002 if (2 * mFrameCount <= maxNormalFrameCount) { 2003 multiplier = 2.0; 2004 } else { 2005 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 2006 } 2007 } else { 2008 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 2009 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 2010 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 2011 // FIXME this rounding up should not be done if no HAL SRC 2012 uint32_t truncMult = (uint32_t) multiplier; 2013 if ((truncMult & 1)) { 2014 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 2015 ++truncMult; 2016 } 2017 } 2018 multiplier = (double) truncMult; 2019 } 2020 } 2021 mNormalFrameCount = multiplier * mFrameCount; 2022 // round up to nearest 16 frames to satisfy AudioMixer 2023 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2024 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2025 2026 delete[] mMixBuffer; 2027 mMixBuffer = new int16_t[mNormalFrameCount * mChannelCount]; 2028 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 2029 2030 // force reconfiguration of effect chains and engines to take new buffer size and audio 2031 // parameters into account 2032 // Note that mLock is not held when readOutputParameters() is called from the constructor 2033 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2034 // matter. 2035 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2036 Vector< sp<EffectChain> > effectChains = mEffectChains; 2037 for (size_t i = 0; i < effectChains.size(); i ++) { 2038 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2039 } 2040} 2041 2042 2043status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2044{ 2045 if (halFrames == NULL || dspFrames == NULL) { 2046 return BAD_VALUE; 2047 } 2048 Mutex::Autolock _l(mLock); 2049 if (initCheck() != NO_ERROR) { 2050 return INVALID_OPERATION; 2051 } 2052 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2053 2054 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2055} 2056 2057uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2058{ 2059 Mutex::Autolock _l(mLock); 2060 uint32_t result = 0; 2061 if (getEffectChain_l(sessionId) != 0) { 2062 result = EFFECT_SESSION; 2063 } 2064 2065 for (size_t i = 0; i < mTracks.size(); ++i) { 2066 sp<Track> track = mTracks[i]; 2067 if (sessionId == track->sessionId() && 2068 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2069 result |= TRACK_SESSION; 2070 break; 2071 } 2072 } 2073 2074 return result; 2075} 2076 2077uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2078{ 2079 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2080 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2081 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2082 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2083 } 2084 for (size_t i = 0; i < mTracks.size(); i++) { 2085 sp<Track> track = mTracks[i]; 2086 if (sessionId == track->sessionId() && 2087 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2088 return AudioSystem::getStrategyForStream(track->streamType()); 2089 } 2090 } 2091 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2092} 2093 2094 2095AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2096{ 2097 Mutex::Autolock _l(mLock); 2098 return mOutput; 2099} 2100 2101AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2102{ 2103 Mutex::Autolock _l(mLock); 2104 AudioStreamOut *output = mOutput; 2105 mOutput = NULL; 2106 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2107 // must push a NULL and wait for ack 2108 mOutputSink.clear(); 2109 mPipeSink.clear(); 2110 mNormalSink.clear(); 2111 return output; 2112} 2113 2114// this method must always be called either with ThreadBase mLock held or inside the thread loop 2115audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2116{ 2117 if (mOutput == NULL) { 2118 return NULL; 2119 } 2120 return &mOutput->stream->common; 2121} 2122 2123uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2124{ 2125 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2126} 2127 2128status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2129{ 2130 if (!isValidSyncEvent(event)) { 2131 return BAD_VALUE; 2132 } 2133 2134 Mutex::Autolock _l(mLock); 2135 2136 for (size_t i = 0; i < mTracks.size(); ++i) { 2137 sp<Track> track = mTracks[i]; 2138 if (event->triggerSession() == track->sessionId()) { 2139 track->setSyncEvent(event); 2140 return NO_ERROR; 2141 } 2142 } 2143 2144 return NAME_NOT_FOUND; 2145} 2146 2147bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2148{ 2149 switch (event->type()) { 2150 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2151 return true; 2152 default: 2153 break; 2154 } 2155 return false; 2156} 2157 2158void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2159{ 2160 size_t count = tracksToRemove.size(); 2161 if (CC_UNLIKELY(count)) { 2162 for (size_t i = 0 ; i < count ; i++) { 2163 const sp<Track>& track = tracksToRemove.itemAt(i); 2164 if ((track->sharedBuffer() != 0) && 2165 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2166 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2167 } 2168 } 2169 } 2170 2171} 2172 2173// ---------------------------------------------------------------------------- 2174 2175AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2176 audio_io_handle_t id, audio_devices_t device, type_t type) 2177 : PlaybackThread(audioFlinger, output, id, device, type), 2178 // mAudioMixer below 2179 // mFastMixer below 2180 mFastMixerFutex(0) 2181 // mOutputSink below 2182 // mPipeSink below 2183 // mNormalSink below 2184{ 2185 ALOGV("MixerThread() id=%d device=%#x type=%d", id, device, type); 2186 ALOGV("mSampleRate=%d, mChannelMask=%#x, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2187 "mFrameCount=%d, mNormalFrameCount=%d", 2188 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2189 mNormalFrameCount); 2190 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2191 2192 // FIXME - Current mixer implementation only supports stereo output 2193 if (mChannelCount != FCC_2) { 2194 ALOGE("Invalid audio hardware channel count %d", mChannelCount); 2195 } 2196 2197 // create an NBAIO sink for the HAL output stream, and negotiate 2198 mOutputSink = new AudioStreamOutSink(output->stream); 2199 size_t numCounterOffers = 0; 2200 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2201 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2202 ALOG_ASSERT(index == 0); 2203 2204 // initialize fast mixer depending on configuration 2205 bool initFastMixer; 2206 switch (kUseFastMixer) { 2207 case FastMixer_Never: 2208 initFastMixer = false; 2209 break; 2210 case FastMixer_Always: 2211 initFastMixer = true; 2212 break; 2213 case FastMixer_Static: 2214 case FastMixer_Dynamic: 2215 initFastMixer = mFrameCount < mNormalFrameCount; 2216 break; 2217 } 2218 if (initFastMixer) { 2219 2220 // create a MonoPipe to connect our submix to FastMixer 2221 NBAIO_Format format = mOutputSink->format(); 2222 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2223 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2224 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2225 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2226 const NBAIO_Format offers[1] = {format}; 2227 size_t numCounterOffers = 0; 2228 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2229 ALOG_ASSERT(index == 0); 2230 monoPipe->setAvgFrames((mScreenState & 1) ? 2231 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2232 mPipeSink = monoPipe; 2233 2234#ifdef TEE_SINK_FRAMES 2235 // create a Pipe to archive a copy of FastMixer's output for dumpsys 2236 Pipe *teeSink = new Pipe(TEE_SINK_FRAMES, format); 2237 numCounterOffers = 0; 2238 index = teeSink->negotiate(offers, 1, NULL, numCounterOffers); 2239 ALOG_ASSERT(index == 0); 2240 mTeeSink = teeSink; 2241 PipeReader *teeSource = new PipeReader(*teeSink); 2242 numCounterOffers = 0; 2243 index = teeSource->negotiate(offers, 1, NULL, numCounterOffers); 2244 ALOG_ASSERT(index == 0); 2245 mTeeSource = teeSource; 2246#endif 2247 2248 // create fast mixer and configure it initially with just one fast track for our submix 2249 mFastMixer = new FastMixer(); 2250 FastMixerStateQueue *sq = mFastMixer->sq(); 2251#ifdef STATE_QUEUE_DUMP 2252 sq->setObserverDump(&mStateQueueObserverDump); 2253 sq->setMutatorDump(&mStateQueueMutatorDump); 2254#endif 2255 FastMixerState *state = sq->begin(); 2256 FastTrack *fastTrack = &state->mFastTracks[0]; 2257 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2258 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2259 fastTrack->mVolumeProvider = NULL; 2260 fastTrack->mGeneration++; 2261 state->mFastTracksGen++; 2262 state->mTrackMask = 1; 2263 // fast mixer will use the HAL output sink 2264 state->mOutputSink = mOutputSink.get(); 2265 state->mOutputSinkGen++; 2266 state->mFrameCount = mFrameCount; 2267 state->mCommand = FastMixerState::COLD_IDLE; 2268 // already done in constructor initialization list 2269 //mFastMixerFutex = 0; 2270 state->mColdFutexAddr = &mFastMixerFutex; 2271 state->mColdGen++; 2272 state->mDumpState = &mFastMixerDumpState; 2273 state->mTeeSink = mTeeSink.get(); 2274 sq->end(); 2275 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2276 2277 // start the fast mixer 2278 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2279 pid_t tid = mFastMixer->getTid(); 2280 int err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2281 if (err != 0) { 2282 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2283 kPriorityFastMixer, getpid_cached, tid, err); 2284 } 2285 2286#ifdef AUDIO_WATCHDOG 2287 // create and start the watchdog 2288 mAudioWatchdog = new AudioWatchdog(); 2289 mAudioWatchdog->setDump(&mAudioWatchdogDump); 2290 mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO); 2291 tid = mAudioWatchdog->getTid(); 2292 err = requestPriority(getpid_cached, tid, kPriorityFastMixer); 2293 if (err != 0) { 2294 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2295 kPriorityFastMixer, getpid_cached, tid, err); 2296 } 2297#endif 2298 2299 } else { 2300 mFastMixer = NULL; 2301 } 2302 2303 switch (kUseFastMixer) { 2304 case FastMixer_Never: 2305 case FastMixer_Dynamic: 2306 mNormalSink = mOutputSink; 2307 break; 2308 case FastMixer_Always: 2309 mNormalSink = mPipeSink; 2310 break; 2311 case FastMixer_Static: 2312 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2313 break; 2314 } 2315} 2316 2317AudioFlinger::MixerThread::~MixerThread() 2318{ 2319 if (mFastMixer != NULL) { 2320 FastMixerStateQueue *sq = mFastMixer->sq(); 2321 FastMixerState *state = sq->begin(); 2322 if (state->mCommand == FastMixerState::COLD_IDLE) { 2323 int32_t old = android_atomic_inc(&mFastMixerFutex); 2324 if (old == -1) { 2325 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2326 } 2327 } 2328 state->mCommand = FastMixerState::EXIT; 2329 sq->end(); 2330 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2331 mFastMixer->join(); 2332 // Though the fast mixer thread has exited, it's state queue is still valid. 2333 // We'll use that extract the final state which contains one remaining fast track 2334 // corresponding to our sub-mix. 2335 state = sq->begin(); 2336 ALOG_ASSERT(state->mTrackMask == 1); 2337 FastTrack *fastTrack = &state->mFastTracks[0]; 2338 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2339 delete fastTrack->mBufferProvider; 2340 sq->end(false /*didModify*/); 2341 delete mFastMixer; 2342 if (mAudioWatchdog != 0) { 2343 mAudioWatchdog->requestExit(); 2344 mAudioWatchdog->requestExitAndWait(); 2345 mAudioWatchdog.clear(); 2346 } 2347 } 2348 delete mAudioMixer; 2349} 2350 2351class CpuStats { 2352public: 2353 CpuStats(); 2354 void sample(const String8 &title); 2355#ifdef DEBUG_CPU_USAGE 2356private: 2357 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2358 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2359 2360 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2361 2362 int mCpuNum; // thread's current CPU number 2363 int mCpukHz; // frequency of thread's current CPU in kHz 2364#endif 2365}; 2366 2367CpuStats::CpuStats() 2368#ifdef DEBUG_CPU_USAGE 2369 : mCpuNum(-1), mCpukHz(-1) 2370#endif 2371{ 2372} 2373 2374void CpuStats::sample(const String8 &title) { 2375#ifdef DEBUG_CPU_USAGE 2376 // get current thread's delta CPU time in wall clock ns 2377 double wcNs; 2378 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2379 2380 // record sample for wall clock statistics 2381 if (valid) { 2382 mWcStats.sample(wcNs); 2383 } 2384 2385 // get the current CPU number 2386 int cpuNum = sched_getcpu(); 2387 2388 // get the current CPU frequency in kHz 2389 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2390 2391 // check if either CPU number or frequency changed 2392 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2393 mCpuNum = cpuNum; 2394 mCpukHz = cpukHz; 2395 // ignore sample for purposes of cycles 2396 valid = false; 2397 } 2398 2399 // if no change in CPU number or frequency, then record sample for cycle statistics 2400 if (valid && mCpukHz > 0) { 2401 double cycles = wcNs * cpukHz * 0.000001; 2402 mHzStats.sample(cycles); 2403 } 2404 2405 unsigned n = mWcStats.n(); 2406 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2407 if ((n & 127) == 1) { 2408 long long elapsed = mCpuUsage.elapsed(); 2409 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2410 double perLoop = elapsed / (double) n; 2411 double perLoop100 = perLoop * 0.01; 2412 double perLoop1k = perLoop * 0.001; 2413 double mean = mWcStats.mean(); 2414 double stddev = mWcStats.stddev(); 2415 double minimum = mWcStats.minimum(); 2416 double maximum = mWcStats.maximum(); 2417 double meanCycles = mHzStats.mean(); 2418 double stddevCycles = mHzStats.stddev(); 2419 double minCycles = mHzStats.minimum(); 2420 double maxCycles = mHzStats.maximum(); 2421 mCpuUsage.resetElapsed(); 2422 mWcStats.reset(); 2423 mHzStats.reset(); 2424 ALOGD("CPU usage for %s over past %.1f secs\n" 2425 " (%u mixer loops at %.1f mean ms per loop):\n" 2426 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2427 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2428 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2429 title.string(), 2430 elapsed * .000000001, n, perLoop * .000001, 2431 mean * .001, 2432 stddev * .001, 2433 minimum * .001, 2434 maximum * .001, 2435 mean / perLoop100, 2436 stddev / perLoop100, 2437 minimum / perLoop100, 2438 maximum / perLoop100, 2439 meanCycles / perLoop1k, 2440 stddevCycles / perLoop1k, 2441 minCycles / perLoop1k, 2442 maxCycles / perLoop1k); 2443 2444 } 2445 } 2446#endif 2447}; 2448 2449void AudioFlinger::PlaybackThread::checkSilentMode_l() 2450{ 2451 if (!mMasterMute) { 2452 char value[PROPERTY_VALUE_MAX]; 2453 if (property_get("ro.audio.silent", value, "0") > 0) { 2454 char *endptr; 2455 unsigned long ul = strtoul(value, &endptr, 0); 2456 if (*endptr == '\0' && ul != 0) { 2457 ALOGD("Silence is golden"); 2458 // The setprop command will not allow a property to be changed after 2459 // the first time it is set, so we don't have to worry about un-muting. 2460 setMasterMute_l(true); 2461 } 2462 } 2463 } 2464} 2465 2466bool AudioFlinger::PlaybackThread::threadLoop() 2467{ 2468 Vector< sp<Track> > tracksToRemove; 2469 2470 standbyTime = systemTime(); 2471 2472 // MIXER 2473 nsecs_t lastWarning = 0; 2474 2475 // DUPLICATING 2476 // FIXME could this be made local to while loop? 2477 writeFrames = 0; 2478 2479 cacheParameters_l(); 2480 sleepTime = idleSleepTime; 2481 2482 if (mType == MIXER) { 2483 sleepTimeShift = 0; 2484 } 2485 2486 CpuStats cpuStats; 2487 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2488 2489 acquireWakeLock(); 2490 2491 while (!exitPending()) 2492 { 2493 cpuStats.sample(myName); 2494 2495 Vector< sp<EffectChain> > effectChains; 2496 2497 processConfigEvents(); 2498 2499 { // scope for mLock 2500 2501 Mutex::Autolock _l(mLock); 2502 2503 if (checkForNewParameters_l()) { 2504 cacheParameters_l(); 2505 } 2506 2507 saveOutputTracks(); 2508 2509 // put audio hardware into standby after short delay 2510 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2511 isSuspended())) { 2512 if (!mStandby) { 2513 2514 threadLoop_standby(); 2515 2516 mStandby = true; 2517 mBytesWritten = 0; 2518 } 2519 2520 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2521 // we're about to wait, flush the binder command buffer 2522 IPCThreadState::self()->flushCommands(); 2523 2524 clearOutputTracks(); 2525 2526 if (exitPending()) break; 2527 2528 releaseWakeLock_l(); 2529 // wait until we have something to do... 2530 ALOGV("%s going to sleep", myName.string()); 2531 mWaitWorkCV.wait(mLock); 2532 ALOGV("%s waking up", myName.string()); 2533 acquireWakeLock_l(); 2534 2535 mMixerStatus = MIXER_IDLE; 2536 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2537 2538 checkSilentMode_l(); 2539 2540 standbyTime = systemTime() + standbyDelay; 2541 sleepTime = idleSleepTime; 2542 if (mType == MIXER) { 2543 sleepTimeShift = 0; 2544 } 2545 2546 continue; 2547 } 2548 } 2549 2550 // mMixerStatusIgnoringFastTracks is also updated internally 2551 mMixerStatus = prepareTracks_l(&tracksToRemove); 2552 2553 // prevent any changes in effect chain list and in each effect chain 2554 // during mixing and effect process as the audio buffers could be deleted 2555 // or modified if an effect is created or deleted 2556 lockEffectChains_l(effectChains); 2557 } 2558 2559 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2560 threadLoop_mix(); 2561 } else { 2562 threadLoop_sleepTime(); 2563 } 2564 2565 if (isSuspended()) { 2566 sleepTime = suspendSleepTimeUs(); 2567 } 2568 2569 // only process effects if we're going to write 2570 if (sleepTime == 0) { 2571 for (size_t i = 0; i < effectChains.size(); i ++) { 2572 effectChains[i]->process_l(); 2573 } 2574 } 2575 2576 // enable changes in effect chain 2577 unlockEffectChains(effectChains); 2578 2579 // sleepTime == 0 means we must write to audio hardware 2580 if (sleepTime == 0) { 2581 2582 threadLoop_write(); 2583 2584if (mType == MIXER) { 2585 // write blocked detection 2586 nsecs_t now = systemTime(); 2587 nsecs_t delta = now - mLastWriteTime; 2588 if (!mStandby && delta > maxPeriod) { 2589 mNumDelayedWrites++; 2590 if ((now - lastWarning) > kWarningThrottleNs) { 2591#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2592 ScopedTrace st(ATRACE_TAG, "underrun"); 2593#endif 2594 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2595 ns2ms(delta), mNumDelayedWrites, this); 2596 lastWarning = now; 2597 } 2598 } 2599} 2600 2601 mStandby = false; 2602 } else { 2603 usleep(sleepTime); 2604 } 2605 2606 // Finally let go of removed track(s), without the lock held 2607 // since we can't guarantee the destructors won't acquire that 2608 // same lock. This will also mutate and push a new fast mixer state. 2609 threadLoop_removeTracks(tracksToRemove); 2610 tracksToRemove.clear(); 2611 2612 // FIXME I don't understand the need for this here; 2613 // it was in the original code but maybe the 2614 // assignment in saveOutputTracks() makes this unnecessary? 2615 clearOutputTracks(); 2616 2617 // Effect chains will be actually deleted here if they were removed from 2618 // mEffectChains list during mixing or effects processing 2619 effectChains.clear(); 2620 2621 // FIXME Note that the above .clear() is no longer necessary since effectChains 2622 // is now local to this block, but will keep it for now (at least until merge done). 2623 } 2624 2625 // for DuplicatingThread, standby mode is handled by the outputTracks, otherwise ... 2626 if (mType == MIXER || mType == DIRECT) { 2627 // put output stream into standby mode 2628 if (!mStandby) { 2629 mOutput->stream->common.standby(&mOutput->stream->common); 2630 } 2631 } 2632 2633 releaseWakeLock(); 2634 2635 ALOGV("Thread %p type %d exiting", this, mType); 2636 return false; 2637} 2638 2639void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2640{ 2641 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2642} 2643 2644void AudioFlinger::MixerThread::threadLoop_write() 2645{ 2646 // FIXME we should only do one push per cycle; confirm this is true 2647 // Start the fast mixer if it's not already running 2648 if (mFastMixer != NULL) { 2649 FastMixerStateQueue *sq = mFastMixer->sq(); 2650 FastMixerState *state = sq->begin(); 2651 if (state->mCommand != FastMixerState::MIX_WRITE && 2652 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2653 if (state->mCommand == FastMixerState::COLD_IDLE) { 2654 int32_t old = android_atomic_inc(&mFastMixerFutex); 2655 if (old == -1) { 2656 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2657 } 2658 if (mAudioWatchdog != 0) { 2659 mAudioWatchdog->resume(); 2660 } 2661 } 2662 state->mCommand = FastMixerState::MIX_WRITE; 2663 sq->end(); 2664 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2665 if (kUseFastMixer == FastMixer_Dynamic) { 2666 mNormalSink = mPipeSink; 2667 } 2668 } else { 2669 sq->end(false /*didModify*/); 2670 } 2671 } 2672 PlaybackThread::threadLoop_write(); 2673} 2674 2675// shared by MIXER and DIRECT, overridden by DUPLICATING 2676void AudioFlinger::PlaybackThread::threadLoop_write() 2677{ 2678 // FIXME rewrite to reduce number of system calls 2679 mLastWriteTime = systemTime(); 2680 mInWrite = true; 2681 int bytesWritten; 2682 2683 // If an NBAIO sink is present, use it to write the normal mixer's submix 2684 if (mNormalSink != 0) { 2685#define mBitShift 2 // FIXME 2686 size_t count = mixBufferSize >> mBitShift; 2687#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2688 Tracer::traceBegin(ATRACE_TAG, "write"); 2689#endif 2690 // update the setpoint when gScreenState changes 2691 uint32_t screenState = gScreenState; 2692 if (screenState != mScreenState) { 2693 mScreenState = screenState; 2694 MonoPipe *pipe = (MonoPipe *)mPipeSink.get(); 2695 if (pipe != NULL) { 2696 pipe->setAvgFrames((mScreenState & 1) ? 2697 (pipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2); 2698 } 2699 } 2700 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2701#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2702 Tracer::traceEnd(ATRACE_TAG); 2703#endif 2704 if (framesWritten > 0) { 2705 bytesWritten = framesWritten << mBitShift; 2706 } else { 2707 bytesWritten = framesWritten; 2708 } 2709 // otherwise use the HAL / AudioStreamOut directly 2710 } else { 2711 // Direct output thread. 2712 bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2713 } 2714 2715 if (bytesWritten > 0) mBytesWritten += mixBufferSize; 2716 mNumWrites++; 2717 mInWrite = false; 2718} 2719 2720void AudioFlinger::MixerThread::threadLoop_standby() 2721{ 2722 // Idle the fast mixer if it's currently running 2723 if (mFastMixer != NULL) { 2724 FastMixerStateQueue *sq = mFastMixer->sq(); 2725 FastMixerState *state = sq->begin(); 2726 if (!(state->mCommand & FastMixerState::IDLE)) { 2727 state->mCommand = FastMixerState::COLD_IDLE; 2728 state->mColdFutexAddr = &mFastMixerFutex; 2729 state->mColdGen++; 2730 mFastMixerFutex = 0; 2731 sq->end(); 2732 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2733 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2734 if (kUseFastMixer == FastMixer_Dynamic) { 2735 mNormalSink = mOutputSink; 2736 } 2737 if (mAudioWatchdog != 0) { 2738 mAudioWatchdog->pause(); 2739 } 2740 } else { 2741 sq->end(false /*didModify*/); 2742 } 2743 } 2744 PlaybackThread::threadLoop_standby(); 2745} 2746 2747// shared by MIXER and DIRECT, overridden by DUPLICATING 2748void AudioFlinger::PlaybackThread::threadLoop_standby() 2749{ 2750 ALOGV("Audio hardware entering standby, mixer %p, suspend count %d", this, mSuspended); 2751 mOutput->stream->common.standby(&mOutput->stream->common); 2752} 2753 2754void AudioFlinger::MixerThread::threadLoop_mix() 2755{ 2756 // obtain the presentation timestamp of the next output buffer 2757 int64_t pts; 2758 status_t status = INVALID_OPERATION; 2759 2760 if (NULL != mOutput->stream->get_next_write_timestamp) { 2761 status = mOutput->stream->get_next_write_timestamp( 2762 mOutput->stream, &pts); 2763 } 2764 2765 if (status != NO_ERROR) { 2766 pts = AudioBufferProvider::kInvalidPTS; 2767 } 2768 2769 // mix buffers... 2770 mAudioMixer->process(pts); 2771 // increase sleep time progressively when application underrun condition clears. 2772 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2773 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2774 // such that we would underrun the audio HAL. 2775 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2776 sleepTimeShift--; 2777 } 2778 sleepTime = 0; 2779 standbyTime = systemTime() + standbyDelay; 2780 //TODO: delay standby when effects have a tail 2781} 2782 2783void AudioFlinger::MixerThread::threadLoop_sleepTime() 2784{ 2785 // If no tracks are ready, sleep once for the duration of an output 2786 // buffer size, then write 0s to the output 2787 if (sleepTime == 0) { 2788 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2789 sleepTime = activeSleepTime >> sleepTimeShift; 2790 if (sleepTime < kMinThreadSleepTimeUs) { 2791 sleepTime = kMinThreadSleepTimeUs; 2792 } 2793 // reduce sleep time in case of consecutive application underruns to avoid 2794 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2795 // duration we would end up writing less data than needed by the audio HAL if 2796 // the condition persists. 2797 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2798 sleepTimeShift++; 2799 } 2800 } else { 2801 sleepTime = idleSleepTime; 2802 } 2803 } else if (mBytesWritten != 0 || (mMixerStatus == MIXER_TRACKS_ENABLED)) { 2804 memset (mMixBuffer, 0, mixBufferSize); 2805 sleepTime = 0; 2806 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED)), "anticipated start"); 2807 } 2808 // TODO add standby time extension fct of effect tail 2809} 2810 2811// prepareTracks_l() must be called with ThreadBase::mLock held 2812AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2813 Vector< sp<Track> > *tracksToRemove) 2814{ 2815 2816 mixer_state mixerStatus = MIXER_IDLE; 2817 // find out which tracks need to be processed 2818 size_t count = mActiveTracks.size(); 2819 size_t mixedTracks = 0; 2820 size_t tracksWithEffect = 0; 2821 // counts only _active_ fast tracks 2822 size_t fastTracks = 0; 2823 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2824 2825 float masterVolume = mMasterVolume; 2826 bool masterMute = mMasterMute; 2827 2828 if (masterMute) { 2829 masterVolume = 0; 2830 } 2831 // Delegate master volume control to effect in output mix effect chain if needed 2832 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2833 if (chain != 0) { 2834 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2835 chain->setVolume_l(&v, &v); 2836 masterVolume = (float)((v + (1 << 23)) >> 24); 2837 chain.clear(); 2838 } 2839 2840 // prepare a new state to push 2841 FastMixerStateQueue *sq = NULL; 2842 FastMixerState *state = NULL; 2843 bool didModify = false; 2844 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2845 if (mFastMixer != NULL) { 2846 sq = mFastMixer->sq(); 2847 state = sq->begin(); 2848 } 2849 2850 for (size_t i=0 ; i<count ; i++) { 2851 sp<Track> t = mActiveTracks[i].promote(); 2852 if (t == 0) continue; 2853 2854 // this const just means the local variable doesn't change 2855 Track* const track = t.get(); 2856 2857 // process fast tracks 2858 if (track->isFastTrack()) { 2859 2860 // It's theoretically possible (though unlikely) for a fast track to be created 2861 // and then removed within the same normal mix cycle. This is not a problem, as 2862 // the track never becomes active so it's fast mixer slot is never touched. 2863 // The converse, of removing an (active) track and then creating a new track 2864 // at the identical fast mixer slot within the same normal mix cycle, 2865 // is impossible because the slot isn't marked available until the end of each cycle. 2866 int j = track->mFastIndex; 2867 ALOG_ASSERT(0 < j && j < (int)FastMixerState::kMaxFastTracks); 2868 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << j))); 2869 FastTrack *fastTrack = &state->mFastTracks[j]; 2870 2871 // Determine whether the track is currently in underrun condition, 2872 // and whether it had a recent underrun. 2873 FastTrackDump *ftDump = &mFastMixerDumpState.mTracks[j]; 2874 FastTrackUnderruns underruns = ftDump->mUnderruns; 2875 uint32_t recentFull = (underruns.mBitFields.mFull - 2876 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2877 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2878 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2879 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2880 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2881 uint32_t recentUnderruns = recentPartial + recentEmpty; 2882 track->mObservedUnderruns = underruns; 2883 // don't count underruns that occur while stopping or pausing 2884 // or stopped which can occur when flush() is called while active 2885 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2886 track->mUnderrunCount += recentUnderruns; 2887 } 2888 2889 // This is similar to the state machine for normal tracks, 2890 // with a few modifications for fast tracks. 2891 bool isActive = true; 2892 switch (track->mState) { 2893 case TrackBase::STOPPING_1: 2894 // track stays active in STOPPING_1 state until first underrun 2895 if (recentUnderruns > 0) { 2896 track->mState = TrackBase::STOPPING_2; 2897 } 2898 break; 2899 case TrackBase::PAUSING: 2900 // ramp down is not yet implemented 2901 track->setPaused(); 2902 break; 2903 case TrackBase::RESUMING: 2904 // ramp up is not yet implemented 2905 track->mState = TrackBase::ACTIVE; 2906 break; 2907 case TrackBase::ACTIVE: 2908 if (recentFull > 0 || recentPartial > 0) { 2909 // track has provided at least some frames recently: reset retry count 2910 track->mRetryCount = kMaxTrackRetries; 2911 } 2912 if (recentUnderruns == 0) { 2913 // no recent underruns: stay active 2914 break; 2915 } 2916 // there has recently been an underrun of some kind 2917 if (track->sharedBuffer() == 0) { 2918 // were any of the recent underruns "empty" (no frames available)? 2919 if (recentEmpty == 0) { 2920 // no, then ignore the partial underruns as they are allowed indefinitely 2921 break; 2922 } 2923 // there has recently been an "empty" underrun: decrement the retry counter 2924 if (--(track->mRetryCount) > 0) { 2925 break; 2926 } 2927 // indicate to client process that the track was disabled because of underrun; 2928 // it will then automatically call start() when data is available 2929 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2930 // remove from active list, but state remains ACTIVE [confusing but true] 2931 isActive = false; 2932 break; 2933 } 2934 // fall through 2935 case TrackBase::STOPPING_2: 2936 case TrackBase::PAUSED: 2937 case TrackBase::TERMINATED: 2938 case TrackBase::STOPPED: 2939 case TrackBase::FLUSHED: // flush() while active 2940 // Check for presentation complete if track is inactive 2941 // We have consumed all the buffers of this track. 2942 // This would be incomplete if we auto-paused on underrun 2943 { 2944 size_t audioHALFrames = 2945 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2946 size_t framesWritten = 2947 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2948 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2949 // track stays in active list until presentation is complete 2950 break; 2951 } 2952 } 2953 if (track->isStopping_2()) { 2954 track->mState = TrackBase::STOPPED; 2955 } 2956 if (track->isStopped()) { 2957 // Can't reset directly, as fast mixer is still polling this track 2958 // track->reset(); 2959 // So instead mark this track as needing to be reset after push with ack 2960 resetMask |= 1 << i; 2961 } 2962 isActive = false; 2963 break; 2964 case TrackBase::IDLE: 2965 default: 2966 LOG_FATAL("unexpected track state %d", track->mState); 2967 } 2968 2969 if (isActive) { 2970 // was it previously inactive? 2971 if (!(state->mTrackMask & (1 << j))) { 2972 ExtendedAudioBufferProvider *eabp = track; 2973 VolumeProvider *vp = track; 2974 fastTrack->mBufferProvider = eabp; 2975 fastTrack->mVolumeProvider = vp; 2976 fastTrack->mSampleRate = track->mSampleRate; 2977 fastTrack->mChannelMask = track->mChannelMask; 2978 fastTrack->mGeneration++; 2979 state->mTrackMask |= 1 << j; 2980 didModify = true; 2981 // no acknowledgement required for newly active tracks 2982 } 2983 // cache the combined master volume and stream type volume for fast mixer; this 2984 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2985 track->mCachedVolume = track->isMuted() ? 2986 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2987 ++fastTracks; 2988 } else { 2989 // was it previously active? 2990 if (state->mTrackMask & (1 << j)) { 2991 fastTrack->mBufferProvider = NULL; 2992 fastTrack->mGeneration++; 2993 state->mTrackMask &= ~(1 << j); 2994 didModify = true; 2995 // If any fast tracks were removed, we must wait for acknowledgement 2996 // because we're about to decrement the last sp<> on those tracks. 2997 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2998 } else { 2999 LOG_FATAL("fast track %d should have been active", j); 3000 } 3001 tracksToRemove->add(track); 3002 // Avoids a misleading display in dumpsys 3003 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 3004 } 3005 continue; 3006 } 3007 3008 { // local variable scope to avoid goto warning 3009 3010 audio_track_cblk_t* cblk = track->cblk(); 3011 3012 // The first time a track is added we wait 3013 // for all its buffers to be filled before processing it 3014 int name = track->name(); 3015 // make sure that we have enough frames to mix one full buffer. 3016 // enforce this condition only once to enable draining the buffer in case the client 3017 // app does not call stop() and relies on underrun to stop: 3018 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 3019 // during last round 3020 uint32_t minFrames = 1; 3021 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 3022 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 3023 if (t->sampleRate() == (int)mSampleRate) { 3024 minFrames = mNormalFrameCount; 3025 } else { 3026 // +1 for rounding and +1 for additional sample needed for interpolation 3027 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 3028 // add frames already consumed but not yet released by the resampler 3029 // because cblk->framesReady() will include these frames 3030 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 3031 // the minimum track buffer size is normally twice the number of frames necessary 3032 // to fill one buffer and the resampler should not leave more than one buffer worth 3033 // of unreleased frames after each pass, but just in case... 3034 ALOG_ASSERT(minFrames <= cblk->frameCount); 3035 } 3036 } 3037 if ((track->framesReady() >= minFrames) && track->isReady() && 3038 !track->isPaused() && !track->isTerminated()) 3039 { 3040 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 3041 3042 mixedTracks++; 3043 3044 // track->mainBuffer() != mMixBuffer means there is an effect chain 3045 // connected to the track 3046 chain.clear(); 3047 if (track->mainBuffer() != mMixBuffer) { 3048 chain = getEffectChain_l(track->sessionId()); 3049 // Delegate volume control to effect in track effect chain if needed 3050 if (chain != 0) { 3051 tracksWithEffect++; 3052 } else { 3053 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3054 name, track->sessionId()); 3055 } 3056 } 3057 3058 3059 int param = AudioMixer::VOLUME; 3060 if (track->mFillingUpStatus == Track::FS_FILLED) { 3061 // no ramp for the first volume setting 3062 track->mFillingUpStatus = Track::FS_ACTIVE; 3063 if (track->mState == TrackBase::RESUMING) { 3064 track->mState = TrackBase::ACTIVE; 3065 param = AudioMixer::RAMP_VOLUME; 3066 } 3067 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3068 } else if (cblk->server != 0) { 3069 // If the track is stopped before the first frame was mixed, 3070 // do not apply ramp 3071 param = AudioMixer::RAMP_VOLUME; 3072 } 3073 3074 // compute volume for this track 3075 uint32_t vl, vr, va; 3076 if (track->isMuted() || track->isPausing() || 3077 mStreamTypes[track->streamType()].mute) { 3078 vl = vr = va = 0; 3079 if (track->isPausing()) { 3080 track->setPaused(); 3081 } 3082 } else { 3083 3084 // read original volumes with volume control 3085 float typeVolume = mStreamTypes[track->streamType()].volume; 3086 float v = masterVolume * typeVolume; 3087 uint32_t vlr = cblk->getVolumeLR(); 3088 vl = vlr & 0xFFFF; 3089 vr = vlr >> 16; 3090 // track volumes come from shared memory, so can't be trusted and must be clamped 3091 if (vl > MAX_GAIN_INT) { 3092 ALOGV("Track left volume out of range: %04X", vl); 3093 vl = MAX_GAIN_INT; 3094 } 3095 if (vr > MAX_GAIN_INT) { 3096 ALOGV("Track right volume out of range: %04X", vr); 3097 vr = MAX_GAIN_INT; 3098 } 3099 // now apply the master volume and stream type volume 3100 vl = (uint32_t)(v * vl) << 12; 3101 vr = (uint32_t)(v * vr) << 12; 3102 // assuming master volume and stream type volume each go up to 1.0, 3103 // vl and vr are now in 8.24 format 3104 3105 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3106 // send level comes from shared memory and so may be corrupt 3107 if (sendLevel > MAX_GAIN_INT) { 3108 ALOGV("Track send level out of range: %04X", sendLevel); 3109 sendLevel = MAX_GAIN_INT; 3110 } 3111 va = (uint32_t)(v * sendLevel); 3112 } 3113 // Delegate volume control to effect in track effect chain if needed 3114 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3115 // Do not ramp volume if volume is controlled by effect 3116 param = AudioMixer::VOLUME; 3117 track->mHasVolumeController = true; 3118 } else { 3119 // force no volume ramp when volume controller was just disabled or removed 3120 // from effect chain to avoid volume spike 3121 if (track->mHasVolumeController) { 3122 param = AudioMixer::VOLUME; 3123 } 3124 track->mHasVolumeController = false; 3125 } 3126 3127 // Convert volumes from 8.24 to 4.12 format 3128 // This additional clamping is needed in case chain->setVolume_l() overshot 3129 vl = (vl + (1 << 11)) >> 12; 3130 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3131 vr = (vr + (1 << 11)) >> 12; 3132 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3133 3134 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3135 3136 // XXX: these things DON'T need to be done each time 3137 mAudioMixer->setBufferProvider(name, track); 3138 mAudioMixer->enable(name); 3139 3140 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3141 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3142 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3143 mAudioMixer->setParameter( 3144 name, 3145 AudioMixer::TRACK, 3146 AudioMixer::FORMAT, (void *)track->format()); 3147 mAudioMixer->setParameter( 3148 name, 3149 AudioMixer::TRACK, 3150 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3151 mAudioMixer->setParameter( 3152 name, 3153 AudioMixer::RESAMPLE, 3154 AudioMixer::SAMPLE_RATE, 3155 (void *)(cblk->sampleRate)); 3156 mAudioMixer->setParameter( 3157 name, 3158 AudioMixer::TRACK, 3159 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3160 mAudioMixer->setParameter( 3161 name, 3162 AudioMixer::TRACK, 3163 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3164 3165 // reset retry count 3166 track->mRetryCount = kMaxTrackRetries; 3167 3168 // If one track is ready, set the mixer ready if: 3169 // - the mixer was not ready during previous round OR 3170 // - no other track is not ready 3171 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3172 mixerStatus != MIXER_TRACKS_ENABLED) { 3173 mixerStatus = MIXER_TRACKS_READY; 3174 } 3175 } else { 3176 // clear effect chain input buffer if an active track underruns to avoid sending 3177 // previous audio buffer again to effects 3178 chain = getEffectChain_l(track->sessionId()); 3179 if (chain != 0) { 3180 chain->clearInputBuffer(); 3181 } 3182 3183 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3184 if ((track->sharedBuffer() != 0) || 3185 track->isStopped() || track->isPaused()) { 3186 // We have consumed all the buffers of this track. 3187 // Remove it from the list of active tracks. 3188 // TODO: use actual buffer filling status instead of latency when available from 3189 // audio HAL 3190 size_t audioHALFrames = 3191 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3192 size_t framesWritten = 3193 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3194 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3195 if (track->isStopped()) { 3196 track->reset(); 3197 } 3198 tracksToRemove->add(track); 3199 } 3200 } else { 3201 track->mUnderrunCount++; 3202 // No buffers for this track. Give it a few chances to 3203 // fill a buffer, then remove it from active list. 3204 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3205 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3206 tracksToRemove->add(track); 3207 // indicate to client process that the track was disabled because of underrun; 3208 // it will then automatically call start() when data is available 3209 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3210 // If one track is not ready, mark the mixer also not ready if: 3211 // - the mixer was ready during previous round OR 3212 // - no other track is ready 3213 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3214 mixerStatus != MIXER_TRACKS_READY) { 3215 mixerStatus = MIXER_TRACKS_ENABLED; 3216 } 3217 } 3218 mAudioMixer->disable(name); 3219 } 3220 3221 } // local variable scope to avoid goto warning 3222track_is_ready: ; 3223 3224 } 3225 3226 // Push the new FastMixer state if necessary 3227 bool pauseAudioWatchdog = false; 3228 if (didModify) { 3229 state->mFastTracksGen++; 3230 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3231 if (kUseFastMixer == FastMixer_Dynamic && 3232 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3233 state->mCommand = FastMixerState::COLD_IDLE; 3234 state->mColdFutexAddr = &mFastMixerFutex; 3235 state->mColdGen++; 3236 mFastMixerFutex = 0; 3237 if (kUseFastMixer == FastMixer_Dynamic) { 3238 mNormalSink = mOutputSink; 3239 } 3240 // If we go into cold idle, need to wait for acknowledgement 3241 // so that fast mixer stops doing I/O. 3242 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3243 pauseAudioWatchdog = true; 3244 } 3245 sq->end(); 3246 } 3247 if (sq != NULL) { 3248 sq->end(didModify); 3249 sq->push(block); 3250 } 3251 if (pauseAudioWatchdog && mAudioWatchdog != 0) { 3252 mAudioWatchdog->pause(); 3253 } 3254 3255 // Now perform the deferred reset on fast tracks that have stopped 3256 while (resetMask != 0) { 3257 size_t i = __builtin_ctz(resetMask); 3258 ALOG_ASSERT(i < count); 3259 resetMask &= ~(1 << i); 3260 sp<Track> t = mActiveTracks[i].promote(); 3261 if (t == 0) continue; 3262 Track* track = t.get(); 3263 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3264 track->reset(); 3265 } 3266 3267 // remove all the tracks that need to be... 3268 count = tracksToRemove->size(); 3269 if (CC_UNLIKELY(count)) { 3270 for (size_t i=0 ; i<count ; i++) { 3271 const sp<Track>& track = tracksToRemove->itemAt(i); 3272 mActiveTracks.remove(track); 3273 if (track->mainBuffer() != mMixBuffer) { 3274 chain = getEffectChain_l(track->sessionId()); 3275 if (chain != 0) { 3276 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3277 chain->decActiveTrackCnt(); 3278 } 3279 } 3280 if (track->isTerminated()) { 3281 removeTrack_l(track); 3282 } 3283 } 3284 } 3285 3286 // mix buffer must be cleared if all tracks are connected to an 3287 // effect chain as in this case the mixer will not write to 3288 // mix buffer and track effects will accumulate into it 3289 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3290 // FIXME as a performance optimization, should remember previous zero status 3291 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3292 } 3293 3294 // if any fast tracks, then status is ready 3295 mMixerStatusIgnoringFastTracks = mixerStatus; 3296 if (fastTracks > 0) { 3297 mixerStatus = MIXER_TRACKS_READY; 3298 } 3299 return mixerStatus; 3300} 3301 3302/* 3303The derived values that are cached: 3304 - mixBufferSize from frame count * frame size 3305 - activeSleepTime from activeSleepTimeUs() 3306 - idleSleepTime from idleSleepTimeUs() 3307 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3308 - maxPeriod from frame count and sample rate (MIXER only) 3309 3310The parameters that affect these derived values are: 3311 - frame count 3312 - frame size 3313 - sample rate 3314 - device type: A2DP or not 3315 - device latency 3316 - format: PCM or not 3317 - active sleep time 3318 - idle sleep time 3319*/ 3320 3321void AudioFlinger::PlaybackThread::cacheParameters_l() 3322{ 3323 mixBufferSize = mNormalFrameCount * mFrameSize; 3324 activeSleepTime = activeSleepTimeUs(); 3325 idleSleepTime = idleSleepTimeUs(); 3326} 3327 3328void AudioFlinger::PlaybackThread::invalidateTracks(audio_stream_type_t streamType) 3329{ 3330 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3331 this, streamType, mTracks.size()); 3332 Mutex::Autolock _l(mLock); 3333 3334 size_t size = mTracks.size(); 3335 for (size_t i = 0; i < size; i++) { 3336 sp<Track> t = mTracks[i]; 3337 if (t->streamType() == streamType) { 3338 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3339 t->mCblk->cv.signal(); 3340 } 3341 } 3342} 3343 3344// getTrackName_l() must be called with ThreadBase::mLock held 3345int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3346{ 3347 return mAudioMixer->getTrackName(channelMask); 3348} 3349 3350// deleteTrackName_l() must be called with ThreadBase::mLock held 3351void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3352{ 3353 ALOGV("remove track (%d) and delete from mixer", name); 3354 mAudioMixer->deleteTrackName(name); 3355} 3356 3357// checkForNewParameters_l() must be called with ThreadBase::mLock held 3358bool AudioFlinger::MixerThread::checkForNewParameters_l() 3359{ 3360 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3361 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3362 bool reconfig = false; 3363 3364 while (!mNewParameters.isEmpty()) { 3365 3366 if (mFastMixer != NULL) { 3367 FastMixerStateQueue *sq = mFastMixer->sq(); 3368 FastMixerState *state = sq->begin(); 3369 if (!(state->mCommand & FastMixerState::IDLE)) { 3370 previousCommand = state->mCommand; 3371 state->mCommand = FastMixerState::HOT_IDLE; 3372 sq->end(); 3373 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3374 } else { 3375 sq->end(false /*didModify*/); 3376 } 3377 } 3378 3379 status_t status = NO_ERROR; 3380 String8 keyValuePair = mNewParameters[0]; 3381 AudioParameter param = AudioParameter(keyValuePair); 3382 int value; 3383 3384 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3385 reconfig = true; 3386 } 3387 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3388 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3389 status = BAD_VALUE; 3390 } else { 3391 reconfig = true; 3392 } 3393 } 3394 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3395 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3396 status = BAD_VALUE; 3397 } else { 3398 reconfig = true; 3399 } 3400 } 3401 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3402 // do not accept frame count changes if tracks are open as the track buffer 3403 // size depends on frame count and correct behavior would not be guaranteed 3404 // if frame count is changed after track creation 3405 if (!mTracks.isEmpty()) { 3406 status = INVALID_OPERATION; 3407 } else { 3408 reconfig = true; 3409 } 3410 } 3411 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3412#ifdef ADD_BATTERY_DATA 3413 // when changing the audio output device, call addBatteryData to notify 3414 // the change 3415 if (mDevice != value) { 3416 uint32_t params = 0; 3417 // check whether speaker is on 3418 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3419 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3420 } 3421 3422 audio_devices_t deviceWithoutSpeaker 3423 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3424 // check if any other device (except speaker) is on 3425 if (value & deviceWithoutSpeaker ) { 3426 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3427 } 3428 3429 if (params != 0) { 3430 addBatteryData(params); 3431 } 3432 } 3433#endif 3434 3435 // forward device change to effects that have requested to be 3436 // aware of attached audio device. 3437 mDevice = value; 3438 for (size_t i = 0; i < mEffectChains.size(); i++) { 3439 mEffectChains[i]->setDevice_l(mDevice); 3440 } 3441 } 3442 3443 if (status == NO_ERROR) { 3444 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3445 keyValuePair.string()); 3446 if (!mStandby && status == INVALID_OPERATION) { 3447 mOutput->stream->common.standby(&mOutput->stream->common); 3448 mStandby = true; 3449 mBytesWritten = 0; 3450 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3451 keyValuePair.string()); 3452 } 3453 if (status == NO_ERROR && reconfig) { 3454 delete mAudioMixer; 3455 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3456 mAudioMixer = NULL; 3457 readOutputParameters(); 3458 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3459 for (size_t i = 0; i < mTracks.size() ; i++) { 3460 int name = getTrackName_l(mTracks[i]->mChannelMask); 3461 if (name < 0) break; 3462 mTracks[i]->mName = name; 3463 // limit track sample rate to 2 x new output sample rate 3464 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3465 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3466 } 3467 } 3468 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3469 } 3470 } 3471 3472 mNewParameters.removeAt(0); 3473 3474 mParamStatus = status; 3475 mParamCond.signal(); 3476 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3477 // already timed out waiting for the status and will never signal the condition. 3478 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3479 } 3480 3481 if (!(previousCommand & FastMixerState::IDLE)) { 3482 ALOG_ASSERT(mFastMixer != NULL); 3483 FastMixerStateQueue *sq = mFastMixer->sq(); 3484 FastMixerState *state = sq->begin(); 3485 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3486 state->mCommand = previousCommand; 3487 sq->end(); 3488 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3489 } 3490 3491 return reconfig; 3492} 3493 3494void AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3495{ 3496 const size_t SIZE = 256; 3497 char buffer[SIZE]; 3498 String8 result; 3499 3500 PlaybackThread::dumpInternals(fd, args); 3501 3502 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3503 result.append(buffer); 3504 write(fd, result.string(), result.size()); 3505 3506 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3507 FastMixerDumpState copy = mFastMixerDumpState; 3508 copy.dump(fd); 3509 3510#ifdef STATE_QUEUE_DUMP 3511 // Similar for state queue 3512 StateQueueObserverDump observerCopy = mStateQueueObserverDump; 3513 observerCopy.dump(fd); 3514 StateQueueMutatorDump mutatorCopy = mStateQueueMutatorDump; 3515 mutatorCopy.dump(fd); 3516#endif 3517 3518 // Write the tee output to a .wav file 3519 NBAIO_Source *teeSource = mTeeSource.get(); 3520 if (teeSource != NULL) { 3521 char teePath[64]; 3522 struct timeval tv; 3523 gettimeofday(&tv, NULL); 3524 struct tm tm; 3525 localtime_r(&tv.tv_sec, &tm); 3526 strftime(teePath, sizeof(teePath), "/data/misc/media/%T.wav", &tm); 3527 int teeFd = open(teePath, O_WRONLY | O_CREAT, S_IRUSR | S_IWUSR); 3528 if (teeFd >= 0) { 3529 char wavHeader[44]; 3530 memcpy(wavHeader, 3531 "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", 3532 sizeof(wavHeader)); 3533 NBAIO_Format format = teeSource->format(); 3534 unsigned channelCount = Format_channelCount(format); 3535 ALOG_ASSERT(channelCount <= FCC_2); 3536 unsigned sampleRate = Format_sampleRate(format); 3537 wavHeader[22] = channelCount; // number of channels 3538 wavHeader[24] = sampleRate; // sample rate 3539 wavHeader[25] = sampleRate >> 8; 3540 wavHeader[32] = channelCount * 2; // block alignment 3541 write(teeFd, wavHeader, sizeof(wavHeader)); 3542 size_t total = 0; 3543 bool firstRead = true; 3544 for (;;) { 3545#define TEE_SINK_READ 1024 3546 short buffer[TEE_SINK_READ * FCC_2]; 3547 size_t count = TEE_SINK_READ; 3548 ssize_t actual = teeSource->read(buffer, count); 3549 bool wasFirstRead = firstRead; 3550 firstRead = false; 3551 if (actual <= 0) { 3552 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3553 continue; 3554 } 3555 break; 3556 } 3557 ALOG_ASSERT(actual <= (ssize_t)count); 3558 write(teeFd, buffer, actual * channelCount * sizeof(short)); 3559 total += actual; 3560 } 3561 lseek(teeFd, (off_t) 4, SEEK_SET); 3562 uint32_t temp = 44 + total * channelCount * sizeof(short) - 8; 3563 write(teeFd, &temp, sizeof(temp)); 3564 lseek(teeFd, (off_t) 40, SEEK_SET); 3565 temp = total * channelCount * sizeof(short); 3566 write(teeFd, &temp, sizeof(temp)); 3567 close(teeFd); 3568 fdprintf(fd, "FastMixer tee copied to %s\n", teePath); 3569 } else { 3570 fdprintf(fd, "FastMixer unable to create tee %s: \n", strerror(errno)); 3571 } 3572 } 3573 3574 if (mAudioWatchdog != 0) { 3575 // Make a non-atomic copy of audio watchdog dump so it won't change underneath us 3576 AudioWatchdogDump wdCopy = mAudioWatchdogDump; 3577 wdCopy.dump(fd); 3578 } 3579} 3580 3581uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3582{ 3583 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3584} 3585 3586uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3587{ 3588 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3589} 3590 3591void AudioFlinger::MixerThread::cacheParameters_l() 3592{ 3593 PlaybackThread::cacheParameters_l(); 3594 3595 // FIXME: Relaxed timing because of a certain device that can't meet latency 3596 // Should be reduced to 2x after the vendor fixes the driver issue 3597 // increase threshold again due to low power audio mode. The way this warning 3598 // threshold is calculated and its usefulness should be reconsidered anyway. 3599 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3600} 3601 3602// ---------------------------------------------------------------------------- 3603AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3604 AudioStreamOut* output, audio_io_handle_t id, audio_devices_t device) 3605 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3606 // mLeftVolFloat, mRightVolFloat 3607{ 3608} 3609 3610AudioFlinger::DirectOutputThread::~DirectOutputThread() 3611{ 3612} 3613 3614AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3615 Vector< sp<Track> > *tracksToRemove 3616) 3617{ 3618 sp<Track> trackToRemove; 3619 3620 mixer_state mixerStatus = MIXER_IDLE; 3621 3622 // find out which tracks need to be processed 3623 if (mActiveTracks.size() != 0) { 3624 sp<Track> t = mActiveTracks[0].promote(); 3625 // The track died recently 3626 if (t == 0) return MIXER_IDLE; 3627 3628 Track* const track = t.get(); 3629 audio_track_cblk_t* cblk = track->cblk(); 3630 3631 // The first time a track is added we wait 3632 // for all its buffers to be filled before processing it 3633 uint32_t minFrames; 3634 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing()) { 3635 minFrames = mNormalFrameCount; 3636 } else { 3637 minFrames = 1; 3638 } 3639 if ((track->framesReady() >= minFrames) && track->isReady() && 3640 !track->isPaused() && !track->isTerminated()) 3641 { 3642 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3643 3644 if (track->mFillingUpStatus == Track::FS_FILLED) { 3645 track->mFillingUpStatus = Track::FS_ACTIVE; 3646 mLeftVolFloat = mRightVolFloat = 0; 3647 if (track->mState == TrackBase::RESUMING) { 3648 track->mState = TrackBase::ACTIVE; 3649 } 3650 } 3651 3652 // compute volume for this track 3653 float left, right; 3654 if (track->isMuted() || mMasterMute || track->isPausing() || 3655 mStreamTypes[track->streamType()].mute) { 3656 left = right = 0; 3657 if (track->isPausing()) { 3658 track->setPaused(); 3659 } 3660 } else { 3661 float typeVolume = mStreamTypes[track->streamType()].volume; 3662 float v = mMasterVolume * typeVolume; 3663 uint32_t vlr = cblk->getVolumeLR(); 3664 float v_clamped = v * (vlr & 0xFFFF); 3665 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3666 left = v_clamped/MAX_GAIN; 3667 v_clamped = v * (vlr >> 16); 3668 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3669 right = v_clamped/MAX_GAIN; 3670 } 3671 3672 if (left != mLeftVolFloat || right != mRightVolFloat) { 3673 mLeftVolFloat = left; 3674 mRightVolFloat = right; 3675 3676 // Convert volumes from float to 8.24 3677 uint32_t vl = (uint32_t)(left * (1 << 24)); 3678 uint32_t vr = (uint32_t)(right * (1 << 24)); 3679 3680 // Delegate volume control to effect in track effect chain if needed 3681 // only one effect chain can be present on DirectOutputThread, so if 3682 // there is one, the track is connected to it 3683 if (!mEffectChains.isEmpty()) { 3684 // Do not ramp volume if volume is controlled by effect 3685 mEffectChains[0]->setVolume_l(&vl, &vr); 3686 left = (float)vl / (1 << 24); 3687 right = (float)vr / (1 << 24); 3688 } 3689 mOutput->stream->set_volume(mOutput->stream, left, right); 3690 } 3691 3692 // reset retry count 3693 track->mRetryCount = kMaxTrackRetriesDirect; 3694 mActiveTrack = t; 3695 mixerStatus = MIXER_TRACKS_READY; 3696 } else { 3697 // clear effect chain input buffer if an active track underruns to avoid sending 3698 // previous audio buffer again to effects 3699 if (!mEffectChains.isEmpty()) { 3700 mEffectChains[0]->clearInputBuffer(); 3701 } 3702 3703 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3704 if ((track->sharedBuffer() != 0) || 3705 track->isStopped() || track->isPaused()) { 3706 // We have consumed all the buffers of this track. 3707 // Remove it from the list of active tracks. 3708 // TODO: implement behavior for compressed audio 3709 size_t audioHALFrames = 3710 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3711 size_t framesWritten = 3712 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3713 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3714 if (track->isStopped()) { 3715 track->reset(); 3716 } 3717 trackToRemove = track; 3718 } 3719 } else { 3720 // No buffers for this track. Give it a few chances to 3721 // fill a buffer, then remove it from active list. 3722 if (--(track->mRetryCount) <= 0 || track->isTerminated()) { 3723 ALOGV_IF(track->mRetryCount <= 0, "BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3724 trackToRemove = track; 3725 } else { 3726 mixerStatus = MIXER_TRACKS_ENABLED; 3727 } 3728 } 3729 } 3730 } 3731 3732 // FIXME merge this with similar code for removing multiple tracks 3733 // remove all the tracks that need to be... 3734 if (CC_UNLIKELY(trackToRemove != 0)) { 3735 tracksToRemove->add(trackToRemove); 3736 mActiveTracks.remove(trackToRemove); 3737 if (!mEffectChains.isEmpty()) { 3738 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3739 trackToRemove->sessionId()); 3740 mEffectChains[0]->decActiveTrackCnt(); 3741 } 3742 if (trackToRemove->isTerminated()) { 3743 removeTrack_l(trackToRemove); 3744 } 3745 } 3746 3747 return mixerStatus; 3748} 3749 3750void AudioFlinger::DirectOutputThread::threadLoop_mix() 3751{ 3752 AudioBufferProvider::Buffer buffer; 3753 size_t frameCount = mFrameCount; 3754 int8_t *curBuf = (int8_t *)mMixBuffer; 3755 // output audio to hardware 3756 while (frameCount) { 3757 buffer.frameCount = frameCount; 3758 mActiveTrack->getNextBuffer(&buffer); 3759 if (CC_UNLIKELY(buffer.raw == NULL)) { 3760 memset(curBuf, 0, frameCount * mFrameSize); 3761 break; 3762 } 3763 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3764 frameCount -= buffer.frameCount; 3765 curBuf += buffer.frameCount * mFrameSize; 3766 mActiveTrack->releaseBuffer(&buffer); 3767 } 3768 sleepTime = 0; 3769 standbyTime = systemTime() + standbyDelay; 3770 mActiveTrack.clear(); 3771 3772} 3773 3774void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3775{ 3776 if (sleepTime == 0) { 3777 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3778 sleepTime = activeSleepTime; 3779 } else { 3780 sleepTime = idleSleepTime; 3781 } 3782 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3783 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3784 sleepTime = 0; 3785 } 3786} 3787 3788// getTrackName_l() must be called with ThreadBase::mLock held 3789int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3790{ 3791 return 0; 3792} 3793 3794// deleteTrackName_l() must be called with ThreadBase::mLock held 3795void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3796{ 3797} 3798 3799// checkForNewParameters_l() must be called with ThreadBase::mLock held 3800bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3801{ 3802 bool reconfig = false; 3803 3804 while (!mNewParameters.isEmpty()) { 3805 status_t status = NO_ERROR; 3806 String8 keyValuePair = mNewParameters[0]; 3807 AudioParameter param = AudioParameter(keyValuePair); 3808 int value; 3809 3810 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3811 // do not accept frame count changes if tracks are open as the track buffer 3812 // size depends on frame count and correct behavior would not be garantied 3813 // if frame count is changed after track creation 3814 if (!mTracks.isEmpty()) { 3815 status = INVALID_OPERATION; 3816 } else { 3817 reconfig = true; 3818 } 3819 } 3820 if (status == NO_ERROR) { 3821 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3822 keyValuePair.string()); 3823 if (!mStandby && status == INVALID_OPERATION) { 3824 mOutput->stream->common.standby(&mOutput->stream->common); 3825 mStandby = true; 3826 mBytesWritten = 0; 3827 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3828 keyValuePair.string()); 3829 } 3830 if (status == NO_ERROR && reconfig) { 3831 readOutputParameters(); 3832 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3833 } 3834 } 3835 3836 mNewParameters.removeAt(0); 3837 3838 mParamStatus = status; 3839 mParamCond.signal(); 3840 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3841 // already timed out waiting for the status and will never signal the condition. 3842 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3843 } 3844 return reconfig; 3845} 3846 3847uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3848{ 3849 uint32_t time; 3850 if (audio_is_linear_pcm(mFormat)) { 3851 time = PlaybackThread::activeSleepTimeUs(); 3852 } else { 3853 time = 10000; 3854 } 3855 return time; 3856} 3857 3858uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3859{ 3860 uint32_t time; 3861 if (audio_is_linear_pcm(mFormat)) { 3862 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3863 } else { 3864 time = 10000; 3865 } 3866 return time; 3867} 3868 3869uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3870{ 3871 uint32_t time; 3872 if (audio_is_linear_pcm(mFormat)) { 3873 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3874 } else { 3875 time = 10000; 3876 } 3877 return time; 3878} 3879 3880void AudioFlinger::DirectOutputThread::cacheParameters_l() 3881{ 3882 PlaybackThread::cacheParameters_l(); 3883 3884 // use shorter standby delay as on normal output to release 3885 // hardware resources as soon as possible 3886 standbyDelay = microseconds(activeSleepTime*2); 3887} 3888 3889// ---------------------------------------------------------------------------- 3890 3891AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3892 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3893 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3894 mWaitTimeMs(UINT_MAX) 3895{ 3896 addOutputTrack(mainThread); 3897} 3898 3899AudioFlinger::DuplicatingThread::~DuplicatingThread() 3900{ 3901 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3902 mOutputTracks[i]->destroy(); 3903 } 3904} 3905 3906void AudioFlinger::DuplicatingThread::threadLoop_mix() 3907{ 3908 // mix buffers... 3909 if (outputsReady(outputTracks)) { 3910 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3911 } else { 3912 memset(mMixBuffer, 0, mixBufferSize); 3913 } 3914 sleepTime = 0; 3915 writeFrames = mNormalFrameCount; 3916 standbyTime = systemTime() + standbyDelay; 3917} 3918 3919void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3920{ 3921 if (sleepTime == 0) { 3922 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3923 sleepTime = activeSleepTime; 3924 } else { 3925 sleepTime = idleSleepTime; 3926 } 3927 } else if (mBytesWritten != 0) { 3928 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3929 writeFrames = mNormalFrameCount; 3930 memset(mMixBuffer, 0, mixBufferSize); 3931 } else { 3932 // flush remaining overflow buffers in output tracks 3933 writeFrames = 0; 3934 } 3935 sleepTime = 0; 3936 } 3937} 3938 3939void AudioFlinger::DuplicatingThread::threadLoop_write() 3940{ 3941 for (size_t i = 0; i < outputTracks.size(); i++) { 3942 outputTracks[i]->write(mMixBuffer, writeFrames); 3943 } 3944 mBytesWritten += mixBufferSize; 3945} 3946 3947void AudioFlinger::DuplicatingThread::threadLoop_standby() 3948{ 3949 // DuplicatingThread implements standby by stopping all tracks 3950 for (size_t i = 0; i < outputTracks.size(); i++) { 3951 outputTracks[i]->stop(); 3952 } 3953} 3954 3955void AudioFlinger::DuplicatingThread::saveOutputTracks() 3956{ 3957 outputTracks = mOutputTracks; 3958} 3959 3960void AudioFlinger::DuplicatingThread::clearOutputTracks() 3961{ 3962 outputTracks.clear(); 3963} 3964 3965void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3966{ 3967 Mutex::Autolock _l(mLock); 3968 // FIXME explain this formula 3969 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3970 OutputTrack *outputTrack = new OutputTrack(thread, 3971 this, 3972 mSampleRate, 3973 mFormat, 3974 mChannelMask, 3975 frameCount); 3976 if (outputTrack->cblk() != NULL) { 3977 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3978 mOutputTracks.add(outputTrack); 3979 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3980 updateWaitTime_l(); 3981 } 3982} 3983 3984void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3985{ 3986 Mutex::Autolock _l(mLock); 3987 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3988 if (mOutputTracks[i]->thread() == thread) { 3989 mOutputTracks[i]->destroy(); 3990 mOutputTracks.removeAt(i); 3991 updateWaitTime_l(); 3992 return; 3993 } 3994 } 3995 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3996} 3997 3998// caller must hold mLock 3999void AudioFlinger::DuplicatingThread::updateWaitTime_l() 4000{ 4001 mWaitTimeMs = UINT_MAX; 4002 for (size_t i = 0; i < mOutputTracks.size(); i++) { 4003 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 4004 if (strong != 0) { 4005 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 4006 if (waitTimeMs < mWaitTimeMs) { 4007 mWaitTimeMs = waitTimeMs; 4008 } 4009 } 4010 } 4011} 4012 4013 4014bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 4015{ 4016 for (size_t i = 0; i < outputTracks.size(); i++) { 4017 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 4018 if (thread == 0) { 4019 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 4020 return false; 4021 } 4022 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4023 // see note at standby() declaration 4024 if (playbackThread->standby() && !playbackThread->isSuspended()) { 4025 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 4026 return false; 4027 } 4028 } 4029 return true; 4030} 4031 4032uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 4033{ 4034 return (mWaitTimeMs * 1000) / 2; 4035} 4036 4037void AudioFlinger::DuplicatingThread::cacheParameters_l() 4038{ 4039 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 4040 updateWaitTime_l(); 4041 4042 MixerThread::cacheParameters_l(); 4043} 4044 4045// ---------------------------------------------------------------------------- 4046 4047// TrackBase constructor must be called with AudioFlinger::mLock held 4048AudioFlinger::ThreadBase::TrackBase::TrackBase( 4049 ThreadBase *thread, 4050 const sp<Client>& client, 4051 uint32_t sampleRate, 4052 audio_format_t format, 4053 audio_channel_mask_t channelMask, 4054 int frameCount, 4055 const sp<IMemory>& sharedBuffer, 4056 int sessionId) 4057 : RefBase(), 4058 mThread(thread), 4059 mClient(client), 4060 mCblk(NULL), 4061 // mBuffer 4062 // mBufferEnd 4063 mFrameCount(0), 4064 mState(IDLE), 4065 mSampleRate(sampleRate), 4066 mFormat(format), 4067 mStepServerFailed(false), 4068 mSessionId(sessionId) 4069 // mChannelCount 4070 // mChannelMask 4071{ 4072 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4073 4074 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4075 size_t size = sizeof(audio_track_cblk_t); 4076 uint8_t channelCount = popcount(channelMask); 4077 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4078 if (sharedBuffer == 0) { 4079 size += bufferSize; 4080 } 4081 4082 if (client != NULL) { 4083 mCblkMemory = client->heap()->allocate(size); 4084 if (mCblkMemory != 0) { 4085 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4086 if (mCblk != NULL) { // construct the shared structure in-place. 4087 new(mCblk) audio_track_cblk_t(); 4088 // clear all buffers 4089 mCblk->frameCount = frameCount; 4090 mCblk->sampleRate = sampleRate; 4091// uncomment the following lines to quickly test 32-bit wraparound 4092// mCblk->user = 0xffff0000; 4093// mCblk->server = 0xffff0000; 4094// mCblk->userBase = 0xffff0000; 4095// mCblk->serverBase = 0xffff0000; 4096 mChannelCount = channelCount; 4097 mChannelMask = channelMask; 4098 if (sharedBuffer == 0) { 4099 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4100 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4101 // Force underrun condition to avoid false underrun callback until first data is 4102 // written to buffer (other flags are cleared) 4103 mCblk->flags = CBLK_UNDERRUN_ON; 4104 } else { 4105 mBuffer = sharedBuffer->pointer(); 4106 } 4107 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4108 } 4109 } else { 4110 ALOGE("not enough memory for AudioTrack size=%u", size); 4111 client->heap()->dump("AudioTrack"); 4112 return; 4113 } 4114 } else { 4115 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4116 // construct the shared structure in-place. 4117 new(mCblk) audio_track_cblk_t(); 4118 // clear all buffers 4119 mCblk->frameCount = frameCount; 4120 mCblk->sampleRate = sampleRate; 4121// uncomment the following lines to quickly test 32-bit wraparound 4122// mCblk->user = 0xffff0000; 4123// mCblk->server = 0xffff0000; 4124// mCblk->userBase = 0xffff0000; 4125// mCblk->serverBase = 0xffff0000; 4126 mChannelCount = channelCount; 4127 mChannelMask = channelMask; 4128 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4129 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4130 // Force underrun condition to avoid false underrun callback until first data is 4131 // written to buffer (other flags are cleared) 4132 mCblk->flags = CBLK_UNDERRUN_ON; 4133 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4134 } 4135} 4136 4137AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4138{ 4139 if (mCblk != NULL) { 4140 if (mClient == 0) { 4141 delete mCblk; 4142 } else { 4143 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4144 } 4145 } 4146 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4147 if (mClient != 0) { 4148 // Client destructor must run with AudioFlinger mutex locked 4149 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4150 // If the client's reference count drops to zero, the associated destructor 4151 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4152 // relying on the automatic clear() at end of scope. 4153 mClient.clear(); 4154 } 4155} 4156 4157// AudioBufferProvider interface 4158// getNextBuffer() = 0; 4159// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4160void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4161{ 4162 buffer->raw = NULL; 4163 mFrameCount = buffer->frameCount; 4164 // FIXME See note at getNextBuffer() 4165 (void) step(); // ignore return value of step() 4166 buffer->frameCount = 0; 4167} 4168 4169bool AudioFlinger::ThreadBase::TrackBase::step() { 4170 bool result; 4171 audio_track_cblk_t* cblk = this->cblk(); 4172 4173 result = cblk->stepServer(mFrameCount); 4174 if (!result) { 4175 ALOGV("stepServer failed acquiring cblk mutex"); 4176 mStepServerFailed = true; 4177 } 4178 return result; 4179} 4180 4181void AudioFlinger::ThreadBase::TrackBase::reset() { 4182 audio_track_cblk_t* cblk = this->cblk(); 4183 4184 cblk->user = 0; 4185 cblk->server = 0; 4186 cblk->userBase = 0; 4187 cblk->serverBase = 0; 4188 mStepServerFailed = false; 4189 ALOGV("TrackBase::reset"); 4190} 4191 4192int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4193 return (int)mCblk->sampleRate; 4194} 4195 4196void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4197 audio_track_cblk_t* cblk = this->cblk(); 4198 size_t frameSize = cblk->frameSize; 4199 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4200 int8_t *bufferEnd = bufferStart + frames * frameSize; 4201 4202 // Check validity of returned pointer in case the track control block would have been corrupted. 4203 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4204 "TrackBase::getBuffer buffer out of range:\n" 4205 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4206 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4207 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4208 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4209 4210 return bufferStart; 4211} 4212 4213status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4214{ 4215 mSyncEvents.add(event); 4216 return NO_ERROR; 4217} 4218 4219// ---------------------------------------------------------------------------- 4220 4221// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4222AudioFlinger::PlaybackThread::Track::Track( 4223 PlaybackThread *thread, 4224 const sp<Client>& client, 4225 audio_stream_type_t streamType, 4226 uint32_t sampleRate, 4227 audio_format_t format, 4228 audio_channel_mask_t channelMask, 4229 int frameCount, 4230 const sp<IMemory>& sharedBuffer, 4231 int sessionId, 4232 IAudioFlinger::track_flags_t flags) 4233 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4234 mMute(false), 4235 mFillingUpStatus(FS_INVALID), 4236 // mRetryCount initialized later when needed 4237 mSharedBuffer(sharedBuffer), 4238 mStreamType(streamType), 4239 mName(-1), // see note below 4240 mMainBuffer(thread->mixBuffer()), 4241 mAuxBuffer(NULL), 4242 mAuxEffectId(0), mHasVolumeController(false), 4243 mPresentationCompleteFrames(0), 4244 mFlags(flags), 4245 mFastIndex(-1), 4246 mUnderrunCount(0), 4247 mCachedVolume(1.0) 4248{ 4249 if (mCblk != NULL) { 4250 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4251 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4252 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4253 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4254 mName = thread->getTrackName_l(channelMask); 4255 mCblk->mName = mName; 4256 if (mName < 0) { 4257 ALOGE("no more track names available"); 4258 return; 4259 } 4260 // only allocate a fast track index if we were able to allocate a normal track name 4261 if (flags & IAudioFlinger::TRACK_FAST) { 4262 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4263 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4264 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4265 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4266 // FIXME This is too eager. We allocate a fast track index before the 4267 // fast track becomes active. Since fast tracks are a scarce resource, 4268 // this means we are potentially denying other more important fast tracks from 4269 // being created. It would be better to allocate the index dynamically. 4270 mFastIndex = i; 4271 mCblk->mName = i; 4272 // Read the initial underruns because this field is never cleared by the fast mixer 4273 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4274 thread->mFastTrackAvailMask &= ~(1 << i); 4275 } 4276 } 4277 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4278} 4279 4280AudioFlinger::PlaybackThread::Track::~Track() 4281{ 4282 ALOGV("PlaybackThread::Track destructor"); 4283} 4284 4285void AudioFlinger::PlaybackThread::Track::destroy() 4286{ 4287 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4288 // by removing it from mTracks vector, so there is a risk that this Tracks's 4289 // destructor is called. As the destructor needs to lock mLock, 4290 // we must acquire a strong reference on this Track before locking mLock 4291 // here so that the destructor is called only when exiting this function. 4292 // On the other hand, as long as Track::destroy() is only called by 4293 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4294 // this Track with its member mTrack. 4295 sp<Track> keep(this); 4296 { // scope for mLock 4297 sp<ThreadBase> thread = mThread.promote(); 4298 if (thread != 0) { 4299 if (!isOutputTrack()) { 4300 if (mState == ACTIVE || mState == RESUMING) { 4301 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4302 4303#ifdef ADD_BATTERY_DATA 4304 // to track the speaker usage 4305 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4306#endif 4307 } 4308 AudioSystem::releaseOutput(thread->id()); 4309 } 4310 Mutex::Autolock _l(thread->mLock); 4311 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4312 playbackThread->destroyTrack_l(this); 4313 } 4314 } 4315} 4316 4317/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4318{ 4319 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4320 " Server User Main buf Aux Buf Flags Underruns\n"); 4321} 4322 4323void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4324{ 4325 uint32_t vlr = mCblk->getVolumeLR(); 4326 if (isFastTrack()) { 4327 sprintf(buffer, " F %2d", mFastIndex); 4328 } else { 4329 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4330 } 4331 track_state state = mState; 4332 char stateChar; 4333 switch (state) { 4334 case IDLE: 4335 stateChar = 'I'; 4336 break; 4337 case TERMINATED: 4338 stateChar = 'T'; 4339 break; 4340 case STOPPING_1: 4341 stateChar = 's'; 4342 break; 4343 case STOPPING_2: 4344 stateChar = '5'; 4345 break; 4346 case STOPPED: 4347 stateChar = 'S'; 4348 break; 4349 case RESUMING: 4350 stateChar = 'R'; 4351 break; 4352 case ACTIVE: 4353 stateChar = 'A'; 4354 break; 4355 case PAUSING: 4356 stateChar = 'p'; 4357 break; 4358 case PAUSED: 4359 stateChar = 'P'; 4360 break; 4361 case FLUSHED: 4362 stateChar = 'F'; 4363 break; 4364 default: 4365 stateChar = '?'; 4366 break; 4367 } 4368 char nowInUnderrun; 4369 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4370 case UNDERRUN_FULL: 4371 nowInUnderrun = ' '; 4372 break; 4373 case UNDERRUN_PARTIAL: 4374 nowInUnderrun = '<'; 4375 break; 4376 case UNDERRUN_EMPTY: 4377 nowInUnderrun = '*'; 4378 break; 4379 default: 4380 nowInUnderrun = '?'; 4381 break; 4382 } 4383 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4384 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4385 (mClient == 0) ? getpid_cached : mClient->pid(), 4386 mStreamType, 4387 mFormat, 4388 mChannelMask, 4389 mSessionId, 4390 mFrameCount, 4391 mCblk->frameCount, 4392 stateChar, 4393 mMute, 4394 mFillingUpStatus, 4395 mCblk->sampleRate, 4396 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4397 20.0 * log10((vlr >> 16) / 4096.0), 4398 mCblk->server, 4399 mCblk->user, 4400 (int)mMainBuffer, 4401 (int)mAuxBuffer, 4402 mCblk->flags, 4403 mUnderrunCount, 4404 nowInUnderrun); 4405} 4406 4407// AudioBufferProvider interface 4408status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4409 AudioBufferProvider::Buffer* buffer, int64_t pts) 4410{ 4411 audio_track_cblk_t* cblk = this->cblk(); 4412 uint32_t framesReady; 4413 uint32_t framesReq = buffer->frameCount; 4414 4415 // Check if last stepServer failed, try to step now 4416 if (mStepServerFailed) { 4417 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4418 // Since the fast mixer is higher priority than client callback thread, 4419 // it does not result in priority inversion for client. 4420 // But a non-blocking solution would be preferable to avoid 4421 // fast mixer being unable to tryLock(), and 4422 // to avoid the extra context switches if the client wakes up, 4423 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4424 if (!step()) goto getNextBuffer_exit; 4425 ALOGV("stepServer recovered"); 4426 mStepServerFailed = false; 4427 } 4428 4429 // FIXME Same as above 4430 framesReady = cblk->framesReady(); 4431 4432 if (CC_LIKELY(framesReady)) { 4433 uint32_t s = cblk->server; 4434 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4435 4436 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4437 if (framesReq > framesReady) { 4438 framesReq = framesReady; 4439 } 4440 if (framesReq > bufferEnd - s) { 4441 framesReq = bufferEnd - s; 4442 } 4443 4444 buffer->raw = getBuffer(s, framesReq); 4445 buffer->frameCount = framesReq; 4446 return NO_ERROR; 4447 } 4448 4449getNextBuffer_exit: 4450 buffer->raw = NULL; 4451 buffer->frameCount = 0; 4452 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4453 return NOT_ENOUGH_DATA; 4454} 4455 4456// Note that framesReady() takes a mutex on the control block using tryLock(). 4457// This could result in priority inversion if framesReady() is called by the normal mixer, 4458// as the normal mixer thread runs at lower 4459// priority than the client's callback thread: there is a short window within framesReady() 4460// during which the normal mixer could be preempted, and the client callback would block. 4461// Another problem can occur if framesReady() is called by the fast mixer: 4462// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4463// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4464size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4465 return mCblk->framesReady(); 4466} 4467 4468// Don't call for fast tracks; the framesReady() could result in priority inversion 4469bool AudioFlinger::PlaybackThread::Track::isReady() const { 4470 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4471 4472 if (framesReady() >= mCblk->frameCount || 4473 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4474 mFillingUpStatus = FS_FILLED; 4475 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4476 return true; 4477 } 4478 return false; 4479} 4480 4481status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4482 int triggerSession) 4483{ 4484 status_t status = NO_ERROR; 4485 ALOGV("start(%d), calling pid %d session %d", 4486 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4487 4488 sp<ThreadBase> thread = mThread.promote(); 4489 if (thread != 0) { 4490 Mutex::Autolock _l(thread->mLock); 4491 track_state state = mState; 4492 // here the track could be either new, or restarted 4493 // in both cases "unstop" the track 4494 if (mState == PAUSED) { 4495 mState = TrackBase::RESUMING; 4496 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4497 } else { 4498 mState = TrackBase::ACTIVE; 4499 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4500 } 4501 4502 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4503 thread->mLock.unlock(); 4504 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4505 thread->mLock.lock(); 4506 4507#ifdef ADD_BATTERY_DATA 4508 // to track the speaker usage 4509 if (status == NO_ERROR) { 4510 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4511 } 4512#endif 4513 } 4514 if (status == NO_ERROR) { 4515 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4516 playbackThread->addTrack_l(this); 4517 } else { 4518 mState = state; 4519 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4520 } 4521 } else { 4522 status = BAD_VALUE; 4523 } 4524 return status; 4525} 4526 4527void AudioFlinger::PlaybackThread::Track::stop() 4528{ 4529 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4530 sp<ThreadBase> thread = mThread.promote(); 4531 if (thread != 0) { 4532 Mutex::Autolock _l(thread->mLock); 4533 track_state state = mState; 4534 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4535 // If the track is not active (PAUSED and buffers full), flush buffers 4536 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4537 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4538 reset(); 4539 mState = STOPPED; 4540 } else if (!isFastTrack()) { 4541 mState = STOPPED; 4542 } else { 4543 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4544 // and then to STOPPED and reset() when presentation is complete 4545 mState = STOPPING_1; 4546 } 4547 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4548 } 4549 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4550 thread->mLock.unlock(); 4551 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4552 thread->mLock.lock(); 4553 4554#ifdef ADD_BATTERY_DATA 4555 // to track the speaker usage 4556 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4557#endif 4558 } 4559 } 4560} 4561 4562void AudioFlinger::PlaybackThread::Track::pause() 4563{ 4564 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4565 sp<ThreadBase> thread = mThread.promote(); 4566 if (thread != 0) { 4567 Mutex::Autolock _l(thread->mLock); 4568 if (mState == ACTIVE || mState == RESUMING) { 4569 mState = PAUSING; 4570 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4571 if (!isOutputTrack()) { 4572 thread->mLock.unlock(); 4573 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4574 thread->mLock.lock(); 4575 4576#ifdef ADD_BATTERY_DATA 4577 // to track the speaker usage 4578 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4579#endif 4580 } 4581 } 4582 } 4583} 4584 4585void AudioFlinger::PlaybackThread::Track::flush() 4586{ 4587 ALOGV("flush(%d)", mName); 4588 sp<ThreadBase> thread = mThread.promote(); 4589 if (thread != 0) { 4590 Mutex::Autolock _l(thread->mLock); 4591 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4592 mState != PAUSING) { 4593 return; 4594 } 4595 // No point remaining in PAUSED state after a flush => go to 4596 // FLUSHED state 4597 mState = FLUSHED; 4598 // do not reset the track if it is still in the process of being stopped or paused. 4599 // this will be done by prepareTracks_l() when the track is stopped. 4600 // prepareTracks_l() will see mState == FLUSHED, then 4601 // remove from active track list, reset(), and trigger presentation complete 4602 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4603 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4604 reset(); 4605 } 4606 } 4607} 4608 4609void AudioFlinger::PlaybackThread::Track::reset() 4610{ 4611 // Do not reset twice to avoid discarding data written just after a flush and before 4612 // the audioflinger thread detects the track is stopped. 4613 if (!mResetDone) { 4614 TrackBase::reset(); 4615 // Force underrun condition to avoid false underrun callback until first data is 4616 // written to buffer 4617 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4618 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4619 mFillingUpStatus = FS_FILLING; 4620 mResetDone = true; 4621 if (mState == FLUSHED) { 4622 mState = IDLE; 4623 } 4624 } 4625} 4626 4627void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4628{ 4629 mMute = muted; 4630} 4631 4632status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4633{ 4634 status_t status = DEAD_OBJECT; 4635 sp<ThreadBase> thread = mThread.promote(); 4636 if (thread != 0) { 4637 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4638 sp<AudioFlinger> af = mClient->audioFlinger(); 4639 4640 Mutex::Autolock _l(af->mLock); 4641 4642 sp<PlaybackThread> srcThread = af->getEffectThread_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 4643 4644 if (EffectId != 0 && srcThread != 0 && playbackThread != srcThread.get()) { 4645 Mutex::Autolock _dl(playbackThread->mLock); 4646 Mutex::Autolock _sl(srcThread->mLock); 4647 sp<EffectChain> chain = srcThread->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 4648 if (chain == 0) { 4649 return INVALID_OPERATION; 4650 } 4651 4652 sp<EffectModule> effect = chain->getEffectFromId_l(EffectId); 4653 if (effect == 0) { 4654 return INVALID_OPERATION; 4655 } 4656 srcThread->removeEffect_l(effect); 4657 playbackThread->addEffect_l(effect); 4658 // removeEffect_l() has stopped the effect if it was active so it must be restarted 4659 if (effect->state() == EffectModule::ACTIVE || 4660 effect->state() == EffectModule::STOPPING) { 4661 effect->start(); 4662 } 4663 4664 sp<EffectChain> dstChain = effect->chain().promote(); 4665 if (dstChain == 0) { 4666 srcThread->addEffect_l(effect); 4667 return INVALID_OPERATION; 4668 } 4669 AudioSystem::unregisterEffect(effect->id()); 4670 AudioSystem::registerEffect(&effect->desc(), 4671 srcThread->id(), 4672 dstChain->strategy(), 4673 AUDIO_SESSION_OUTPUT_MIX, 4674 effect->id()); 4675 } 4676 status = playbackThread->attachAuxEffect(this, EffectId); 4677 } 4678 return status; 4679} 4680 4681void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4682{ 4683 mAuxEffectId = EffectId; 4684 mAuxBuffer = buffer; 4685} 4686 4687bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4688 size_t audioHalFrames) 4689{ 4690 // a track is considered presented when the total number of frames written to audio HAL 4691 // corresponds to the number of frames written when presentationComplete() is called for the 4692 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4693 if (mPresentationCompleteFrames == 0) { 4694 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4695 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4696 mPresentationCompleteFrames, audioHalFrames); 4697 } 4698 if (framesWritten >= mPresentationCompleteFrames) { 4699 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4700 mSessionId, framesWritten); 4701 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4702 return true; 4703 } 4704 return false; 4705} 4706 4707void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4708{ 4709 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4710 if (mSyncEvents[i]->type() == type) { 4711 mSyncEvents[i]->trigger(); 4712 mSyncEvents.removeAt(i); 4713 i--; 4714 } 4715 } 4716} 4717 4718// implement VolumeBufferProvider interface 4719 4720uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4721{ 4722 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4723 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4724 uint32_t vlr = mCblk->getVolumeLR(); 4725 uint32_t vl = vlr & 0xFFFF; 4726 uint32_t vr = vlr >> 16; 4727 // track volumes come from shared memory, so can't be trusted and must be clamped 4728 if (vl > MAX_GAIN_INT) { 4729 vl = MAX_GAIN_INT; 4730 } 4731 if (vr > MAX_GAIN_INT) { 4732 vr = MAX_GAIN_INT; 4733 } 4734 // now apply the cached master volume and stream type volume; 4735 // this is trusted but lacks any synchronization or barrier so may be stale 4736 float v = mCachedVolume; 4737 vl *= v; 4738 vr *= v; 4739 // re-combine into U4.16 4740 vlr = (vr << 16) | (vl & 0xFFFF); 4741 // FIXME look at mute, pause, and stop flags 4742 return vlr; 4743} 4744 4745status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4746{ 4747 if (mState == TERMINATED || mState == PAUSED || 4748 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4749 (mState == STOPPED)))) { 4750 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4751 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4752 event->cancel(); 4753 return INVALID_OPERATION; 4754 } 4755 TrackBase::setSyncEvent(event); 4756 return NO_ERROR; 4757} 4758 4759// timed audio tracks 4760 4761sp<AudioFlinger::PlaybackThread::TimedTrack> 4762AudioFlinger::PlaybackThread::TimedTrack::create( 4763 PlaybackThread *thread, 4764 const sp<Client>& client, 4765 audio_stream_type_t streamType, 4766 uint32_t sampleRate, 4767 audio_format_t format, 4768 audio_channel_mask_t channelMask, 4769 int frameCount, 4770 const sp<IMemory>& sharedBuffer, 4771 int sessionId) { 4772 if (!client->reserveTimedTrack()) 4773 return 0; 4774 4775 return new TimedTrack( 4776 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4777 sharedBuffer, sessionId); 4778} 4779 4780AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4781 PlaybackThread *thread, 4782 const sp<Client>& client, 4783 audio_stream_type_t streamType, 4784 uint32_t sampleRate, 4785 audio_format_t format, 4786 audio_channel_mask_t channelMask, 4787 int frameCount, 4788 const sp<IMemory>& sharedBuffer, 4789 int sessionId) 4790 : Track(thread, client, streamType, sampleRate, format, channelMask, 4791 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4792 mQueueHeadInFlight(false), 4793 mTrimQueueHeadOnRelease(false), 4794 mFramesPendingInQueue(0), 4795 mTimedSilenceBuffer(NULL), 4796 mTimedSilenceBufferSize(0), 4797 mTimedAudioOutputOnTime(false), 4798 mMediaTimeTransformValid(false) 4799{ 4800 LocalClock lc; 4801 mLocalTimeFreq = lc.getLocalFreq(); 4802 4803 mLocalTimeToSampleTransform.a_zero = 0; 4804 mLocalTimeToSampleTransform.b_zero = 0; 4805 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4806 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4807 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4808 &mLocalTimeToSampleTransform.a_to_b_denom); 4809 4810 mMediaTimeToSampleTransform.a_zero = 0; 4811 mMediaTimeToSampleTransform.b_zero = 0; 4812 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4813 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4814 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4815 &mMediaTimeToSampleTransform.a_to_b_denom); 4816} 4817 4818AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4819 mClient->releaseTimedTrack(); 4820 delete [] mTimedSilenceBuffer; 4821} 4822 4823status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4824 size_t size, sp<IMemory>* buffer) { 4825 4826 Mutex::Autolock _l(mTimedBufferQueueLock); 4827 4828 trimTimedBufferQueue_l(); 4829 4830 // lazily initialize the shared memory heap for timed buffers 4831 if (mTimedMemoryDealer == NULL) { 4832 const int kTimedBufferHeapSize = 512 << 10; 4833 4834 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4835 "AudioFlingerTimed"); 4836 if (mTimedMemoryDealer == NULL) 4837 return NO_MEMORY; 4838 } 4839 4840 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4841 if (newBuffer == NULL) { 4842 newBuffer = mTimedMemoryDealer->allocate(size); 4843 if (newBuffer == NULL) 4844 return NO_MEMORY; 4845 } 4846 4847 *buffer = newBuffer; 4848 return NO_ERROR; 4849} 4850 4851// caller must hold mTimedBufferQueueLock 4852void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4853 int64_t mediaTimeNow; 4854 { 4855 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4856 if (!mMediaTimeTransformValid) 4857 return; 4858 4859 int64_t targetTimeNow; 4860 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4861 ? mCCHelper.getCommonTime(&targetTimeNow) 4862 : mCCHelper.getLocalTime(&targetTimeNow); 4863 4864 if (OK != res) 4865 return; 4866 4867 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4868 &mediaTimeNow)) { 4869 return; 4870 } 4871 } 4872 4873 size_t trimEnd; 4874 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4875 int64_t bufEnd; 4876 4877 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4878 // We have a next buffer. Just use its PTS as the PTS of the frame 4879 // following the last frame in this buffer. If the stream is sparse 4880 // (ie, there are deliberate gaps left in the stream which should be 4881 // filled with silence by the TimedAudioTrack), then this can result 4882 // in one extra buffer being left un-trimmed when it could have 4883 // been. In general, this is not typical, and we would rather 4884 // optimized away the TS calculation below for the more common case 4885 // where PTSes are contiguous. 4886 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4887 } else { 4888 // We have no next buffer. Compute the PTS of the frame following 4889 // the last frame in this buffer by computing the duration of of 4890 // this frame in media time units and adding it to the PTS of the 4891 // buffer. 4892 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4893 / mCblk->frameSize; 4894 4895 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4896 &bufEnd)) { 4897 ALOGE("Failed to convert frame count of %lld to media time" 4898 " duration" " (scale factor %d/%u) in %s", 4899 frameCount, 4900 mMediaTimeToSampleTransform.a_to_b_numer, 4901 mMediaTimeToSampleTransform.a_to_b_denom, 4902 __PRETTY_FUNCTION__); 4903 break; 4904 } 4905 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4906 } 4907 4908 if (bufEnd > mediaTimeNow) 4909 break; 4910 4911 // Is the buffer we want to use in the middle of a mix operation right 4912 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4913 // from the mixer which should be coming back shortly. 4914 if (!trimEnd && mQueueHeadInFlight) { 4915 mTrimQueueHeadOnRelease = true; 4916 } 4917 } 4918 4919 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4920 if (trimStart < trimEnd) { 4921 // Update the bookkeeping for framesReady() 4922 for (size_t i = trimStart; i < trimEnd; ++i) { 4923 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4924 } 4925 4926 // Now actually remove the buffers from the queue. 4927 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4928 } 4929} 4930 4931void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4932 const char* logTag) { 4933 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4934 "%s called (reason \"%s\"), but timed buffer queue has no" 4935 " elements to trim.", __FUNCTION__, logTag); 4936 4937 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4938 mTimedBufferQueue.removeAt(0); 4939} 4940 4941void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4942 const TimedBuffer& buf, 4943 const char* logTag) { 4944 uint32_t bufBytes = buf.buffer()->size(); 4945 uint32_t consumedAlready = buf.position(); 4946 4947 ALOG_ASSERT(consumedAlready <= bufBytes, 4948 "Bad bookkeeping while updating frames pending. Timed buffer is" 4949 " only %u bytes long, but claims to have consumed %u" 4950 " bytes. (update reason: \"%s\")", 4951 bufBytes, consumedAlready, logTag); 4952 4953 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4954 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4955 "Bad bookkeeping while updating frames pending. Should have at" 4956 " least %u queued frames, but we think we have only %u. (update" 4957 " reason: \"%s\")", 4958 bufFrames, mFramesPendingInQueue, logTag); 4959 4960 mFramesPendingInQueue -= bufFrames; 4961} 4962 4963status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4964 const sp<IMemory>& buffer, int64_t pts) { 4965 4966 { 4967 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4968 if (!mMediaTimeTransformValid) 4969 return INVALID_OPERATION; 4970 } 4971 4972 Mutex::Autolock _l(mTimedBufferQueueLock); 4973 4974 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4975 mFramesPendingInQueue += bufFrames; 4976 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4977 4978 return NO_ERROR; 4979} 4980 4981status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4982 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4983 4984 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4985 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4986 target); 4987 4988 if (!(target == TimedAudioTrack::LOCAL_TIME || 4989 target == TimedAudioTrack::COMMON_TIME)) { 4990 return BAD_VALUE; 4991 } 4992 4993 Mutex::Autolock lock(mMediaTimeTransformLock); 4994 mMediaTimeTransform = xform; 4995 mMediaTimeTransformTarget = target; 4996 mMediaTimeTransformValid = true; 4997 4998 return NO_ERROR; 4999} 5000 5001#define min(a, b) ((a) < (b) ? (a) : (b)) 5002 5003// implementation of getNextBuffer for tracks whose buffers have timestamps 5004status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 5005 AudioBufferProvider::Buffer* buffer, int64_t pts) 5006{ 5007 if (pts == AudioBufferProvider::kInvalidPTS) { 5008 buffer->raw = NULL; 5009 buffer->frameCount = 0; 5010 mTimedAudioOutputOnTime = false; 5011 return INVALID_OPERATION; 5012 } 5013 5014 Mutex::Autolock _l(mTimedBufferQueueLock); 5015 5016 ALOG_ASSERT(!mQueueHeadInFlight, 5017 "getNextBuffer called without releaseBuffer!"); 5018 5019 while (true) { 5020 5021 // if we have no timed buffers, then fail 5022 if (mTimedBufferQueue.isEmpty()) { 5023 buffer->raw = NULL; 5024 buffer->frameCount = 0; 5025 return NOT_ENOUGH_DATA; 5026 } 5027 5028 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5029 5030 // calculate the PTS of the head of the timed buffer queue expressed in 5031 // local time 5032 int64_t headLocalPTS; 5033 { 5034 Mutex::Autolock mttLock(mMediaTimeTransformLock); 5035 5036 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 5037 5038 if (mMediaTimeTransform.a_to_b_denom == 0) { 5039 // the transform represents a pause, so yield silence 5040 timedYieldSilence_l(buffer->frameCount, buffer); 5041 return NO_ERROR; 5042 } 5043 5044 int64_t transformedPTS; 5045 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 5046 &transformedPTS)) { 5047 // the transform failed. this shouldn't happen, but if it does 5048 // then just drop this buffer 5049 ALOGW("timedGetNextBuffer transform failed"); 5050 buffer->raw = NULL; 5051 buffer->frameCount = 0; 5052 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 5053 return NO_ERROR; 5054 } 5055 5056 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 5057 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 5058 &headLocalPTS)) { 5059 buffer->raw = NULL; 5060 buffer->frameCount = 0; 5061 return INVALID_OPERATION; 5062 } 5063 } else { 5064 headLocalPTS = transformedPTS; 5065 } 5066 } 5067 5068 // adjust the head buffer's PTS to reflect the portion of the head buffer 5069 // that has already been consumed 5070 int64_t effectivePTS = headLocalPTS + 5071 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 5072 5073 // Calculate the delta in samples between the head of the input buffer 5074 // queue and the start of the next output buffer that will be written. 5075 // If the transformation fails because of over or underflow, it means 5076 // that the sample's position in the output stream is so far out of 5077 // whack that it should just be dropped. 5078 int64_t sampleDelta; 5079 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 5080 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 5081 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 5082 " mix"); 5083 continue; 5084 } 5085 if (!mLocalTimeToSampleTransform.doForwardTransform( 5086 (effectivePTS - pts) << 32, &sampleDelta)) { 5087 ALOGV("*** too late during sample rate transform: dropped buffer"); 5088 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5089 continue; 5090 } 5091 5092 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5093 " sampleDelta=[%d.%08x]", 5094 head.pts(), head.position(), pts, 5095 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5096 + (sampleDelta >> 32)), 5097 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5098 5099 // if the delta between the ideal placement for the next input sample and 5100 // the current output position is within this threshold, then we will 5101 // concatenate the next input samples to the previous output 5102 const int64_t kSampleContinuityThreshold = 5103 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5104 5105 // if this is the first buffer of audio that we're emitting from this track 5106 // then it should be almost exactly on time. 5107 const int64_t kSampleStartupThreshold = 1LL << 32; 5108 5109 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5110 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5111 // the next input is close enough to being on time, so concatenate it 5112 // with the last output 5113 timedYieldSamples_l(buffer); 5114 5115 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5116 head.position(), buffer->frameCount); 5117 return NO_ERROR; 5118 } 5119 5120 // Looks like our output is not on time. Reset our on timed status. 5121 // Next time we mix samples from our input queue, then should be within 5122 // the StartupThreshold. 5123 mTimedAudioOutputOnTime = false; 5124 if (sampleDelta > 0) { 5125 // the gap between the current output position and the proper start of 5126 // the next input sample is too big, so fill it with silence 5127 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5128 5129 timedYieldSilence_l(framesUntilNextInput, buffer); 5130 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5131 return NO_ERROR; 5132 } else { 5133 // the next input sample is late 5134 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5135 size_t onTimeSamplePosition = 5136 head.position() + lateFrames * mCblk->frameSize; 5137 5138 if (onTimeSamplePosition > head.buffer()->size()) { 5139 // all the remaining samples in the head are too late, so 5140 // drop it and move on 5141 ALOGV("*** too late: dropped buffer"); 5142 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5143 continue; 5144 } else { 5145 // skip over the late samples 5146 head.setPosition(onTimeSamplePosition); 5147 5148 // yield the available samples 5149 timedYieldSamples_l(buffer); 5150 5151 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5152 return NO_ERROR; 5153 } 5154 } 5155 } 5156} 5157 5158// Yield samples from the timed buffer queue head up to the given output 5159// buffer's capacity. 5160// 5161// Caller must hold mTimedBufferQueueLock 5162void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5163 AudioBufferProvider::Buffer* buffer) { 5164 5165 const TimedBuffer& head = mTimedBufferQueue[0]; 5166 5167 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5168 head.position()); 5169 5170 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5171 mCblk->frameSize); 5172 size_t framesRequested = buffer->frameCount; 5173 buffer->frameCount = min(framesLeftInHead, framesRequested); 5174 5175 mQueueHeadInFlight = true; 5176 mTimedAudioOutputOnTime = true; 5177} 5178 5179// Yield samples of silence up to the given output buffer's capacity 5180// 5181// Caller must hold mTimedBufferQueueLock 5182void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5183 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5184 5185 // lazily allocate a buffer filled with silence 5186 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5187 delete [] mTimedSilenceBuffer; 5188 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5189 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5190 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5191 } 5192 5193 buffer->raw = mTimedSilenceBuffer; 5194 size_t framesRequested = buffer->frameCount; 5195 buffer->frameCount = min(numFrames, framesRequested); 5196 5197 mTimedAudioOutputOnTime = false; 5198} 5199 5200// AudioBufferProvider interface 5201void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5202 AudioBufferProvider::Buffer* buffer) { 5203 5204 Mutex::Autolock _l(mTimedBufferQueueLock); 5205 5206 // If the buffer which was just released is part of the buffer at the head 5207 // of the queue, be sure to update the amt of the buffer which has been 5208 // consumed. If the buffer being returned is not part of the head of the 5209 // queue, its either because the buffer is part of the silence buffer, or 5210 // because the head of the timed queue was trimmed after the mixer called 5211 // getNextBuffer but before the mixer called releaseBuffer. 5212 if (buffer->raw == mTimedSilenceBuffer) { 5213 ALOG_ASSERT(!mQueueHeadInFlight, 5214 "Queue head in flight during release of silence buffer!"); 5215 goto done; 5216 } 5217 5218 ALOG_ASSERT(mQueueHeadInFlight, 5219 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5220 " head in flight."); 5221 5222 if (mTimedBufferQueue.size()) { 5223 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5224 5225 void* start = head.buffer()->pointer(); 5226 void* end = reinterpret_cast<void*>( 5227 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5228 + head.buffer()->size()); 5229 5230 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5231 "released buffer not within the head of the timed buffer" 5232 " queue; qHead = [%p, %p], released buffer = %p", 5233 start, end, buffer->raw); 5234 5235 head.setPosition(head.position() + 5236 (buffer->frameCount * mCblk->frameSize)); 5237 mQueueHeadInFlight = false; 5238 5239 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5240 "Bad bookkeeping during releaseBuffer! Should have at" 5241 " least %u queued frames, but we think we have only %u", 5242 buffer->frameCount, mFramesPendingInQueue); 5243 5244 mFramesPendingInQueue -= buffer->frameCount; 5245 5246 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5247 || mTrimQueueHeadOnRelease) { 5248 trimTimedBufferQueueHead_l("releaseBuffer"); 5249 mTrimQueueHeadOnRelease = false; 5250 } 5251 } else { 5252 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5253 " buffers in the timed buffer queue"); 5254 } 5255 5256done: 5257 buffer->raw = 0; 5258 buffer->frameCount = 0; 5259} 5260 5261size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5262 Mutex::Autolock _l(mTimedBufferQueueLock); 5263 return mFramesPendingInQueue; 5264} 5265 5266AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5267 : mPTS(0), mPosition(0) {} 5268 5269AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5270 const sp<IMemory>& buffer, int64_t pts) 5271 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5272 5273// ---------------------------------------------------------------------------- 5274 5275// RecordTrack constructor must be called with AudioFlinger::mLock held 5276AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5277 RecordThread *thread, 5278 const sp<Client>& client, 5279 uint32_t sampleRate, 5280 audio_format_t format, 5281 audio_channel_mask_t channelMask, 5282 int frameCount, 5283 int sessionId) 5284 : TrackBase(thread, client, sampleRate, format, 5285 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5286 mOverflow(false) 5287{ 5288 if (mCblk != NULL) { 5289 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5290 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5291 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5292 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5293 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5294 } else { 5295 mCblk->frameSize = sizeof(int8_t); 5296 } 5297 } 5298} 5299 5300AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5301{ 5302 ALOGV("%s", __func__); 5303} 5304 5305// AudioBufferProvider interface 5306status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5307{ 5308 audio_track_cblk_t* cblk = this->cblk(); 5309 uint32_t framesAvail; 5310 uint32_t framesReq = buffer->frameCount; 5311 5312 // Check if last stepServer failed, try to step now 5313 if (mStepServerFailed) { 5314 if (!step()) goto getNextBuffer_exit; 5315 ALOGV("stepServer recovered"); 5316 mStepServerFailed = false; 5317 } 5318 5319 framesAvail = cblk->framesAvailable_l(); 5320 5321 if (CC_LIKELY(framesAvail)) { 5322 uint32_t s = cblk->server; 5323 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5324 5325 if (framesReq > framesAvail) { 5326 framesReq = framesAvail; 5327 } 5328 if (framesReq > bufferEnd - s) { 5329 framesReq = bufferEnd - s; 5330 } 5331 5332 buffer->raw = getBuffer(s, framesReq); 5333 buffer->frameCount = framesReq; 5334 return NO_ERROR; 5335 } 5336 5337getNextBuffer_exit: 5338 buffer->raw = NULL; 5339 buffer->frameCount = 0; 5340 return NOT_ENOUGH_DATA; 5341} 5342 5343status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5344 int triggerSession) 5345{ 5346 sp<ThreadBase> thread = mThread.promote(); 5347 if (thread != 0) { 5348 RecordThread *recordThread = (RecordThread *)thread.get(); 5349 return recordThread->start(this, event, triggerSession); 5350 } else { 5351 return BAD_VALUE; 5352 } 5353} 5354 5355void AudioFlinger::RecordThread::RecordTrack::stop() 5356{ 5357 sp<ThreadBase> thread = mThread.promote(); 5358 if (thread != 0) { 5359 RecordThread *recordThread = (RecordThread *)thread.get(); 5360 recordThread->mLock.lock(); 5361 bool doStop = recordThread->stop_l(this); 5362 if (doStop) { 5363 TrackBase::reset(); 5364 // Force overrun condition to avoid false overrun callback until first data is 5365 // read from buffer 5366 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5367 } 5368 recordThread->mLock.unlock(); 5369 if (doStop) { 5370 AudioSystem::stopInput(recordThread->id()); 5371 } 5372 } 5373} 5374 5375/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result) 5376{ 5377 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5378} 5379 5380void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5381{ 5382 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5383 (mClient == 0) ? getpid_cached : mClient->pid(), 5384 mFormat, 5385 mChannelMask, 5386 mSessionId, 5387 mFrameCount, 5388 mState, 5389 mCblk->sampleRate, 5390 mCblk->server, 5391 mCblk->user); 5392} 5393 5394 5395// ---------------------------------------------------------------------------- 5396 5397AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5398 PlaybackThread *playbackThread, 5399 DuplicatingThread *sourceThread, 5400 uint32_t sampleRate, 5401 audio_format_t format, 5402 audio_channel_mask_t channelMask, 5403 int frameCount) 5404 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5405 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5406 mActive(false), mSourceThread(sourceThread) 5407{ 5408 5409 if (mCblk != NULL) { 5410 mCblk->flags |= CBLK_DIRECTION_OUT; 5411 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5412 mOutBuffer.frameCount = 0; 5413 playbackThread->mTracks.add(this); 5414 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5415 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5416 mCblk, mBuffer, mCblk->buffers, 5417 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5418 } else { 5419 ALOGW("Error creating output track on thread %p", playbackThread); 5420 } 5421} 5422 5423AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5424{ 5425 clearBufferQueue(); 5426} 5427 5428status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5429 int triggerSession) 5430{ 5431 status_t status = Track::start(event, triggerSession); 5432 if (status != NO_ERROR) { 5433 return status; 5434 } 5435 5436 mActive = true; 5437 mRetryCount = 127; 5438 return status; 5439} 5440 5441void AudioFlinger::PlaybackThread::OutputTrack::stop() 5442{ 5443 Track::stop(); 5444 clearBufferQueue(); 5445 mOutBuffer.frameCount = 0; 5446 mActive = false; 5447} 5448 5449bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5450{ 5451 Buffer *pInBuffer; 5452 Buffer inBuffer; 5453 uint32_t channelCount = mChannelCount; 5454 bool outputBufferFull = false; 5455 inBuffer.frameCount = frames; 5456 inBuffer.i16 = data; 5457 5458 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5459 5460 if (!mActive && frames != 0) { 5461 start(); 5462 sp<ThreadBase> thread = mThread.promote(); 5463 if (thread != 0) { 5464 MixerThread *mixerThread = (MixerThread *)thread.get(); 5465 if (mCblk->frameCount > frames){ 5466 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5467 uint32_t startFrames = (mCblk->frameCount - frames); 5468 pInBuffer = new Buffer; 5469 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5470 pInBuffer->frameCount = startFrames; 5471 pInBuffer->i16 = pInBuffer->mBuffer; 5472 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5473 mBufferQueue.add(pInBuffer); 5474 } else { 5475 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5476 } 5477 } 5478 } 5479 } 5480 5481 while (waitTimeLeftMs) { 5482 // First write pending buffers, then new data 5483 if (mBufferQueue.size()) { 5484 pInBuffer = mBufferQueue.itemAt(0); 5485 } else { 5486 pInBuffer = &inBuffer; 5487 } 5488 5489 if (pInBuffer->frameCount == 0) { 5490 break; 5491 } 5492 5493 if (mOutBuffer.frameCount == 0) { 5494 mOutBuffer.frameCount = pInBuffer->frameCount; 5495 nsecs_t startTime = systemTime(); 5496 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5497 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5498 outputBufferFull = true; 5499 break; 5500 } 5501 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5502 if (waitTimeLeftMs >= waitTimeMs) { 5503 waitTimeLeftMs -= waitTimeMs; 5504 } else { 5505 waitTimeLeftMs = 0; 5506 } 5507 } 5508 5509 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5510 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5511 mCblk->stepUser(outFrames); 5512 pInBuffer->frameCount -= outFrames; 5513 pInBuffer->i16 += outFrames * channelCount; 5514 mOutBuffer.frameCount -= outFrames; 5515 mOutBuffer.i16 += outFrames * channelCount; 5516 5517 if (pInBuffer->frameCount == 0) { 5518 if (mBufferQueue.size()) { 5519 mBufferQueue.removeAt(0); 5520 delete [] pInBuffer->mBuffer; 5521 delete pInBuffer; 5522 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5523 } else { 5524 break; 5525 } 5526 } 5527 } 5528 5529 // If we could not write all frames, allocate a buffer and queue it for next time. 5530 if (inBuffer.frameCount) { 5531 sp<ThreadBase> thread = mThread.promote(); 5532 if (thread != 0 && !thread->standby()) { 5533 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5534 pInBuffer = new Buffer; 5535 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5536 pInBuffer->frameCount = inBuffer.frameCount; 5537 pInBuffer->i16 = pInBuffer->mBuffer; 5538 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5539 mBufferQueue.add(pInBuffer); 5540 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5541 } else { 5542 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5543 } 5544 } 5545 } 5546 5547 // Calling write() with a 0 length buffer, means that no more data will be written: 5548 // If no more buffers are pending, fill output track buffer to make sure it is started 5549 // by output mixer. 5550 if (frames == 0 && mBufferQueue.size() == 0) { 5551 if (mCblk->user < mCblk->frameCount) { 5552 frames = mCblk->frameCount - mCblk->user; 5553 pInBuffer = new Buffer; 5554 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5555 pInBuffer->frameCount = frames; 5556 pInBuffer->i16 = pInBuffer->mBuffer; 5557 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5558 mBufferQueue.add(pInBuffer); 5559 } else if (mActive) { 5560 stop(); 5561 } 5562 } 5563 5564 return outputBufferFull; 5565} 5566 5567status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5568{ 5569 int active; 5570 status_t result; 5571 audio_track_cblk_t* cblk = mCblk; 5572 uint32_t framesReq = buffer->frameCount; 5573 5574// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5575 buffer->frameCount = 0; 5576 5577 uint32_t framesAvail = cblk->framesAvailable(); 5578 5579 5580 if (framesAvail == 0) { 5581 Mutex::Autolock _l(cblk->lock); 5582 goto start_loop_here; 5583 while (framesAvail == 0) { 5584 active = mActive; 5585 if (CC_UNLIKELY(!active)) { 5586 ALOGV("Not active and NO_MORE_BUFFERS"); 5587 return NO_MORE_BUFFERS; 5588 } 5589 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5590 if (result != NO_ERROR) { 5591 return NO_MORE_BUFFERS; 5592 } 5593 // read the server count again 5594 start_loop_here: 5595 framesAvail = cblk->framesAvailable_l(); 5596 } 5597 } 5598 5599// if (framesAvail < framesReq) { 5600// return NO_MORE_BUFFERS; 5601// } 5602 5603 if (framesReq > framesAvail) { 5604 framesReq = framesAvail; 5605 } 5606 5607 uint32_t u = cblk->user; 5608 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5609 5610 if (framesReq > bufferEnd - u) { 5611 framesReq = bufferEnd - u; 5612 } 5613 5614 buffer->frameCount = framesReq; 5615 buffer->raw = (void *)cblk->buffer(u); 5616 return NO_ERROR; 5617} 5618 5619 5620void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5621{ 5622 size_t size = mBufferQueue.size(); 5623 5624 for (size_t i = 0; i < size; i++) { 5625 Buffer *pBuffer = mBufferQueue.itemAt(i); 5626 delete [] pBuffer->mBuffer; 5627 delete pBuffer; 5628 } 5629 mBufferQueue.clear(); 5630} 5631 5632// ---------------------------------------------------------------------------- 5633 5634AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5635 : RefBase(), 5636 mAudioFlinger(audioFlinger), 5637 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5638 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5639 mPid(pid), 5640 mTimedTrackCount(0) 5641{ 5642 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5643} 5644 5645// Client destructor must be called with AudioFlinger::mLock held 5646AudioFlinger::Client::~Client() 5647{ 5648 mAudioFlinger->removeClient_l(mPid); 5649} 5650 5651sp<MemoryDealer> AudioFlinger::Client::heap() const 5652{ 5653 return mMemoryDealer; 5654} 5655 5656// Reserve one of the limited slots for a timed audio track associated 5657// with this client 5658bool AudioFlinger::Client::reserveTimedTrack() 5659{ 5660 const int kMaxTimedTracksPerClient = 4; 5661 5662 Mutex::Autolock _l(mTimedTrackLock); 5663 5664 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5665 ALOGW("can not create timed track - pid %d has exceeded the limit", 5666 mPid); 5667 return false; 5668 } 5669 5670 mTimedTrackCount++; 5671 return true; 5672} 5673 5674// Release a slot for a timed audio track 5675void AudioFlinger::Client::releaseTimedTrack() 5676{ 5677 Mutex::Autolock _l(mTimedTrackLock); 5678 mTimedTrackCount--; 5679} 5680 5681// ---------------------------------------------------------------------------- 5682 5683AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5684 const sp<IAudioFlingerClient>& client, 5685 pid_t pid) 5686 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5687{ 5688} 5689 5690AudioFlinger::NotificationClient::~NotificationClient() 5691{ 5692} 5693 5694void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5695{ 5696 sp<NotificationClient> keep(this); 5697 mAudioFlinger->removeNotificationClient(mPid); 5698} 5699 5700// ---------------------------------------------------------------------------- 5701 5702AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5703 : BnAudioTrack(), 5704 mTrack(track) 5705{ 5706} 5707 5708AudioFlinger::TrackHandle::~TrackHandle() { 5709 // just stop the track on deletion, associated resources 5710 // will be freed from the main thread once all pending buffers have 5711 // been played. Unless it's not in the active track list, in which 5712 // case we free everything now... 5713 mTrack->destroy(); 5714} 5715 5716sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5717 return mTrack->getCblk(); 5718} 5719 5720status_t AudioFlinger::TrackHandle::start() { 5721 return mTrack->start(); 5722} 5723 5724void AudioFlinger::TrackHandle::stop() { 5725 mTrack->stop(); 5726} 5727 5728void AudioFlinger::TrackHandle::flush() { 5729 mTrack->flush(); 5730} 5731 5732void AudioFlinger::TrackHandle::mute(bool e) { 5733 mTrack->mute(e); 5734} 5735 5736void AudioFlinger::TrackHandle::pause() { 5737 mTrack->pause(); 5738} 5739 5740status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5741{ 5742 return mTrack->attachAuxEffect(EffectId); 5743} 5744 5745status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5746 sp<IMemory>* buffer) { 5747 if (!mTrack->isTimedTrack()) 5748 return INVALID_OPERATION; 5749 5750 PlaybackThread::TimedTrack* tt = 5751 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5752 return tt->allocateTimedBuffer(size, buffer); 5753} 5754 5755status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5756 int64_t pts) { 5757 if (!mTrack->isTimedTrack()) 5758 return INVALID_OPERATION; 5759 5760 PlaybackThread::TimedTrack* tt = 5761 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5762 return tt->queueTimedBuffer(buffer, pts); 5763} 5764 5765status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5766 const LinearTransform& xform, int target) { 5767 5768 if (!mTrack->isTimedTrack()) 5769 return INVALID_OPERATION; 5770 5771 PlaybackThread::TimedTrack* tt = 5772 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5773 return tt->setMediaTimeTransform( 5774 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5775} 5776 5777status_t AudioFlinger::TrackHandle::onTransact( 5778 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5779{ 5780 return BnAudioTrack::onTransact(code, data, reply, flags); 5781} 5782 5783// ---------------------------------------------------------------------------- 5784 5785sp<IAudioRecord> AudioFlinger::openRecord( 5786 pid_t pid, 5787 audio_io_handle_t input, 5788 uint32_t sampleRate, 5789 audio_format_t format, 5790 audio_channel_mask_t channelMask, 5791 int frameCount, 5792 IAudioFlinger::track_flags_t flags, 5793 pid_t tid, 5794 int *sessionId, 5795 status_t *status) 5796{ 5797 sp<RecordThread::RecordTrack> recordTrack; 5798 sp<RecordHandle> recordHandle; 5799 sp<Client> client; 5800 status_t lStatus; 5801 RecordThread *thread; 5802 size_t inFrameCount; 5803 int lSessionId; 5804 5805 // check calling permissions 5806 if (!recordingAllowed()) { 5807 lStatus = PERMISSION_DENIED; 5808 goto Exit; 5809 } 5810 5811 // add client to list 5812 { // scope for mLock 5813 Mutex::Autolock _l(mLock); 5814 thread = checkRecordThread_l(input); 5815 if (thread == NULL) { 5816 lStatus = BAD_VALUE; 5817 goto Exit; 5818 } 5819 5820 client = registerPid_l(pid); 5821 5822 // If no audio session id is provided, create one here 5823 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5824 lSessionId = *sessionId; 5825 } else { 5826 lSessionId = nextUniqueId(); 5827 if (sessionId != NULL) { 5828 *sessionId = lSessionId; 5829 } 5830 } 5831 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5832 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 5833 frameCount, lSessionId, flags, tid, &lStatus); 5834 } 5835 if (lStatus != NO_ERROR) { 5836 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5837 // destructor is called by the TrackBase destructor with mLock held 5838 client.clear(); 5839 recordTrack.clear(); 5840 goto Exit; 5841 } 5842 5843 // return to handle to client 5844 recordHandle = new RecordHandle(recordTrack); 5845 lStatus = NO_ERROR; 5846 5847Exit: 5848 if (status) { 5849 *status = lStatus; 5850 } 5851 return recordHandle; 5852} 5853 5854// ---------------------------------------------------------------------------- 5855 5856AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5857 : BnAudioRecord(), 5858 mRecordTrack(recordTrack) 5859{ 5860} 5861 5862AudioFlinger::RecordHandle::~RecordHandle() { 5863 stop_nonvirtual(); 5864 mRecordTrack->destroy(); 5865} 5866 5867sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5868 return mRecordTrack->getCblk(); 5869} 5870 5871status_t AudioFlinger::RecordHandle::start(int /*AudioSystem::sync_event_t*/ event, int triggerSession) { 5872 ALOGV("RecordHandle::start()"); 5873 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5874} 5875 5876void AudioFlinger::RecordHandle::stop() { 5877 stop_nonvirtual(); 5878} 5879 5880void AudioFlinger::RecordHandle::stop_nonvirtual() { 5881 ALOGV("RecordHandle::stop()"); 5882 mRecordTrack->stop(); 5883} 5884 5885status_t AudioFlinger::RecordHandle::onTransact( 5886 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5887{ 5888 return BnAudioRecord::onTransact(code, data, reply, flags); 5889} 5890 5891// ---------------------------------------------------------------------------- 5892 5893AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5894 AudioStreamIn *input, 5895 uint32_t sampleRate, 5896 audio_channel_mask_t channelMask, 5897 audio_io_handle_t id, 5898 audio_devices_t device) : 5899 ThreadBase(audioFlinger, id, device, RECORD), 5900 mInput(input), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5901 // mRsmpInIndex and mInputBytes set by readInputParameters() 5902 mReqChannelCount(popcount(channelMask)), 5903 mReqSampleRate(sampleRate) 5904 // mBytesRead is only meaningful while active, and so is cleared in start() 5905 // (but might be better to also clear here for dump?) 5906{ 5907 snprintf(mName, kNameLength, "AudioIn_%X", id); 5908 5909 readInputParameters(); 5910} 5911 5912 5913AudioFlinger::RecordThread::~RecordThread() 5914{ 5915 delete[] mRsmpInBuffer; 5916 delete mResampler; 5917 delete[] mRsmpOutBuffer; 5918} 5919 5920void AudioFlinger::RecordThread::onFirstRef() 5921{ 5922 run(mName, PRIORITY_URGENT_AUDIO); 5923} 5924 5925status_t AudioFlinger::RecordThread::readyToRun() 5926{ 5927 status_t status = initCheck(); 5928 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5929 return status; 5930} 5931 5932bool AudioFlinger::RecordThread::threadLoop() 5933{ 5934 AudioBufferProvider::Buffer buffer; 5935 sp<RecordTrack> activeTrack; 5936 Vector< sp<EffectChain> > effectChains; 5937 5938 nsecs_t lastWarning = 0; 5939 5940 inputStandBy(); 5941 acquireWakeLock(); 5942 5943 // start recording 5944 while (!exitPending()) { 5945 5946 processConfigEvents(); 5947 5948 { // scope for mLock 5949 Mutex::Autolock _l(mLock); 5950 checkForNewParameters_l(); 5951 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5952 standby(); 5953 5954 if (exitPending()) break; 5955 5956 releaseWakeLock_l(); 5957 ALOGV("RecordThread: loop stopping"); 5958 // go to sleep 5959 mWaitWorkCV.wait(mLock); 5960 ALOGV("RecordThread: loop starting"); 5961 acquireWakeLock_l(); 5962 continue; 5963 } 5964 if (mActiveTrack != 0) { 5965 if (mActiveTrack->mState == TrackBase::PAUSING) { 5966 standby(); 5967 mActiveTrack.clear(); 5968 mStartStopCond.broadcast(); 5969 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5970 if (mReqChannelCount != mActiveTrack->channelCount()) { 5971 mActiveTrack.clear(); 5972 mStartStopCond.broadcast(); 5973 } else if (mBytesRead != 0) { 5974 // record start succeeds only if first read from audio input 5975 // succeeds 5976 if (mBytesRead > 0) { 5977 mActiveTrack->mState = TrackBase::ACTIVE; 5978 } else { 5979 mActiveTrack.clear(); 5980 } 5981 mStartStopCond.broadcast(); 5982 } 5983 mStandby = false; 5984 } else if (mActiveTrack->mState == TrackBase::TERMINATED) { 5985 removeTrack_l(mActiveTrack); 5986 mActiveTrack.clear(); 5987 } 5988 } 5989 lockEffectChains_l(effectChains); 5990 } 5991 5992 if (mActiveTrack != 0) { 5993 if (mActiveTrack->mState != TrackBase::ACTIVE && 5994 mActiveTrack->mState != TrackBase::RESUMING) { 5995 unlockEffectChains(effectChains); 5996 usleep(kRecordThreadSleepUs); 5997 continue; 5998 } 5999 for (size_t i = 0; i < effectChains.size(); i ++) { 6000 effectChains[i]->process_l(); 6001 } 6002 6003 buffer.frameCount = mFrameCount; 6004 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 6005 size_t framesOut = buffer.frameCount; 6006 if (mResampler == NULL) { 6007 // no resampling 6008 while (framesOut) { 6009 size_t framesIn = mFrameCount - mRsmpInIndex; 6010 if (framesIn) { 6011 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 6012 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 6013 if (framesIn > framesOut) 6014 framesIn = framesOut; 6015 mRsmpInIndex += framesIn; 6016 framesOut -= framesIn; 6017 if ((int)mChannelCount == mReqChannelCount || 6018 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 6019 memcpy(dst, src, framesIn * mFrameSize); 6020 } else { 6021 if (mChannelCount == 1) { 6022 upmix_to_stereo_i16_from_mono_i16((int16_t *)dst, 6023 (int16_t *)src, framesIn); 6024 } else { 6025 downmix_to_mono_i16_from_stereo_i16((int16_t *)dst, 6026 (int16_t *)src, framesIn); 6027 } 6028 } 6029 } 6030 if (framesOut && mFrameCount == mRsmpInIndex) { 6031 if (framesOut == mFrameCount && 6032 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 6033 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 6034 framesOut = 0; 6035 } else { 6036 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6037 mRsmpInIndex = 0; 6038 } 6039 if (mBytesRead < 0) { 6040 ALOGE("Error reading audio input"); 6041 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6042 // Force input into standby so that it tries to 6043 // recover at next read attempt 6044 inputStandBy(); 6045 usleep(kRecordThreadSleepUs); 6046 } 6047 mRsmpInIndex = mFrameCount; 6048 framesOut = 0; 6049 buffer.frameCount = 0; 6050 } 6051 } 6052 } 6053 } else { 6054 // resampling 6055 6056 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 6057 // alter output frame count as if we were expecting stereo samples 6058 if (mChannelCount == 1 && mReqChannelCount == 1) { 6059 framesOut >>= 1; 6060 } 6061 mResampler->resample(mRsmpOutBuffer, framesOut, this); 6062 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 6063 // are 32 bit aligned which should be always true. 6064 if (mChannelCount == 2 && mReqChannelCount == 1) { 6065 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 6066 // the resampler always outputs stereo samples: do post stereo to mono conversion 6067 downmix_to_mono_i16_from_stereo_i16(buffer.i16, (int16_t *)mRsmpOutBuffer, 6068 framesOut); 6069 } else { 6070 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 6071 } 6072 6073 } 6074 if (mFramestoDrop == 0) { 6075 mActiveTrack->releaseBuffer(&buffer); 6076 } else { 6077 if (mFramestoDrop > 0) { 6078 mFramestoDrop -= buffer.frameCount; 6079 if (mFramestoDrop <= 0) { 6080 clearSyncStartEvent(); 6081 } 6082 } else { 6083 mFramestoDrop += buffer.frameCount; 6084 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6085 mSyncStartEvent->isCancelled()) { 6086 ALOGW("Synced record %s, session %d, trigger session %d", 6087 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6088 mActiveTrack->sessionId(), 6089 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6090 clearSyncStartEvent(); 6091 } 6092 } 6093 } 6094 mActiveTrack->clearOverflow(); 6095 } 6096 // client isn't retrieving buffers fast enough 6097 else { 6098 if (!mActiveTrack->setOverflow()) { 6099 nsecs_t now = systemTime(); 6100 if ((now - lastWarning) > kWarningThrottleNs) { 6101 ALOGW("RecordThread: buffer overflow"); 6102 lastWarning = now; 6103 } 6104 } 6105 // Release the processor for a while before asking for a new buffer. 6106 // This will give the application more chance to read from the buffer and 6107 // clear the overflow. 6108 usleep(kRecordThreadSleepUs); 6109 } 6110 } 6111 // enable changes in effect chain 6112 unlockEffectChains(effectChains); 6113 effectChains.clear(); 6114 } 6115 6116 standby(); 6117 6118 { 6119 Mutex::Autolock _l(mLock); 6120 mActiveTrack.clear(); 6121 mStartStopCond.broadcast(); 6122 } 6123 6124 releaseWakeLock(); 6125 6126 ALOGV("RecordThread %p exiting", this); 6127 return false; 6128} 6129 6130void AudioFlinger::RecordThread::standby() 6131{ 6132 if (!mStandby) { 6133 inputStandBy(); 6134 mStandby = true; 6135 } 6136} 6137 6138void AudioFlinger::RecordThread::inputStandBy() 6139{ 6140 mInput->stream->common.standby(&mInput->stream->common); 6141} 6142 6143sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6144 const sp<AudioFlinger::Client>& client, 6145 uint32_t sampleRate, 6146 audio_format_t format, 6147 audio_channel_mask_t channelMask, 6148 int frameCount, 6149 int sessionId, 6150 IAudioFlinger::track_flags_t flags, 6151 pid_t tid, 6152 status_t *status) 6153{ 6154 sp<RecordTrack> track; 6155 status_t lStatus; 6156 6157 lStatus = initCheck(); 6158 if (lStatus != NO_ERROR) { 6159 ALOGE("Audio driver not initialized."); 6160 goto Exit; 6161 } 6162 6163 // FIXME use flags and tid similar to createTrack_l() 6164 6165 { // scope for mLock 6166 Mutex::Autolock _l(mLock); 6167 6168 track = new RecordTrack(this, client, sampleRate, 6169 format, channelMask, frameCount, sessionId); 6170 6171 if (track->getCblk() == 0) { 6172 lStatus = NO_MEMORY; 6173 goto Exit; 6174 } 6175 mTracks.add(track); 6176 6177 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6178 bool suspend = audio_is_bluetooth_sco_device(mDevice & AUDIO_DEVICE_IN_ALL) && 6179 mAudioFlinger->btNrecIsOff(); 6180 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6181 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6182 } 6183 lStatus = NO_ERROR; 6184 6185Exit: 6186 if (status) { 6187 *status = lStatus; 6188 } 6189 return track; 6190} 6191 6192status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6193 AudioSystem::sync_event_t event, 6194 int triggerSession) 6195{ 6196 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6197 sp<ThreadBase> strongMe = this; 6198 status_t status = NO_ERROR; 6199 6200 if (event == AudioSystem::SYNC_EVENT_NONE) { 6201 clearSyncStartEvent(); 6202 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6203 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6204 triggerSession, 6205 recordTrack->sessionId(), 6206 syncStartEventCallback, 6207 this); 6208 // Sync event can be cancelled by the trigger session if the track is not in a 6209 // compatible state in which case we start record immediately 6210 if (mSyncStartEvent->isCancelled()) { 6211 clearSyncStartEvent(); 6212 } else { 6213 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6214 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6215 } 6216 } 6217 6218 { 6219 AutoMutex lock(mLock); 6220 if (mActiveTrack != 0) { 6221 if (recordTrack != mActiveTrack.get()) { 6222 status = -EBUSY; 6223 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6224 mActiveTrack->mState = TrackBase::ACTIVE; 6225 } 6226 return status; 6227 } 6228 6229 recordTrack->mState = TrackBase::IDLE; 6230 mActiveTrack = recordTrack; 6231 mLock.unlock(); 6232 status_t status = AudioSystem::startInput(mId); 6233 mLock.lock(); 6234 if (status != NO_ERROR) { 6235 mActiveTrack.clear(); 6236 clearSyncStartEvent(); 6237 return status; 6238 } 6239 mRsmpInIndex = mFrameCount; 6240 mBytesRead = 0; 6241 if (mResampler != NULL) { 6242 mResampler->reset(); 6243 } 6244 mActiveTrack->mState = TrackBase::RESUMING; 6245 // signal thread to start 6246 ALOGV("Signal record thread"); 6247 mWaitWorkCV.signal(); 6248 // do not wait for mStartStopCond if exiting 6249 if (exitPending()) { 6250 mActiveTrack.clear(); 6251 status = INVALID_OPERATION; 6252 goto startError; 6253 } 6254 mStartStopCond.wait(mLock); 6255 if (mActiveTrack == 0) { 6256 ALOGV("Record failed to start"); 6257 status = BAD_VALUE; 6258 goto startError; 6259 } 6260 ALOGV("Record started OK"); 6261 return status; 6262 } 6263startError: 6264 AudioSystem::stopInput(mId); 6265 clearSyncStartEvent(); 6266 return status; 6267} 6268 6269void AudioFlinger::RecordThread::clearSyncStartEvent() 6270{ 6271 if (mSyncStartEvent != 0) { 6272 mSyncStartEvent->cancel(); 6273 } 6274 mSyncStartEvent.clear(); 6275 mFramestoDrop = 0; 6276} 6277 6278void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6279{ 6280 sp<SyncEvent> strongEvent = event.promote(); 6281 6282 if (strongEvent != 0) { 6283 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6284 me->handleSyncStartEvent(strongEvent); 6285 } 6286} 6287 6288void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6289{ 6290 if (event == mSyncStartEvent) { 6291 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6292 // from audio HAL 6293 mFramestoDrop = mFrameCount * 2; 6294 } 6295} 6296 6297bool AudioFlinger::RecordThread::stop_l(RecordThread::RecordTrack* recordTrack) { 6298 ALOGV("RecordThread::stop"); 6299 if (recordTrack != mActiveTrack.get() || recordTrack->mState == TrackBase::PAUSING) { 6300 return false; 6301 } 6302 recordTrack->mState = TrackBase::PAUSING; 6303 // do not wait for mStartStopCond if exiting 6304 if (exitPending()) { 6305 return true; 6306 } 6307 mStartStopCond.wait(mLock); 6308 // if we have been restarted, recordTrack == mActiveTrack.get() here 6309 if (exitPending() || recordTrack != mActiveTrack.get()) { 6310 ALOGV("Record stopped OK"); 6311 return true; 6312 } 6313 return false; 6314} 6315 6316bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6317{ 6318 return false; 6319} 6320 6321status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6322{ 6323 if (!isValidSyncEvent(event)) { 6324 return BAD_VALUE; 6325 } 6326 6327 int eventSession = event->triggerSession(); 6328 status_t ret = NAME_NOT_FOUND; 6329 6330 Mutex::Autolock _l(mLock); 6331 6332 for (size_t i = 0; i < mTracks.size(); i++) { 6333 sp<RecordTrack> track = mTracks[i]; 6334 if (eventSession == track->sessionId()) { 6335 track->setSyncEvent(event); 6336 ret = NO_ERROR; 6337 } 6338 } 6339 return ret; 6340} 6341 6342void AudioFlinger::RecordThread::RecordTrack::destroy() 6343{ 6344 // see comments at AudioFlinger::PlaybackThread::Track::destroy() 6345 sp<RecordTrack> keep(this); 6346 { 6347 sp<ThreadBase> thread = mThread.promote(); 6348 if (thread != 0) { 6349 if (mState == ACTIVE || mState == RESUMING) { 6350 AudioSystem::stopInput(thread->id()); 6351 } 6352 AudioSystem::releaseInput(thread->id()); 6353 Mutex::Autolock _l(thread->mLock); 6354 RecordThread *recordThread = (RecordThread *) thread.get(); 6355 recordThread->destroyTrack_l(this); 6356 } 6357 } 6358} 6359 6360// destroyTrack_l() must be called with ThreadBase::mLock held 6361void AudioFlinger::RecordThread::destroyTrack_l(const sp<RecordTrack>& track) 6362{ 6363 track->mState = TrackBase::TERMINATED; 6364 // active tracks are removed by threadLoop() 6365 if (mActiveTrack != track) { 6366 removeTrack_l(track); 6367 } 6368} 6369 6370void AudioFlinger::RecordThread::removeTrack_l(const sp<RecordTrack>& track) 6371{ 6372 mTracks.remove(track); 6373 // need anything related to effects here? 6374} 6375 6376void AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6377{ 6378 dumpInternals(fd, args); 6379 dumpTracks(fd, args); 6380 dumpEffectChains(fd, args); 6381} 6382 6383void AudioFlinger::RecordThread::dumpInternals(int fd, const Vector<String16>& args) 6384{ 6385 const size_t SIZE = 256; 6386 char buffer[SIZE]; 6387 String8 result; 6388 6389 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6390 result.append(buffer); 6391 6392 if (mActiveTrack != 0) { 6393 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6394 result.append(buffer); 6395 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6396 result.append(buffer); 6397 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6398 result.append(buffer); 6399 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6400 result.append(buffer); 6401 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6402 result.append(buffer); 6403 } else { 6404 result.append("No active record client\n"); 6405 } 6406 6407 write(fd, result.string(), result.size()); 6408 6409 dumpBase(fd, args); 6410} 6411 6412void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args) 6413{ 6414 const size_t SIZE = 256; 6415 char buffer[SIZE]; 6416 String8 result; 6417 6418 snprintf(buffer, SIZE, "Input thread %p tracks\n", this); 6419 result.append(buffer); 6420 RecordTrack::appendDumpHeader(result); 6421 for (size_t i = 0; i < mTracks.size(); ++i) { 6422 sp<RecordTrack> track = mTracks[i]; 6423 if (track != 0) { 6424 track->dump(buffer, SIZE); 6425 result.append(buffer); 6426 } 6427 } 6428 6429 if (mActiveTrack != 0) { 6430 snprintf(buffer, SIZE, "\nInput thread %p active tracks\n", this); 6431 result.append(buffer); 6432 RecordTrack::appendDumpHeader(result); 6433 mActiveTrack->dump(buffer, SIZE); 6434 result.append(buffer); 6435 6436 } 6437 write(fd, result.string(), result.size()); 6438} 6439 6440// AudioBufferProvider interface 6441status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6442{ 6443 size_t framesReq = buffer->frameCount; 6444 size_t framesReady = mFrameCount - mRsmpInIndex; 6445 int channelCount; 6446 6447 if (framesReady == 0) { 6448 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6449 if (mBytesRead < 0) { 6450 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6451 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6452 // Force input into standby so that it tries to 6453 // recover at next read attempt 6454 inputStandBy(); 6455 usleep(kRecordThreadSleepUs); 6456 } 6457 buffer->raw = NULL; 6458 buffer->frameCount = 0; 6459 return NOT_ENOUGH_DATA; 6460 } 6461 mRsmpInIndex = 0; 6462 framesReady = mFrameCount; 6463 } 6464 6465 if (framesReq > framesReady) { 6466 framesReq = framesReady; 6467 } 6468 6469 if (mChannelCount == 1 && mReqChannelCount == 2) { 6470 channelCount = 1; 6471 } else { 6472 channelCount = 2; 6473 } 6474 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6475 buffer->frameCount = framesReq; 6476 return NO_ERROR; 6477} 6478 6479// AudioBufferProvider interface 6480void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6481{ 6482 mRsmpInIndex += buffer->frameCount; 6483 buffer->frameCount = 0; 6484} 6485 6486bool AudioFlinger::RecordThread::checkForNewParameters_l() 6487{ 6488 bool reconfig = false; 6489 6490 while (!mNewParameters.isEmpty()) { 6491 status_t status = NO_ERROR; 6492 String8 keyValuePair = mNewParameters[0]; 6493 AudioParameter param = AudioParameter(keyValuePair); 6494 int value; 6495 audio_format_t reqFormat = mFormat; 6496 int reqSamplingRate = mReqSampleRate; 6497 int reqChannelCount = mReqChannelCount; 6498 6499 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6500 reqSamplingRate = value; 6501 reconfig = true; 6502 } 6503 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6504 reqFormat = (audio_format_t) value; 6505 reconfig = true; 6506 } 6507 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6508 reqChannelCount = popcount(value); 6509 reconfig = true; 6510 } 6511 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6512 // do not accept frame count changes if tracks are open as the track buffer 6513 // size depends on frame count and correct behavior would not be guaranteed 6514 // if frame count is changed after track creation 6515 if (mActiveTrack != 0) { 6516 status = INVALID_OPERATION; 6517 } else { 6518 reconfig = true; 6519 } 6520 } 6521 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6522 // forward device change to effects that have requested to be 6523 // aware of attached audio device. 6524 for (size_t i = 0; i < mEffectChains.size(); i++) { 6525 mEffectChains[i]->setDevice_l(value); 6526 } 6527 // store input device and output device but do not forward output device to audio HAL. 6528 // Note that status is ignored by the caller for output device 6529 // (see AudioFlinger::setParameters() 6530 audio_devices_t newDevice = mDevice; 6531 if (value & AUDIO_DEVICE_OUT_ALL) { 6532 newDevice &= ~(value & AUDIO_DEVICE_OUT_ALL); 6533 status = BAD_VALUE; 6534 } else { 6535 newDevice &= ~(value & AUDIO_DEVICE_IN_ALL); 6536 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6537 if (mTracks.size() > 0) { 6538 bool suspend = audio_is_bluetooth_sco_device( 6539 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6540 for (size_t i = 0; i < mTracks.size(); i++) { 6541 sp<RecordTrack> track = mTracks[i]; 6542 setEffectSuspended_l(FX_IID_AEC, suspend, track->sessionId()); 6543 setEffectSuspended_l(FX_IID_NS, suspend, track->sessionId()); 6544 } 6545 } 6546 } 6547 newDevice |= value; 6548 mDevice = newDevice; // since mDevice is read by other threads, only write to it once 6549 } 6550 if (status == NO_ERROR) { 6551 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6552 if (status == INVALID_OPERATION) { 6553 inputStandBy(); 6554 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6555 keyValuePair.string()); 6556 } 6557 if (reconfig) { 6558 if (status == BAD_VALUE && 6559 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6560 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6561 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6562 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6563 (reqChannelCount <= FCC_2)) { 6564 status = NO_ERROR; 6565 } 6566 if (status == NO_ERROR) { 6567 readInputParameters(); 6568 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6569 } 6570 } 6571 } 6572 6573 mNewParameters.removeAt(0); 6574 6575 mParamStatus = status; 6576 mParamCond.signal(); 6577 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6578 // already timed out waiting for the status and will never signal the condition. 6579 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6580 } 6581 return reconfig; 6582} 6583 6584String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6585{ 6586 char *s; 6587 String8 out_s8 = String8(); 6588 6589 Mutex::Autolock _l(mLock); 6590 if (initCheck() != NO_ERROR) { 6591 return out_s8; 6592 } 6593 6594 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6595 out_s8 = String8(s); 6596 free(s); 6597 return out_s8; 6598} 6599 6600void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6601 AudioSystem::OutputDescriptor desc; 6602 void *param2 = NULL; 6603 6604 switch (event) { 6605 case AudioSystem::INPUT_OPENED: 6606 case AudioSystem::INPUT_CONFIG_CHANGED: 6607 desc.channels = mChannelMask; 6608 desc.samplingRate = mSampleRate; 6609 desc.format = mFormat; 6610 desc.frameCount = mFrameCount; 6611 desc.latency = 0; 6612 param2 = &desc; 6613 break; 6614 6615 case AudioSystem::INPUT_CLOSED: 6616 default: 6617 break; 6618 } 6619 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6620} 6621 6622void AudioFlinger::RecordThread::readInputParameters() 6623{ 6624 delete mRsmpInBuffer; 6625 // mRsmpInBuffer is always assigned a new[] below 6626 delete mRsmpOutBuffer; 6627 mRsmpOutBuffer = NULL; 6628 delete mResampler; 6629 mResampler = NULL; 6630 6631 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6632 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6633 mChannelCount = (uint16_t)popcount(mChannelMask); 6634 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6635 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6636 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6637 mFrameCount = mInputBytes / mFrameSize; 6638 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6639 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6640 6641 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6642 { 6643 int channelCount; 6644 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6645 // stereo to mono post process as the resampler always outputs stereo. 6646 if (mChannelCount == 1 && mReqChannelCount == 2) { 6647 channelCount = 1; 6648 } else { 6649 channelCount = 2; 6650 } 6651 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6652 mResampler->setSampleRate(mSampleRate); 6653 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6654 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6655 6656 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6657 if (mChannelCount == 1 && mReqChannelCount == 1) { 6658 mFrameCount >>= 1; 6659 } 6660 6661 } 6662 mRsmpInIndex = mFrameCount; 6663} 6664 6665unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6666{ 6667 Mutex::Autolock _l(mLock); 6668 if (initCheck() != NO_ERROR) { 6669 return 0; 6670 } 6671 6672 return mInput->stream->get_input_frames_lost(mInput->stream); 6673} 6674 6675uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6676{ 6677 Mutex::Autolock _l(mLock); 6678 uint32_t result = 0; 6679 if (getEffectChain_l(sessionId) != 0) { 6680 result = EFFECT_SESSION; 6681 } 6682 6683 for (size_t i = 0; i < mTracks.size(); ++i) { 6684 if (sessionId == mTracks[i]->sessionId()) { 6685 result |= TRACK_SESSION; 6686 break; 6687 } 6688 } 6689 6690 return result; 6691} 6692 6693KeyedVector<int, bool> AudioFlinger::RecordThread::sessionIds() 6694{ 6695 KeyedVector<int, bool> ids; 6696 Mutex::Autolock _l(mLock); 6697 for (size_t j = 0; j < mTracks.size(); ++j) { 6698 sp<RecordThread::RecordTrack> track = mTracks[j]; 6699 int sessionId = track->sessionId(); 6700 if (ids.indexOfKey(sessionId) < 0) { 6701 ids.add(sessionId, true); 6702 } 6703 } 6704 return ids; 6705} 6706 6707AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6708{ 6709 Mutex::Autolock _l(mLock); 6710 AudioStreamIn *input = mInput; 6711 mInput = NULL; 6712 return input; 6713} 6714 6715// this method must always be called either with ThreadBase mLock held or inside the thread loop 6716audio_stream_t* AudioFlinger::RecordThread::stream() const 6717{ 6718 if (mInput == NULL) { 6719 return NULL; 6720 } 6721 return &mInput->stream->common; 6722} 6723 6724 6725// ---------------------------------------------------------------------------- 6726 6727audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6728{ 6729 if (!settingsAllowed()) { 6730 return 0; 6731 } 6732 Mutex::Autolock _l(mLock); 6733 return loadHwModule_l(name); 6734} 6735 6736// loadHwModule_l() must be called with AudioFlinger::mLock held 6737audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6738{ 6739 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6740 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6741 ALOGW("loadHwModule() module %s already loaded", name); 6742 return mAudioHwDevs.keyAt(i); 6743 } 6744 } 6745 6746 audio_hw_device_t *dev; 6747 6748 int rc = load_audio_interface(name, &dev); 6749 if (rc) { 6750 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6751 return 0; 6752 } 6753 6754 mHardwareStatus = AUDIO_HW_INIT; 6755 rc = dev->init_check(dev); 6756 mHardwareStatus = AUDIO_HW_IDLE; 6757 if (rc) { 6758 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6759 return 0; 6760 } 6761 6762 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6763 (NULL != dev->set_master_volume)) { 6764 AutoMutex lock(mHardwareLock); 6765 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6766 dev->set_master_volume(dev, mMasterVolume); 6767 mHardwareStatus = AUDIO_HW_IDLE; 6768 } 6769 6770 audio_module_handle_t handle = nextUniqueId(); 6771 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6772 6773 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6774 name, dev->common.module->name, dev->common.module->id, handle); 6775 6776 return handle; 6777 6778} 6779 6780audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6781 audio_devices_t *pDevices, 6782 uint32_t *pSamplingRate, 6783 audio_format_t *pFormat, 6784 audio_channel_mask_t *pChannelMask, 6785 uint32_t *pLatencyMs, 6786 audio_output_flags_t flags) 6787{ 6788 status_t status; 6789 PlaybackThread *thread = NULL; 6790 struct audio_config config = { 6791 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6792 channel_mask: pChannelMask ? *pChannelMask : 0, 6793 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6794 }; 6795 audio_stream_out_t *outStream = NULL; 6796 audio_hw_device_t *outHwDev; 6797 6798 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6799 module, 6800 (pDevices != NULL) ? *pDevices : 0, 6801 config.sample_rate, 6802 config.format, 6803 config.channel_mask, 6804 flags); 6805 6806 if (pDevices == NULL || *pDevices == 0) { 6807 return 0; 6808 } 6809 6810 Mutex::Autolock _l(mLock); 6811 6812 outHwDev = findSuitableHwDev_l(module, *pDevices); 6813 if (outHwDev == NULL) 6814 return 0; 6815 6816 audio_io_handle_t id = nextUniqueId(); 6817 6818 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6819 6820 status = outHwDev->open_output_stream(outHwDev, 6821 id, 6822 *pDevices, 6823 (audio_output_flags_t)flags, 6824 &config, 6825 &outStream); 6826 6827 mHardwareStatus = AUDIO_HW_IDLE; 6828 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6829 outStream, 6830 config.sample_rate, 6831 config.format, 6832 config.channel_mask, 6833 status); 6834 6835 if (status == NO_ERROR && outStream != NULL) { 6836 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6837 6838 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6839 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6840 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6841 thread = new DirectOutputThread(this, output, id, *pDevices); 6842 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6843 } else { 6844 thread = new MixerThread(this, output, id, *pDevices); 6845 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6846 } 6847 mPlaybackThreads.add(id, thread); 6848 6849 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6850 if (pFormat != NULL) *pFormat = config.format; 6851 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6852 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6853 6854 // notify client processes of the new output creation 6855 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6856 6857 // the first primary output opened designates the primary hw device 6858 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6859 ALOGI("Using module %d has the primary audio interface", module); 6860 mPrimaryHardwareDev = outHwDev; 6861 6862 AutoMutex lock(mHardwareLock); 6863 mHardwareStatus = AUDIO_HW_SET_MODE; 6864 outHwDev->set_mode(outHwDev, mMode); 6865 6866 // Determine the level of master volume support the primary audio HAL has, 6867 // and set the initial master volume at the same time. 6868 float initialVolume = 1.0; 6869 mMasterVolumeSupportLvl = MVS_NONE; 6870 6871 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6872 if ((NULL != outHwDev->get_master_volume) && 6873 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6874 mMasterVolumeSupportLvl = MVS_FULL; 6875 } else { 6876 mMasterVolumeSupportLvl = MVS_SETONLY; 6877 initialVolume = 1.0; 6878 } 6879 6880 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6881 if ((NULL == outHwDev->set_master_volume) || 6882 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6883 mMasterVolumeSupportLvl = MVS_NONE; 6884 } 6885 // now that we have a primary device, initialize master volume on other devices 6886 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6887 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6888 6889 if ((dev != mPrimaryHardwareDev) && 6890 (NULL != dev->set_master_volume)) { 6891 dev->set_master_volume(dev, initialVolume); 6892 } 6893 } 6894 mHardwareStatus = AUDIO_HW_IDLE; 6895 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6896 ? initialVolume 6897 : 1.0; 6898 mMasterVolume = initialVolume; 6899 } 6900 return id; 6901 } 6902 6903 return 0; 6904} 6905 6906audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6907 audio_io_handle_t output2) 6908{ 6909 Mutex::Autolock _l(mLock); 6910 MixerThread *thread1 = checkMixerThread_l(output1); 6911 MixerThread *thread2 = checkMixerThread_l(output2); 6912 6913 if (thread1 == NULL || thread2 == NULL) { 6914 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6915 return 0; 6916 } 6917 6918 audio_io_handle_t id = nextUniqueId(); 6919 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6920 thread->addOutputTrack(thread2); 6921 mPlaybackThreads.add(id, thread); 6922 // notify client processes of the new output creation 6923 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6924 return id; 6925} 6926 6927status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6928{ 6929 return closeOutput_nonvirtual(output); 6930} 6931 6932status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 6933{ 6934 // keep strong reference on the playback thread so that 6935 // it is not destroyed while exit() is executed 6936 sp<PlaybackThread> thread; 6937 { 6938 Mutex::Autolock _l(mLock); 6939 thread = checkPlaybackThread_l(output); 6940 if (thread == NULL) { 6941 return BAD_VALUE; 6942 } 6943 6944 ALOGV("closeOutput() %d", output); 6945 6946 if (thread->type() == ThreadBase::MIXER) { 6947 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6948 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6949 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6950 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6951 } 6952 } 6953 } 6954 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6955 mPlaybackThreads.removeItem(output); 6956 } 6957 thread->exit(); 6958 // The thread entity (active unit of execution) is no longer running here, 6959 // but the ThreadBase container still exists. 6960 6961 if (thread->type() != ThreadBase::DUPLICATING) { 6962 AudioStreamOut *out = thread->clearOutput(); 6963 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6964 // from now on thread->mOutput is NULL 6965 out->hwDev->close_output_stream(out->hwDev, out->stream); 6966 delete out; 6967 } 6968 return NO_ERROR; 6969} 6970 6971status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6972{ 6973 Mutex::Autolock _l(mLock); 6974 PlaybackThread *thread = checkPlaybackThread_l(output); 6975 6976 if (thread == NULL) { 6977 return BAD_VALUE; 6978 } 6979 6980 ALOGV("suspendOutput() %d", output); 6981 thread->suspend(); 6982 6983 return NO_ERROR; 6984} 6985 6986status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6987{ 6988 Mutex::Autolock _l(mLock); 6989 PlaybackThread *thread = checkPlaybackThread_l(output); 6990 6991 if (thread == NULL) { 6992 return BAD_VALUE; 6993 } 6994 6995 ALOGV("restoreOutput() %d", output); 6996 6997 thread->restore(); 6998 6999 return NO_ERROR; 7000} 7001 7002audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 7003 audio_devices_t *pDevices, 7004 uint32_t *pSamplingRate, 7005 audio_format_t *pFormat, 7006 audio_channel_mask_t *pChannelMask) 7007{ 7008 status_t status; 7009 RecordThread *thread = NULL; 7010 struct audio_config config = { 7011 sample_rate: pSamplingRate ? *pSamplingRate : 0, 7012 channel_mask: pChannelMask ? *pChannelMask : 0, 7013 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 7014 }; 7015 uint32_t reqSamplingRate = config.sample_rate; 7016 audio_format_t reqFormat = config.format; 7017 audio_channel_mask_t reqChannels = config.channel_mask; 7018 audio_stream_in_t *inStream = NULL; 7019 audio_hw_device_t *inHwDev; 7020 7021 if (pDevices == NULL || *pDevices == 0) { 7022 return 0; 7023 } 7024 7025 Mutex::Autolock _l(mLock); 7026 7027 inHwDev = findSuitableHwDev_l(module, *pDevices); 7028 if (inHwDev == NULL) 7029 return 0; 7030 7031 audio_io_handle_t id = nextUniqueId(); 7032 7033 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 7034 &inStream); 7035 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 7036 inStream, 7037 config.sample_rate, 7038 config.format, 7039 config.channel_mask, 7040 status); 7041 7042 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 7043 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 7044 // or stereo to mono conversions on 16 bit PCM inputs. 7045 if (status == BAD_VALUE && 7046 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 7047 (config.sample_rate <= 2 * reqSamplingRate) && 7048 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 7049 ALOGV("openInput() reopening with proposed sampling rate and channel mask"); 7050 inStream = NULL; 7051 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 7052 } 7053 7054 if (status == NO_ERROR && inStream != NULL) { 7055 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 7056 7057 // Start record thread 7058 // RecorThread require both input and output device indication to forward to audio 7059 // pre processing modules 7060 audio_devices_t device = (*pDevices) | primaryOutputDevice_l(); 7061 thread = new RecordThread(this, 7062 input, 7063 reqSamplingRate, 7064 reqChannels, 7065 id, 7066 device); 7067 mRecordThreads.add(id, thread); 7068 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 7069 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 7070 if (pFormat != NULL) *pFormat = config.format; 7071 if (pChannelMask != NULL) *pChannelMask = reqChannels; 7072 7073 // notify client processes of the new input creation 7074 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 7075 return id; 7076 } 7077 7078 return 0; 7079} 7080 7081status_t AudioFlinger::closeInput(audio_io_handle_t input) 7082{ 7083 return closeInput_nonvirtual(input); 7084} 7085 7086status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 7087{ 7088 // keep strong reference on the record thread so that 7089 // it is not destroyed while exit() is executed 7090 sp<RecordThread> thread; 7091 { 7092 Mutex::Autolock _l(mLock); 7093 thread = checkRecordThread_l(input); 7094 if (thread == 0) { 7095 return BAD_VALUE; 7096 } 7097 7098 ALOGV("closeInput() %d", input); 7099 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 7100 mRecordThreads.removeItem(input); 7101 } 7102 thread->exit(); 7103 // The thread entity (active unit of execution) is no longer running here, 7104 // but the ThreadBase container still exists. 7105 7106 AudioStreamIn *in = thread->clearInput(); 7107 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 7108 // from now on thread->mInput is NULL 7109 in->hwDev->close_input_stream(in->hwDev, in->stream); 7110 delete in; 7111 7112 return NO_ERROR; 7113} 7114 7115status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 7116{ 7117 Mutex::Autolock _l(mLock); 7118 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 7119 7120 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7121 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7122 thread->invalidateTracks(stream); 7123 } 7124 7125 return NO_ERROR; 7126} 7127 7128 7129int AudioFlinger::newAudioSessionId() 7130{ 7131 return nextUniqueId(); 7132} 7133 7134void AudioFlinger::acquireAudioSessionId(int audioSession) 7135{ 7136 Mutex::Autolock _l(mLock); 7137 pid_t caller = IPCThreadState::self()->getCallingPid(); 7138 ALOGV("acquiring %d from %d", audioSession, caller); 7139 size_t num = mAudioSessionRefs.size(); 7140 for (size_t i = 0; i< num; i++) { 7141 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 7142 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7143 ref->mCnt++; 7144 ALOGV(" incremented refcount to %d", ref->mCnt); 7145 return; 7146 } 7147 } 7148 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 7149 ALOGV(" added new entry for %d", audioSession); 7150} 7151 7152void AudioFlinger::releaseAudioSessionId(int audioSession) 7153{ 7154 Mutex::Autolock _l(mLock); 7155 pid_t caller = IPCThreadState::self()->getCallingPid(); 7156 ALOGV("releasing %d from %d", audioSession, caller); 7157 size_t num = mAudioSessionRefs.size(); 7158 for (size_t i = 0; i< num; i++) { 7159 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 7160 if (ref->mSessionid == audioSession && ref->mPid == caller) { 7161 ref->mCnt--; 7162 ALOGV(" decremented refcount to %d", ref->mCnt); 7163 if (ref->mCnt == 0) { 7164 mAudioSessionRefs.removeAt(i); 7165 delete ref; 7166 purgeStaleEffects_l(); 7167 } 7168 return; 7169 } 7170 } 7171 ALOGW("session id %d not found for pid %d", audioSession, caller); 7172} 7173 7174void AudioFlinger::purgeStaleEffects_l() { 7175 7176 ALOGV("purging stale effects"); 7177 7178 Vector< sp<EffectChain> > chains; 7179 7180 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7181 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7182 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7183 sp<EffectChain> ec = t->mEffectChains[j]; 7184 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7185 chains.push(ec); 7186 } 7187 } 7188 } 7189 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7190 sp<RecordThread> t = mRecordThreads.valueAt(i); 7191 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7192 sp<EffectChain> ec = t->mEffectChains[j]; 7193 chains.push(ec); 7194 } 7195 } 7196 7197 for (size_t i = 0; i < chains.size(); i++) { 7198 sp<EffectChain> ec = chains[i]; 7199 int sessionid = ec->sessionId(); 7200 sp<ThreadBase> t = ec->mThread.promote(); 7201 if (t == 0) { 7202 continue; 7203 } 7204 size_t numsessionrefs = mAudioSessionRefs.size(); 7205 bool found = false; 7206 for (size_t k = 0; k < numsessionrefs; k++) { 7207 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7208 if (ref->mSessionid == sessionid) { 7209 ALOGV(" session %d still exists for %d with %d refs", 7210 sessionid, ref->mPid, ref->mCnt); 7211 found = true; 7212 break; 7213 } 7214 } 7215 if (!found) { 7216 Mutex::Autolock _l (t->mLock); 7217 // remove all effects from the chain 7218 while (ec->mEffects.size()) { 7219 sp<EffectModule> effect = ec->mEffects[0]; 7220 effect->unPin(); 7221 t->removeEffect_l(effect); 7222 if (effect->purgeHandles()) { 7223 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7224 } 7225 AudioSystem::unregisterEffect(effect->id()); 7226 } 7227 } 7228 } 7229 return; 7230} 7231 7232// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7233AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7234{ 7235 return mPlaybackThreads.valueFor(output).get(); 7236} 7237 7238// checkMixerThread_l() must be called with AudioFlinger::mLock held 7239AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7240{ 7241 PlaybackThread *thread = checkPlaybackThread_l(output); 7242 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7243} 7244 7245// checkRecordThread_l() must be called with AudioFlinger::mLock held 7246AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7247{ 7248 return mRecordThreads.valueFor(input).get(); 7249} 7250 7251uint32_t AudioFlinger::nextUniqueId() 7252{ 7253 return android_atomic_inc(&mNextUniqueId); 7254} 7255 7256AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7257{ 7258 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7259 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7260 AudioStreamOut *output = thread->getOutput(); 7261 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7262 return thread; 7263 } 7264 } 7265 return NULL; 7266} 7267 7268audio_devices_t AudioFlinger::primaryOutputDevice_l() const 7269{ 7270 PlaybackThread *thread = primaryPlaybackThread_l(); 7271 7272 if (thread == NULL) { 7273 return 0; 7274 } 7275 7276 return thread->device(); 7277} 7278 7279sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7280 int triggerSession, 7281 int listenerSession, 7282 sync_event_callback_t callBack, 7283 void *cookie) 7284{ 7285 Mutex::Autolock _l(mLock); 7286 7287 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7288 status_t playStatus = NAME_NOT_FOUND; 7289 status_t recStatus = NAME_NOT_FOUND; 7290 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7291 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7292 if (playStatus == NO_ERROR) { 7293 return event; 7294 } 7295 } 7296 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7297 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7298 if (recStatus == NO_ERROR) { 7299 return event; 7300 } 7301 } 7302 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7303 mPendingSyncEvents.add(event); 7304 } else { 7305 ALOGV("createSyncEvent() invalid event %d", event->type()); 7306 event.clear(); 7307 } 7308 return event; 7309} 7310 7311// ---------------------------------------------------------------------------- 7312// Effect management 7313// ---------------------------------------------------------------------------- 7314 7315 7316status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7317{ 7318 Mutex::Autolock _l(mLock); 7319 return EffectQueryNumberEffects(numEffects); 7320} 7321 7322status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7323{ 7324 Mutex::Autolock _l(mLock); 7325 return EffectQueryEffect(index, descriptor); 7326} 7327 7328status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7329 effect_descriptor_t *descriptor) const 7330{ 7331 Mutex::Autolock _l(mLock); 7332 return EffectGetDescriptor(pUuid, descriptor); 7333} 7334 7335 7336sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7337 effect_descriptor_t *pDesc, 7338 const sp<IEffectClient>& effectClient, 7339 int32_t priority, 7340 audio_io_handle_t io, 7341 int sessionId, 7342 status_t *status, 7343 int *id, 7344 int *enabled) 7345{ 7346 status_t lStatus = NO_ERROR; 7347 sp<EffectHandle> handle; 7348 effect_descriptor_t desc; 7349 7350 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7351 pid, effectClient.get(), priority, sessionId, io); 7352 7353 if (pDesc == NULL) { 7354 lStatus = BAD_VALUE; 7355 goto Exit; 7356 } 7357 7358 // check audio settings permission for global effects 7359 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7360 lStatus = PERMISSION_DENIED; 7361 goto Exit; 7362 } 7363 7364 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7365 // that can only be created by audio policy manager (running in same process) 7366 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7367 lStatus = PERMISSION_DENIED; 7368 goto Exit; 7369 } 7370 7371 if (io == 0) { 7372 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7373 // output must be specified by AudioPolicyManager when using session 7374 // AUDIO_SESSION_OUTPUT_STAGE 7375 lStatus = BAD_VALUE; 7376 goto Exit; 7377 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7378 // if the output returned by getOutputForEffect() is removed before we lock the 7379 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7380 // and we will exit safely 7381 io = AudioSystem::getOutputForEffect(&desc); 7382 } 7383 } 7384 7385 { 7386 Mutex::Autolock _l(mLock); 7387 7388 7389 if (!EffectIsNullUuid(&pDesc->uuid)) { 7390 // if uuid is specified, request effect descriptor 7391 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7392 if (lStatus < 0) { 7393 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7394 goto Exit; 7395 } 7396 } else { 7397 // if uuid is not specified, look for an available implementation 7398 // of the required type in effect factory 7399 if (EffectIsNullUuid(&pDesc->type)) { 7400 ALOGW("createEffect() no effect type"); 7401 lStatus = BAD_VALUE; 7402 goto Exit; 7403 } 7404 uint32_t numEffects = 0; 7405 effect_descriptor_t d; 7406 d.flags = 0; // prevent compiler warning 7407 bool found = false; 7408 7409 lStatus = EffectQueryNumberEffects(&numEffects); 7410 if (lStatus < 0) { 7411 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7412 goto Exit; 7413 } 7414 for (uint32_t i = 0; i < numEffects; i++) { 7415 lStatus = EffectQueryEffect(i, &desc); 7416 if (lStatus < 0) { 7417 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7418 continue; 7419 } 7420 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7421 // If matching type found save effect descriptor. If the session is 7422 // 0 and the effect is not auxiliary, continue enumeration in case 7423 // an auxiliary version of this effect type is available 7424 found = true; 7425 d = desc; 7426 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7427 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7428 break; 7429 } 7430 } 7431 } 7432 if (!found) { 7433 lStatus = BAD_VALUE; 7434 ALOGW("createEffect() effect not found"); 7435 goto Exit; 7436 } 7437 // For same effect type, chose auxiliary version over insert version if 7438 // connect to output mix (Compliance to OpenSL ES) 7439 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7440 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7441 desc = d; 7442 } 7443 } 7444 7445 // Do not allow auxiliary effects on a session different from 0 (output mix) 7446 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7447 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7448 lStatus = INVALID_OPERATION; 7449 goto Exit; 7450 } 7451 7452 // check recording permission for visualizer 7453 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7454 !recordingAllowed()) { 7455 lStatus = PERMISSION_DENIED; 7456 goto Exit; 7457 } 7458 7459 // return effect descriptor 7460 *pDesc = desc; 7461 7462 // If output is not specified try to find a matching audio session ID in one of the 7463 // output threads. 7464 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7465 // because of code checking output when entering the function. 7466 // Note: io is never 0 when creating an effect on an input 7467 if (io == 0) { 7468 // look for the thread where the specified audio session is present 7469 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7470 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7471 io = mPlaybackThreads.keyAt(i); 7472 break; 7473 } 7474 } 7475 if (io == 0) { 7476 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7477 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7478 io = mRecordThreads.keyAt(i); 7479 break; 7480 } 7481 } 7482 } 7483 // If no output thread contains the requested session ID, default to 7484 // first output. The effect chain will be moved to the correct output 7485 // thread when a track with the same session ID is created 7486 if (io == 0 && mPlaybackThreads.size()) { 7487 io = mPlaybackThreads.keyAt(0); 7488 } 7489 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7490 } 7491 ThreadBase *thread = checkRecordThread_l(io); 7492 if (thread == NULL) { 7493 thread = checkPlaybackThread_l(io); 7494 if (thread == NULL) { 7495 ALOGE("createEffect() unknown output thread"); 7496 lStatus = BAD_VALUE; 7497 goto Exit; 7498 } 7499 } 7500 7501 sp<Client> client = registerPid_l(pid); 7502 7503 // create effect on selected output thread 7504 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7505 &desc, enabled, &lStatus); 7506 if (handle != 0 && id != NULL) { 7507 *id = handle->id(); 7508 } 7509 } 7510 7511Exit: 7512 if (status != NULL) { 7513 *status = lStatus; 7514 } 7515 return handle; 7516} 7517 7518status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7519 audio_io_handle_t dstOutput) 7520{ 7521 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7522 sessionId, srcOutput, dstOutput); 7523 Mutex::Autolock _l(mLock); 7524 if (srcOutput == dstOutput) { 7525 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7526 return NO_ERROR; 7527 } 7528 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7529 if (srcThread == NULL) { 7530 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7531 return BAD_VALUE; 7532 } 7533 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7534 if (dstThread == NULL) { 7535 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7536 return BAD_VALUE; 7537 } 7538 7539 Mutex::Autolock _dl(dstThread->mLock); 7540 Mutex::Autolock _sl(srcThread->mLock); 7541 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7542 7543 return NO_ERROR; 7544} 7545 7546// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7547status_t AudioFlinger::moveEffectChain_l(int sessionId, 7548 AudioFlinger::PlaybackThread *srcThread, 7549 AudioFlinger::PlaybackThread *dstThread, 7550 bool reRegister) 7551{ 7552 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7553 sessionId, srcThread, dstThread); 7554 7555 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7556 if (chain == 0) { 7557 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7558 sessionId, srcThread); 7559 return INVALID_OPERATION; 7560 } 7561 7562 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7563 // so that a new chain is created with correct parameters when first effect is added. This is 7564 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7565 // removed. 7566 srcThread->removeEffectChain_l(chain); 7567 7568 // transfer all effects one by one so that new effect chain is created on new thread with 7569 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7570 audio_io_handle_t dstOutput = dstThread->id(); 7571 sp<EffectChain> dstChain; 7572 uint32_t strategy = 0; // prevent compiler warning 7573 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7574 while (effect != 0) { 7575 srcThread->removeEffect_l(effect); 7576 dstThread->addEffect_l(effect); 7577 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7578 if (effect->state() == EffectModule::ACTIVE || 7579 effect->state() == EffectModule::STOPPING) { 7580 effect->start(); 7581 } 7582 // if the move request is not received from audio policy manager, the effect must be 7583 // re-registered with the new strategy and output 7584 if (dstChain == 0) { 7585 dstChain = effect->chain().promote(); 7586 if (dstChain == 0) { 7587 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7588 srcThread->addEffect_l(effect); 7589 return NO_INIT; 7590 } 7591 strategy = dstChain->strategy(); 7592 } 7593 if (reRegister) { 7594 AudioSystem::unregisterEffect(effect->id()); 7595 AudioSystem::registerEffect(&effect->desc(), 7596 dstOutput, 7597 strategy, 7598 sessionId, 7599 effect->id()); 7600 } 7601 effect = chain->getEffectFromId_l(0); 7602 } 7603 7604 return NO_ERROR; 7605} 7606 7607 7608// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7609sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7610 const sp<AudioFlinger::Client>& client, 7611 const sp<IEffectClient>& effectClient, 7612 int32_t priority, 7613 int sessionId, 7614 effect_descriptor_t *desc, 7615 int *enabled, 7616 status_t *status 7617 ) 7618{ 7619 sp<EffectModule> effect; 7620 sp<EffectHandle> handle; 7621 status_t lStatus; 7622 sp<EffectChain> chain; 7623 bool chainCreated = false; 7624 bool effectCreated = false; 7625 bool effectRegistered = false; 7626 7627 lStatus = initCheck(); 7628 if (lStatus != NO_ERROR) { 7629 ALOGW("createEffect_l() Audio driver not initialized."); 7630 goto Exit; 7631 } 7632 7633 // Do not allow effects with session ID 0 on direct output or duplicating threads 7634 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7635 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7636 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7637 desc->name, sessionId); 7638 lStatus = BAD_VALUE; 7639 goto Exit; 7640 } 7641 // Only Pre processor effects are allowed on input threads and only on input threads 7642 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7643 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7644 desc->name, desc->flags, mType); 7645 lStatus = BAD_VALUE; 7646 goto Exit; 7647 } 7648 7649 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7650 7651 { // scope for mLock 7652 Mutex::Autolock _l(mLock); 7653 7654 // check for existing effect chain with the requested audio session 7655 chain = getEffectChain_l(sessionId); 7656 if (chain == 0) { 7657 // create a new chain for this session 7658 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7659 chain = new EffectChain(this, sessionId); 7660 addEffectChain_l(chain); 7661 chain->setStrategy(getStrategyForSession_l(sessionId)); 7662 chainCreated = true; 7663 } else { 7664 effect = chain->getEffectFromDesc_l(desc); 7665 } 7666 7667 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7668 7669 if (effect == 0) { 7670 int id = mAudioFlinger->nextUniqueId(); 7671 // Check CPU and memory usage 7672 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7673 if (lStatus != NO_ERROR) { 7674 goto Exit; 7675 } 7676 effectRegistered = true; 7677 // create a new effect module if none present in the chain 7678 effect = new EffectModule(this, chain, desc, id, sessionId); 7679 lStatus = effect->status(); 7680 if (lStatus != NO_ERROR) { 7681 goto Exit; 7682 } 7683 lStatus = chain->addEffect_l(effect); 7684 if (lStatus != NO_ERROR) { 7685 goto Exit; 7686 } 7687 effectCreated = true; 7688 7689 effect->setDevice(mDevice); 7690 effect->setMode(mAudioFlinger->getMode()); 7691 } 7692 // create effect handle and connect it to effect module 7693 handle = new EffectHandle(effect, client, effectClient, priority); 7694 lStatus = effect->addHandle(handle.get()); 7695 if (enabled != NULL) { 7696 *enabled = (int)effect->isEnabled(); 7697 } 7698 } 7699 7700Exit: 7701 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7702 Mutex::Autolock _l(mLock); 7703 if (effectCreated) { 7704 chain->removeEffect_l(effect); 7705 } 7706 if (effectRegistered) { 7707 AudioSystem::unregisterEffect(effect->id()); 7708 } 7709 if (chainCreated) { 7710 removeEffectChain_l(chain); 7711 } 7712 handle.clear(); 7713 } 7714 7715 if (status != NULL) { 7716 *status = lStatus; 7717 } 7718 return handle; 7719} 7720 7721sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect(int sessionId, int effectId) 7722{ 7723 Mutex::Autolock _l(mLock); 7724 return getEffect_l(sessionId, effectId); 7725} 7726 7727sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7728{ 7729 sp<EffectChain> chain = getEffectChain_l(sessionId); 7730 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7731} 7732 7733// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7734// PlaybackThread::mLock held 7735status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7736{ 7737 // check for existing effect chain with the requested audio session 7738 int sessionId = effect->sessionId(); 7739 sp<EffectChain> chain = getEffectChain_l(sessionId); 7740 bool chainCreated = false; 7741 7742 if (chain == 0) { 7743 // create a new chain for this session 7744 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7745 chain = new EffectChain(this, sessionId); 7746 addEffectChain_l(chain); 7747 chain->setStrategy(getStrategyForSession_l(sessionId)); 7748 chainCreated = true; 7749 } 7750 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7751 7752 if (chain->getEffectFromId_l(effect->id()) != 0) { 7753 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7754 this, effect->desc().name, chain.get()); 7755 return BAD_VALUE; 7756 } 7757 7758 status_t status = chain->addEffect_l(effect); 7759 if (status != NO_ERROR) { 7760 if (chainCreated) { 7761 removeEffectChain_l(chain); 7762 } 7763 return status; 7764 } 7765 7766 effect->setDevice(mDevice); 7767 effect->setMode(mAudioFlinger->getMode()); 7768 return NO_ERROR; 7769} 7770 7771void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7772 7773 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7774 effect_descriptor_t desc = effect->desc(); 7775 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7776 detachAuxEffect_l(effect->id()); 7777 } 7778 7779 sp<EffectChain> chain = effect->chain().promote(); 7780 if (chain != 0) { 7781 // remove effect chain if removing last effect 7782 if (chain->removeEffect_l(effect) == 0) { 7783 removeEffectChain_l(chain); 7784 } 7785 } else { 7786 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7787 } 7788} 7789 7790void AudioFlinger::ThreadBase::lockEffectChains_l( 7791 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7792{ 7793 effectChains = mEffectChains; 7794 for (size_t i = 0; i < mEffectChains.size(); i++) { 7795 mEffectChains[i]->lock(); 7796 } 7797} 7798 7799void AudioFlinger::ThreadBase::unlockEffectChains( 7800 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7801{ 7802 for (size_t i = 0; i < effectChains.size(); i++) { 7803 effectChains[i]->unlock(); 7804 } 7805} 7806 7807sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7808{ 7809 Mutex::Autolock _l(mLock); 7810 return getEffectChain_l(sessionId); 7811} 7812 7813sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7814{ 7815 size_t size = mEffectChains.size(); 7816 for (size_t i = 0; i < size; i++) { 7817 if (mEffectChains[i]->sessionId() == sessionId) { 7818 return mEffectChains[i]; 7819 } 7820 } 7821 return 0; 7822} 7823 7824void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7825{ 7826 Mutex::Autolock _l(mLock); 7827 size_t size = mEffectChains.size(); 7828 for (size_t i = 0; i < size; i++) { 7829 mEffectChains[i]->setMode_l(mode); 7830 } 7831} 7832 7833void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7834 EffectHandle *handle, 7835 bool unpinIfLast) { 7836 7837 Mutex::Autolock _l(mLock); 7838 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7839 // delete the effect module if removing last handle on it 7840 if (effect->removeHandle(handle) == 0) { 7841 if (!effect->isPinned() || unpinIfLast) { 7842 removeEffect_l(effect); 7843 AudioSystem::unregisterEffect(effect->id()); 7844 } 7845 } 7846} 7847 7848status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7849{ 7850 int session = chain->sessionId(); 7851 int16_t *buffer = mMixBuffer; 7852 bool ownsBuffer = false; 7853 7854 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7855 if (session > 0) { 7856 // Only one effect chain can be present in direct output thread and it uses 7857 // the mix buffer as input 7858 if (mType != DIRECT) { 7859 size_t numSamples = mNormalFrameCount * mChannelCount; 7860 buffer = new int16_t[numSamples]; 7861 memset(buffer, 0, numSamples * sizeof(int16_t)); 7862 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7863 ownsBuffer = true; 7864 } 7865 7866 // Attach all tracks with same session ID to this chain. 7867 for (size_t i = 0; i < mTracks.size(); ++i) { 7868 sp<Track> track = mTracks[i]; 7869 if (session == track->sessionId()) { 7870 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7871 track->setMainBuffer(buffer); 7872 chain->incTrackCnt(); 7873 } 7874 } 7875 7876 // indicate all active tracks in the chain 7877 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7878 sp<Track> track = mActiveTracks[i].promote(); 7879 if (track == 0) continue; 7880 if (session == track->sessionId()) { 7881 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7882 chain->incActiveTrackCnt(); 7883 } 7884 } 7885 } 7886 7887 chain->setInBuffer(buffer, ownsBuffer); 7888 chain->setOutBuffer(mMixBuffer); 7889 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7890 // chains list in order to be processed last as it contains output stage effects 7891 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7892 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7893 // after track specific effects and before output stage 7894 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7895 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7896 // Effect chain for other sessions are inserted at beginning of effect 7897 // chains list to be processed before output mix effects. Relative order between other 7898 // sessions is not important 7899 size_t size = mEffectChains.size(); 7900 size_t i = 0; 7901 for (i = 0; i < size; i++) { 7902 if (mEffectChains[i]->sessionId() < session) break; 7903 } 7904 mEffectChains.insertAt(chain, i); 7905 checkSuspendOnAddEffectChain_l(chain); 7906 7907 return NO_ERROR; 7908} 7909 7910size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7911{ 7912 int session = chain->sessionId(); 7913 7914 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7915 7916 for (size_t i = 0; i < mEffectChains.size(); i++) { 7917 if (chain == mEffectChains[i]) { 7918 mEffectChains.removeAt(i); 7919 // detach all active tracks from the chain 7920 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7921 sp<Track> track = mActiveTracks[i].promote(); 7922 if (track == 0) continue; 7923 if (session == track->sessionId()) { 7924 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7925 chain.get(), session); 7926 chain->decActiveTrackCnt(); 7927 } 7928 } 7929 7930 // detach all tracks with same session ID from this chain 7931 for (size_t i = 0; i < mTracks.size(); ++i) { 7932 sp<Track> track = mTracks[i]; 7933 if (session == track->sessionId()) { 7934 track->setMainBuffer(mMixBuffer); 7935 chain->decTrackCnt(); 7936 } 7937 } 7938 break; 7939 } 7940 } 7941 return mEffectChains.size(); 7942} 7943 7944status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7945 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7946{ 7947 Mutex::Autolock _l(mLock); 7948 return attachAuxEffect_l(track, EffectId); 7949} 7950 7951status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7952 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7953{ 7954 status_t status = NO_ERROR; 7955 7956 if (EffectId == 0) { 7957 track->setAuxBuffer(0, NULL); 7958 } else { 7959 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7960 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7961 if (effect != 0) { 7962 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7963 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7964 } else { 7965 status = INVALID_OPERATION; 7966 } 7967 } else { 7968 status = BAD_VALUE; 7969 } 7970 } 7971 return status; 7972} 7973 7974void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7975{ 7976 for (size_t i = 0; i < mTracks.size(); ++i) { 7977 sp<Track> track = mTracks[i]; 7978 if (track->auxEffectId() == effectId) { 7979 attachAuxEffect_l(track, 0); 7980 } 7981 } 7982} 7983 7984status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7985{ 7986 // only one chain per input thread 7987 if (mEffectChains.size() != 0) { 7988 return INVALID_OPERATION; 7989 } 7990 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7991 7992 chain->setInBuffer(NULL); 7993 chain->setOutBuffer(NULL); 7994 7995 checkSuspendOnAddEffectChain_l(chain); 7996 7997 mEffectChains.add(chain); 7998 7999 return NO_ERROR; 8000} 8001 8002size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 8003{ 8004 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 8005 ALOGW_IF(mEffectChains.size() != 1, 8006 "removeEffectChain_l() %p invalid chain size %d on thread %p", 8007 chain.get(), mEffectChains.size(), this); 8008 if (mEffectChains.size() == 1) { 8009 mEffectChains.removeAt(0); 8010 } 8011 return 0; 8012} 8013 8014// ---------------------------------------------------------------------------- 8015// EffectModule implementation 8016// ---------------------------------------------------------------------------- 8017 8018#undef LOG_TAG 8019#define LOG_TAG "AudioFlinger::EffectModule" 8020 8021AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 8022 const wp<AudioFlinger::EffectChain>& chain, 8023 effect_descriptor_t *desc, 8024 int id, 8025 int sessionId) 8026 : mPinned(sessionId > AUDIO_SESSION_OUTPUT_MIX), 8027 mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), 8028 mDescriptor(*desc), 8029 // mConfig is set by configure() and not used before then 8030 mEffectInterface(NULL), 8031 mStatus(NO_INIT), mState(IDLE), 8032 // mMaxDisableWaitCnt is set by configure() and not used before then 8033 // mDisableWaitCnt is set by process() and updateState() and not used before then 8034 mSuspended(false) 8035{ 8036 ALOGV("Constructor %p", this); 8037 int lStatus; 8038 8039 // create effect engine from effect factory 8040 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 8041 8042 if (mStatus != NO_ERROR) { 8043 return; 8044 } 8045 lStatus = init(); 8046 if (lStatus < 0) { 8047 mStatus = lStatus; 8048 goto Error; 8049 } 8050 8051 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 8052 return; 8053Error: 8054 EffectRelease(mEffectInterface); 8055 mEffectInterface = NULL; 8056 ALOGV("Constructor Error %d", mStatus); 8057} 8058 8059AudioFlinger::EffectModule::~EffectModule() 8060{ 8061 ALOGV("Destructor %p", this); 8062 if (mEffectInterface != NULL) { 8063 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8064 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 8065 sp<ThreadBase> thread = mThread.promote(); 8066 if (thread != 0) { 8067 audio_stream_t *stream = thread->stream(); 8068 if (stream != NULL) { 8069 stream->remove_audio_effect(stream, mEffectInterface); 8070 } 8071 } 8072 } 8073 // release effect engine 8074 EffectRelease(mEffectInterface); 8075 } 8076} 8077 8078status_t AudioFlinger::EffectModule::addHandle(EffectHandle *handle) 8079{ 8080 status_t status; 8081 8082 Mutex::Autolock _l(mLock); 8083 int priority = handle->priority(); 8084 size_t size = mHandles.size(); 8085 EffectHandle *controlHandle = NULL; 8086 size_t i; 8087 for (i = 0; i < size; i++) { 8088 EffectHandle *h = mHandles[i]; 8089 if (h == NULL || h->destroyed_l()) continue; 8090 // first non destroyed handle is considered in control 8091 if (controlHandle == NULL) 8092 controlHandle = h; 8093 if (h->priority() <= priority) break; 8094 } 8095 // if inserted in first place, move effect control from previous owner to this handle 8096 if (i == 0) { 8097 bool enabled = false; 8098 if (controlHandle != NULL) { 8099 enabled = controlHandle->enabled(); 8100 controlHandle->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 8101 } 8102 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 8103 status = NO_ERROR; 8104 } else { 8105 status = ALREADY_EXISTS; 8106 } 8107 ALOGV("addHandle() %p added handle %p in position %d", this, handle, i); 8108 mHandles.insertAt(handle, i); 8109 return status; 8110} 8111 8112size_t AudioFlinger::EffectModule::removeHandle(EffectHandle *handle) 8113{ 8114 Mutex::Autolock _l(mLock); 8115 size_t size = mHandles.size(); 8116 size_t i; 8117 for (i = 0; i < size; i++) { 8118 if (mHandles[i] == handle) break; 8119 } 8120 if (i == size) { 8121 return size; 8122 } 8123 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle, i); 8124 8125 mHandles.removeAt(i); 8126 // if removed from first place, move effect control from this handle to next in line 8127 if (i == 0) { 8128 EffectHandle *h = controlHandle_l(); 8129 if (h != NULL) { 8130 h->setControl(true /*hasControl*/, true /*signal*/ , handle->enabled() /*enabled*/); 8131 } 8132 } 8133 8134 // Prevent calls to process() and other functions on effect interface from now on. 8135 // The effect engine will be released by the destructor when the last strong reference on 8136 // this object is released which can happen after next process is called. 8137 if (mHandles.size() == 0 && !mPinned) { 8138 mState = DESTROYED; 8139 } 8140 8141 return mHandles.size(); 8142} 8143 8144// must be called with EffectModule::mLock held 8145AudioFlinger::EffectHandle *AudioFlinger::EffectModule::controlHandle_l() 8146{ 8147 // the first valid handle in the list has control over the module 8148 for (size_t i = 0; i < mHandles.size(); i++) { 8149 EffectHandle *h = mHandles[i]; 8150 if (h != NULL && !h->destroyed_l()) { 8151 return h; 8152 } 8153 } 8154 8155 return NULL; 8156} 8157 8158size_t AudioFlinger::EffectModule::disconnect(EffectHandle *handle, bool unpinIfLast) 8159{ 8160 ALOGV("disconnect() %p handle %p", this, handle); 8161 // keep a strong reference on this EffectModule to avoid calling the 8162 // destructor before we exit 8163 sp<EffectModule> keep(this); 8164 { 8165 sp<ThreadBase> thread = mThread.promote(); 8166 if (thread != 0) { 8167 thread->disconnectEffect(keep, handle, unpinIfLast); 8168 } 8169 } 8170 return mHandles.size(); 8171} 8172 8173void AudioFlinger::EffectModule::updateState() { 8174 Mutex::Autolock _l(mLock); 8175 8176 switch (mState) { 8177 case RESTART: 8178 reset_l(); 8179 // FALL THROUGH 8180 8181 case STARTING: 8182 // clear auxiliary effect input buffer for next accumulation 8183 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8184 memset(mConfig.inputCfg.buffer.raw, 8185 0, 8186 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8187 } 8188 start_l(); 8189 mState = ACTIVE; 8190 break; 8191 case STOPPING: 8192 stop_l(); 8193 mDisableWaitCnt = mMaxDisableWaitCnt; 8194 mState = STOPPED; 8195 break; 8196 case STOPPED: 8197 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8198 // turn off sequence. 8199 if (--mDisableWaitCnt == 0) { 8200 reset_l(); 8201 mState = IDLE; 8202 } 8203 break; 8204 default: //IDLE , ACTIVE, DESTROYED 8205 break; 8206 } 8207} 8208 8209void AudioFlinger::EffectModule::process() 8210{ 8211 Mutex::Autolock _l(mLock); 8212 8213 if (mState == DESTROYED || mEffectInterface == NULL || 8214 mConfig.inputCfg.buffer.raw == NULL || 8215 mConfig.outputCfg.buffer.raw == NULL) { 8216 return; 8217 } 8218 8219 if (isProcessEnabled()) { 8220 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8221 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8222 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8223 mConfig.inputCfg.buffer.s32, 8224 mConfig.inputCfg.buffer.frameCount/2); 8225 } 8226 8227 // do the actual processing in the effect engine 8228 int ret = (*mEffectInterface)->process(mEffectInterface, 8229 &mConfig.inputCfg.buffer, 8230 &mConfig.outputCfg.buffer); 8231 8232 // force transition to IDLE state when engine is ready 8233 if (mState == STOPPED && ret == -ENODATA) { 8234 mDisableWaitCnt = 1; 8235 } 8236 8237 // clear auxiliary effect input buffer for next accumulation 8238 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8239 memset(mConfig.inputCfg.buffer.raw, 0, 8240 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8241 } 8242 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8243 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8244 // If an insert effect is idle and input buffer is different from output buffer, 8245 // accumulate input onto output 8246 sp<EffectChain> chain = mChain.promote(); 8247 if (chain != 0 && chain->activeTrackCnt() != 0) { 8248 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8249 int16_t *in = mConfig.inputCfg.buffer.s16; 8250 int16_t *out = mConfig.outputCfg.buffer.s16; 8251 for (size_t i = 0; i < frameCnt; i++) { 8252 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8253 } 8254 } 8255 } 8256} 8257 8258void AudioFlinger::EffectModule::reset_l() 8259{ 8260 if (mEffectInterface == NULL) { 8261 return; 8262 } 8263 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8264} 8265 8266status_t AudioFlinger::EffectModule::configure() 8267{ 8268 if (mEffectInterface == NULL) { 8269 return NO_INIT; 8270 } 8271 8272 sp<ThreadBase> thread = mThread.promote(); 8273 if (thread == 0) { 8274 return DEAD_OBJECT; 8275 } 8276 8277 // TODO: handle configuration of effects replacing track process 8278 audio_channel_mask_t channelMask = thread->channelMask(); 8279 8280 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8281 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8282 } else { 8283 mConfig.inputCfg.channels = channelMask; 8284 } 8285 mConfig.outputCfg.channels = channelMask; 8286 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8287 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8288 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8289 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8290 mConfig.inputCfg.bufferProvider.cookie = NULL; 8291 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8292 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8293 mConfig.outputCfg.bufferProvider.cookie = NULL; 8294 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8295 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8296 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8297 // Insert effect: 8298 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8299 // always overwrites output buffer: input buffer == output buffer 8300 // - in other sessions: 8301 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8302 // other effect: overwrites output buffer: input buffer == output buffer 8303 // Auxiliary effect: 8304 // accumulates in output buffer: input buffer != output buffer 8305 // Therefore: accumulate <=> input buffer != output buffer 8306 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8307 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8308 } else { 8309 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8310 } 8311 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8312 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8313 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8314 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8315 8316 ALOGV("configure() %p thread %p buffer %p framecount %d", 8317 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8318 8319 status_t cmdStatus; 8320 uint32_t size = sizeof(int); 8321 status_t status = (*mEffectInterface)->command(mEffectInterface, 8322 EFFECT_CMD_SET_CONFIG, 8323 sizeof(effect_config_t), 8324 &mConfig, 8325 &size, 8326 &cmdStatus); 8327 if (status == 0) { 8328 status = cmdStatus; 8329 } 8330 8331 if (status == 0 && 8332 (memcmp(&mDescriptor.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0)) { 8333 uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; 8334 effect_param_t *p = (effect_param_t *)buf32; 8335 8336 p->psize = sizeof(uint32_t); 8337 p->vsize = sizeof(uint32_t); 8338 size = sizeof(int); 8339 *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY; 8340 8341 uint32_t latency = 0; 8342 PlaybackThread *pbt = thread->mAudioFlinger->checkPlaybackThread_l(thread->mId); 8343 if (pbt != NULL) { 8344 latency = pbt->latency_l(); 8345 } 8346 8347 *((int32_t *)p->data + 1)= latency; 8348 (*mEffectInterface)->command(mEffectInterface, 8349 EFFECT_CMD_SET_PARAM, 8350 sizeof(effect_param_t) + 8, 8351 &buf32, 8352 &size, 8353 &cmdStatus); 8354 } 8355 8356 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8357 (1000 * mConfig.outputCfg.buffer.frameCount); 8358 8359 return status; 8360} 8361 8362status_t AudioFlinger::EffectModule::init() 8363{ 8364 Mutex::Autolock _l(mLock); 8365 if (mEffectInterface == NULL) { 8366 return NO_INIT; 8367 } 8368 status_t cmdStatus; 8369 uint32_t size = sizeof(status_t); 8370 status_t status = (*mEffectInterface)->command(mEffectInterface, 8371 EFFECT_CMD_INIT, 8372 0, 8373 NULL, 8374 &size, 8375 &cmdStatus); 8376 if (status == 0) { 8377 status = cmdStatus; 8378 } 8379 return status; 8380} 8381 8382status_t AudioFlinger::EffectModule::start() 8383{ 8384 Mutex::Autolock _l(mLock); 8385 return start_l(); 8386} 8387 8388status_t AudioFlinger::EffectModule::start_l() 8389{ 8390 if (mEffectInterface == NULL) { 8391 return NO_INIT; 8392 } 8393 status_t cmdStatus; 8394 uint32_t size = sizeof(status_t); 8395 status_t status = (*mEffectInterface)->command(mEffectInterface, 8396 EFFECT_CMD_ENABLE, 8397 0, 8398 NULL, 8399 &size, 8400 &cmdStatus); 8401 if (status == 0) { 8402 status = cmdStatus; 8403 } 8404 if (status == 0 && 8405 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8406 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8407 sp<ThreadBase> thread = mThread.promote(); 8408 if (thread != 0) { 8409 audio_stream_t *stream = thread->stream(); 8410 if (stream != NULL) { 8411 stream->add_audio_effect(stream, mEffectInterface); 8412 } 8413 } 8414 } 8415 return status; 8416} 8417 8418status_t AudioFlinger::EffectModule::stop() 8419{ 8420 Mutex::Autolock _l(mLock); 8421 return stop_l(); 8422} 8423 8424status_t AudioFlinger::EffectModule::stop_l() 8425{ 8426 if (mEffectInterface == NULL) { 8427 return NO_INIT; 8428 } 8429 status_t cmdStatus; 8430 uint32_t size = sizeof(status_t); 8431 status_t status = (*mEffectInterface)->command(mEffectInterface, 8432 EFFECT_CMD_DISABLE, 8433 0, 8434 NULL, 8435 &size, 8436 &cmdStatus); 8437 if (status == 0) { 8438 status = cmdStatus; 8439 } 8440 if (status == 0 && 8441 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8442 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8443 sp<ThreadBase> thread = mThread.promote(); 8444 if (thread != 0) { 8445 audio_stream_t *stream = thread->stream(); 8446 if (stream != NULL) { 8447 stream->remove_audio_effect(stream, mEffectInterface); 8448 } 8449 } 8450 } 8451 return status; 8452} 8453 8454status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8455 uint32_t cmdSize, 8456 void *pCmdData, 8457 uint32_t *replySize, 8458 void *pReplyData) 8459{ 8460 Mutex::Autolock _l(mLock); 8461// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8462 8463 if (mState == DESTROYED || mEffectInterface == NULL) { 8464 return NO_INIT; 8465 } 8466 status_t status = (*mEffectInterface)->command(mEffectInterface, 8467 cmdCode, 8468 cmdSize, 8469 pCmdData, 8470 replySize, 8471 pReplyData); 8472 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8473 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8474 for (size_t i = 1; i < mHandles.size(); i++) { 8475 EffectHandle *h = mHandles[i]; 8476 if (h != NULL && !h->destroyed_l()) { 8477 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8478 } 8479 } 8480 } 8481 return status; 8482} 8483 8484status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8485{ 8486 Mutex::Autolock _l(mLock); 8487 return setEnabled_l(enabled); 8488} 8489 8490// must be called with EffectModule::mLock held 8491status_t AudioFlinger::EffectModule::setEnabled_l(bool enabled) 8492{ 8493 8494 ALOGV("setEnabled %p enabled %d", this, enabled); 8495 8496 if (enabled != isEnabled()) { 8497 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8498 if (enabled && status != NO_ERROR) { 8499 return status; 8500 } 8501 8502 switch (mState) { 8503 // going from disabled to enabled 8504 case IDLE: 8505 mState = STARTING; 8506 break; 8507 case STOPPED: 8508 mState = RESTART; 8509 break; 8510 case STOPPING: 8511 mState = ACTIVE; 8512 break; 8513 8514 // going from enabled to disabled 8515 case RESTART: 8516 mState = STOPPED; 8517 break; 8518 case STARTING: 8519 mState = IDLE; 8520 break; 8521 case ACTIVE: 8522 mState = STOPPING; 8523 break; 8524 case DESTROYED: 8525 return NO_ERROR; // simply ignore as we are being destroyed 8526 } 8527 for (size_t i = 1; i < mHandles.size(); i++) { 8528 EffectHandle *h = mHandles[i]; 8529 if (h != NULL && !h->destroyed_l()) { 8530 h->setEnabled(enabled); 8531 } 8532 } 8533 } 8534 return NO_ERROR; 8535} 8536 8537bool AudioFlinger::EffectModule::isEnabled() const 8538{ 8539 switch (mState) { 8540 case RESTART: 8541 case STARTING: 8542 case ACTIVE: 8543 return true; 8544 case IDLE: 8545 case STOPPING: 8546 case STOPPED: 8547 case DESTROYED: 8548 default: 8549 return false; 8550 } 8551} 8552 8553bool AudioFlinger::EffectModule::isProcessEnabled() const 8554{ 8555 switch (mState) { 8556 case RESTART: 8557 case ACTIVE: 8558 case STOPPING: 8559 case STOPPED: 8560 return true; 8561 case IDLE: 8562 case STARTING: 8563 case DESTROYED: 8564 default: 8565 return false; 8566 } 8567} 8568 8569status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8570{ 8571 Mutex::Autolock _l(mLock); 8572 status_t status = NO_ERROR; 8573 8574 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8575 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8576 if (isProcessEnabled() && 8577 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8578 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8579 status_t cmdStatus; 8580 uint32_t volume[2]; 8581 uint32_t *pVolume = NULL; 8582 uint32_t size = sizeof(volume); 8583 volume[0] = *left; 8584 volume[1] = *right; 8585 if (controller) { 8586 pVolume = volume; 8587 } 8588 status = (*mEffectInterface)->command(mEffectInterface, 8589 EFFECT_CMD_SET_VOLUME, 8590 size, 8591 volume, 8592 &size, 8593 pVolume); 8594 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8595 *left = volume[0]; 8596 *right = volume[1]; 8597 } 8598 } 8599 return status; 8600} 8601 8602status_t AudioFlinger::EffectModule::setDevice(audio_devices_t device) 8603{ 8604 Mutex::Autolock _l(mLock); 8605 status_t status = NO_ERROR; 8606 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8607 // audio pre processing modules on RecordThread can receive both output and 8608 // input device indication in the same call 8609 audio_devices_t dev = device & AUDIO_DEVICE_OUT_ALL; 8610 if (dev) { 8611 status_t cmdStatus; 8612 uint32_t size = sizeof(status_t); 8613 8614 status = (*mEffectInterface)->command(mEffectInterface, 8615 EFFECT_CMD_SET_DEVICE, 8616 sizeof(uint32_t), 8617 &dev, 8618 &size, 8619 &cmdStatus); 8620 if (status == NO_ERROR) { 8621 status = cmdStatus; 8622 } 8623 } 8624 dev = device & AUDIO_DEVICE_IN_ALL; 8625 if (dev) { 8626 status_t cmdStatus; 8627 uint32_t size = sizeof(status_t); 8628 8629 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8630 EFFECT_CMD_SET_INPUT_DEVICE, 8631 sizeof(uint32_t), 8632 &dev, 8633 &size, 8634 &cmdStatus); 8635 if (status2 == NO_ERROR) { 8636 status2 = cmdStatus; 8637 } 8638 if (status == NO_ERROR) { 8639 status = status2; 8640 } 8641 } 8642 } 8643 return status; 8644} 8645 8646status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8647{ 8648 Mutex::Autolock _l(mLock); 8649 status_t status = NO_ERROR; 8650 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8651 status_t cmdStatus; 8652 uint32_t size = sizeof(status_t); 8653 status = (*mEffectInterface)->command(mEffectInterface, 8654 EFFECT_CMD_SET_AUDIO_MODE, 8655 sizeof(audio_mode_t), 8656 &mode, 8657 &size, 8658 &cmdStatus); 8659 if (status == NO_ERROR) { 8660 status = cmdStatus; 8661 } 8662 } 8663 return status; 8664} 8665 8666void AudioFlinger::EffectModule::setSuspended(bool suspended) 8667{ 8668 Mutex::Autolock _l(mLock); 8669 mSuspended = suspended; 8670} 8671 8672bool AudioFlinger::EffectModule::suspended() const 8673{ 8674 Mutex::Autolock _l(mLock); 8675 return mSuspended; 8676} 8677 8678bool AudioFlinger::EffectModule::purgeHandles() 8679{ 8680 bool enabled = false; 8681 Mutex::Autolock _l(mLock); 8682 for (size_t i = 0; i < mHandles.size(); i++) { 8683 EffectHandle *handle = mHandles[i]; 8684 if (handle != NULL && !handle->destroyed_l()) { 8685 handle->effect().clear(); 8686 if (handle->hasControl()) { 8687 enabled = handle->enabled(); 8688 } 8689 } 8690 } 8691 return enabled; 8692} 8693 8694void AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8695{ 8696 const size_t SIZE = 256; 8697 char buffer[SIZE]; 8698 String8 result; 8699 8700 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8701 result.append(buffer); 8702 8703 bool locked = tryLock(mLock); 8704 // failed to lock - AudioFlinger is probably deadlocked 8705 if (!locked) { 8706 result.append("\t\tCould not lock Fx mutex:\n"); 8707 } 8708 8709 result.append("\t\tSession Status State Engine:\n"); 8710 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8711 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8712 result.append(buffer); 8713 8714 result.append("\t\tDescriptor:\n"); 8715 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8716 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8717 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8718 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8719 result.append(buffer); 8720 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8721 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8722 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8723 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8724 result.append(buffer); 8725 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8726 mDescriptor.apiVersion, 8727 mDescriptor.flags); 8728 result.append(buffer); 8729 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8730 mDescriptor.name); 8731 result.append(buffer); 8732 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8733 mDescriptor.implementor); 8734 result.append(buffer); 8735 8736 result.append("\t\t- Input configuration:\n"); 8737 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8738 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8739 (uint32_t)mConfig.inputCfg.buffer.raw, 8740 mConfig.inputCfg.buffer.frameCount, 8741 mConfig.inputCfg.samplingRate, 8742 mConfig.inputCfg.channels, 8743 mConfig.inputCfg.format); 8744 result.append(buffer); 8745 8746 result.append("\t\t- Output configuration:\n"); 8747 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8748 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8749 (uint32_t)mConfig.outputCfg.buffer.raw, 8750 mConfig.outputCfg.buffer.frameCount, 8751 mConfig.outputCfg.samplingRate, 8752 mConfig.outputCfg.channels, 8753 mConfig.outputCfg.format); 8754 result.append(buffer); 8755 8756 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8757 result.append(buffer); 8758 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8759 for (size_t i = 0; i < mHandles.size(); ++i) { 8760 EffectHandle *handle = mHandles[i]; 8761 if (handle != NULL && !handle->destroyed_l()) { 8762 handle->dump(buffer, SIZE); 8763 result.append(buffer); 8764 } 8765 } 8766 8767 result.append("\n"); 8768 8769 write(fd, result.string(), result.length()); 8770 8771 if (locked) { 8772 mLock.unlock(); 8773 } 8774} 8775 8776// ---------------------------------------------------------------------------- 8777// EffectHandle implementation 8778// ---------------------------------------------------------------------------- 8779 8780#undef LOG_TAG 8781#define LOG_TAG "AudioFlinger::EffectHandle" 8782 8783AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8784 const sp<AudioFlinger::Client>& client, 8785 const sp<IEffectClient>& effectClient, 8786 int32_t priority) 8787 : BnEffect(), 8788 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8789 mPriority(priority), mHasControl(false), mEnabled(false), mDestroyed(false) 8790{ 8791 ALOGV("constructor %p", this); 8792 8793 if (client == 0) { 8794 return; 8795 } 8796 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8797 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8798 if (mCblkMemory != 0) { 8799 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8800 8801 if (mCblk != NULL) { 8802 new(mCblk) effect_param_cblk_t(); 8803 mBuffer = (uint8_t *)mCblk + bufOffset; 8804 } 8805 } else { 8806 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8807 return; 8808 } 8809} 8810 8811AudioFlinger::EffectHandle::~EffectHandle() 8812{ 8813 ALOGV("Destructor %p", this); 8814 8815 if (mEffect == 0) { 8816 mDestroyed = true; 8817 return; 8818 } 8819 mEffect->lock(); 8820 mDestroyed = true; 8821 mEffect->unlock(); 8822 disconnect(false); 8823} 8824 8825status_t AudioFlinger::EffectHandle::enable() 8826{ 8827 ALOGV("enable %p", this); 8828 if (!mHasControl) return INVALID_OPERATION; 8829 if (mEffect == 0) return DEAD_OBJECT; 8830 8831 if (mEnabled) { 8832 return NO_ERROR; 8833 } 8834 8835 mEnabled = true; 8836 8837 sp<ThreadBase> thread = mEffect->thread().promote(); 8838 if (thread != 0) { 8839 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8840 } 8841 8842 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8843 if (mEffect->suspended()) { 8844 return NO_ERROR; 8845 } 8846 8847 status_t status = mEffect->setEnabled(true); 8848 if (status != NO_ERROR) { 8849 if (thread != 0) { 8850 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8851 } 8852 mEnabled = false; 8853 } 8854 return status; 8855} 8856 8857status_t AudioFlinger::EffectHandle::disable() 8858{ 8859 ALOGV("disable %p", this); 8860 if (!mHasControl) return INVALID_OPERATION; 8861 if (mEffect == 0) return DEAD_OBJECT; 8862 8863 if (!mEnabled) { 8864 return NO_ERROR; 8865 } 8866 mEnabled = false; 8867 8868 if (mEffect->suspended()) { 8869 return NO_ERROR; 8870 } 8871 8872 status_t status = mEffect->setEnabled(false); 8873 8874 sp<ThreadBase> thread = mEffect->thread().promote(); 8875 if (thread != 0) { 8876 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8877 } 8878 8879 return status; 8880} 8881 8882void AudioFlinger::EffectHandle::disconnect() 8883{ 8884 disconnect(true); 8885} 8886 8887void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8888{ 8889 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8890 if (mEffect == 0) { 8891 return; 8892 } 8893 // restore suspended effects if the disconnected handle was enabled and the last one. 8894 if ((mEffect->disconnect(this, unpinIfLast) == 0) && mEnabled) { 8895 sp<ThreadBase> thread = mEffect->thread().promote(); 8896 if (thread != 0) { 8897 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8898 } 8899 } 8900 8901 // release sp on module => module destructor can be called now 8902 mEffect.clear(); 8903 if (mClient != 0) { 8904 if (mCblk != NULL) { 8905 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8906 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8907 } 8908 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8909 // Client destructor must run with AudioFlinger mutex locked 8910 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8911 mClient.clear(); 8912 } 8913} 8914 8915status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8916 uint32_t cmdSize, 8917 void *pCmdData, 8918 uint32_t *replySize, 8919 void *pReplyData) 8920{ 8921// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8922// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8923 8924 // only get parameter command is permitted for applications not controlling the effect 8925 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8926 return INVALID_OPERATION; 8927 } 8928 if (mEffect == 0) return DEAD_OBJECT; 8929 if (mClient == 0) return INVALID_OPERATION; 8930 8931 // handle commands that are not forwarded transparently to effect engine 8932 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8933 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8934 // no risk to block the whole media server process or mixer threads is we are stuck here 8935 Mutex::Autolock _l(mCblk->lock); 8936 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8937 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8938 mCblk->serverIndex = 0; 8939 mCblk->clientIndex = 0; 8940 return BAD_VALUE; 8941 } 8942 status_t status = NO_ERROR; 8943 while (mCblk->serverIndex < mCblk->clientIndex) { 8944 int reply; 8945 uint32_t rsize = sizeof(int); 8946 int *p = (int *)(mBuffer + mCblk->serverIndex); 8947 int size = *p++; 8948 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8949 ALOGW("command(): invalid parameter block size"); 8950 break; 8951 } 8952 effect_param_t *param = (effect_param_t *)p; 8953 if (param->psize == 0 || param->vsize == 0) { 8954 ALOGW("command(): null parameter or value size"); 8955 mCblk->serverIndex += size; 8956 continue; 8957 } 8958 uint32_t psize = sizeof(effect_param_t) + 8959 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8960 param->vsize; 8961 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8962 psize, 8963 p, 8964 &rsize, 8965 &reply); 8966 // stop at first error encountered 8967 if (ret != NO_ERROR) { 8968 status = ret; 8969 *(int *)pReplyData = reply; 8970 break; 8971 } else if (reply != NO_ERROR) { 8972 *(int *)pReplyData = reply; 8973 break; 8974 } 8975 mCblk->serverIndex += size; 8976 } 8977 mCblk->serverIndex = 0; 8978 mCblk->clientIndex = 0; 8979 return status; 8980 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8981 *(int *)pReplyData = NO_ERROR; 8982 return enable(); 8983 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8984 *(int *)pReplyData = NO_ERROR; 8985 return disable(); 8986 } 8987 8988 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8989} 8990 8991void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8992{ 8993 ALOGV("setControl %p control %d", this, hasControl); 8994 8995 mHasControl = hasControl; 8996 mEnabled = enabled; 8997 8998 if (signal && mEffectClient != 0) { 8999 mEffectClient->controlStatusChanged(hasControl); 9000 } 9001} 9002 9003void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 9004 uint32_t cmdSize, 9005 void *pCmdData, 9006 uint32_t replySize, 9007 void *pReplyData) 9008{ 9009 if (mEffectClient != 0) { 9010 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 9011 } 9012} 9013 9014 9015 9016void AudioFlinger::EffectHandle::setEnabled(bool enabled) 9017{ 9018 if (mEffectClient != 0) { 9019 mEffectClient->enableStatusChanged(enabled); 9020 } 9021} 9022 9023status_t AudioFlinger::EffectHandle::onTransact( 9024 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9025{ 9026 return BnEffect::onTransact(code, data, reply, flags); 9027} 9028 9029 9030void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 9031{ 9032 bool locked = mCblk != NULL && tryLock(mCblk->lock); 9033 9034 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 9035 (mClient == 0) ? getpid_cached : mClient->pid(), 9036 mPriority, 9037 mHasControl, 9038 !locked, 9039 mCblk ? mCblk->clientIndex : 0, 9040 mCblk ? mCblk->serverIndex : 0 9041 ); 9042 9043 if (locked) { 9044 mCblk->lock.unlock(); 9045 } 9046} 9047 9048#undef LOG_TAG 9049#define LOG_TAG "AudioFlinger::EffectChain" 9050 9051AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 9052 int sessionId) 9053 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 9054 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 9055 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 9056{ 9057 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 9058 if (thread == NULL) { 9059 return; 9060 } 9061 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 9062 thread->frameCount(); 9063} 9064 9065AudioFlinger::EffectChain::~EffectChain() 9066{ 9067 if (mOwnInBuffer) { 9068 delete mInBuffer; 9069 } 9070 9071} 9072 9073// getEffectFromDesc_l() must be called with ThreadBase::mLock held 9074sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 9075{ 9076 size_t size = mEffects.size(); 9077 9078 for (size_t i = 0; i < size; i++) { 9079 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 9080 return mEffects[i]; 9081 } 9082 } 9083 return 0; 9084} 9085 9086// getEffectFromId_l() must be called with ThreadBase::mLock held 9087sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 9088{ 9089 size_t size = mEffects.size(); 9090 9091 for (size_t i = 0; i < size; i++) { 9092 // by convention, return first effect if id provided is 0 (0 is never a valid id) 9093 if (id == 0 || mEffects[i]->id() == id) { 9094 return mEffects[i]; 9095 } 9096 } 9097 return 0; 9098} 9099 9100// getEffectFromType_l() must be called with ThreadBase::mLock held 9101sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 9102 const effect_uuid_t *type) 9103{ 9104 size_t size = mEffects.size(); 9105 9106 for (size_t i = 0; i < size; i++) { 9107 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 9108 return mEffects[i]; 9109 } 9110 } 9111 return 0; 9112} 9113 9114void AudioFlinger::EffectChain::clearInputBuffer() 9115{ 9116 Mutex::Autolock _l(mLock); 9117 sp<ThreadBase> thread = mThread.promote(); 9118 if (thread == 0) { 9119 ALOGW("clearInputBuffer(): cannot promote mixer thread"); 9120 return; 9121 } 9122 clearInputBuffer_l(thread); 9123} 9124 9125// Must be called with EffectChain::mLock locked 9126void AudioFlinger::EffectChain::clearInputBuffer_l(sp<ThreadBase> thread) 9127{ 9128 size_t numSamples = thread->frameCount() * thread->channelCount(); 9129 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 9130 9131} 9132 9133// Must be called with EffectChain::mLock locked 9134void AudioFlinger::EffectChain::process_l() 9135{ 9136 sp<ThreadBase> thread = mThread.promote(); 9137 if (thread == 0) { 9138 ALOGW("process_l(): cannot promote mixer thread"); 9139 return; 9140 } 9141 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 9142 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 9143 // always process effects unless no more tracks are on the session and the effect tail 9144 // has been rendered 9145 bool doProcess = true; 9146 if (!isGlobalSession) { 9147 bool tracksOnSession = (trackCnt() != 0); 9148 9149 if (!tracksOnSession && mTailBufferCount == 0) { 9150 doProcess = false; 9151 } 9152 9153 if (activeTrackCnt() == 0) { 9154 // if no track is active and the effect tail has not been rendered, 9155 // the input buffer must be cleared here as the mixer process will not do it 9156 if (tracksOnSession || mTailBufferCount > 0) { 9157 clearInputBuffer_l(thread); 9158 if (mTailBufferCount > 0) { 9159 mTailBufferCount--; 9160 } 9161 } 9162 } 9163 } 9164 9165 size_t size = mEffects.size(); 9166 if (doProcess) { 9167 for (size_t i = 0; i < size; i++) { 9168 mEffects[i]->process(); 9169 } 9170 } 9171 for (size_t i = 0; i < size; i++) { 9172 mEffects[i]->updateState(); 9173 } 9174} 9175 9176// addEffect_l() must be called with PlaybackThread::mLock held 9177status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 9178{ 9179 effect_descriptor_t desc = effect->desc(); 9180 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 9181 9182 Mutex::Autolock _l(mLock); 9183 effect->setChain(this); 9184 sp<ThreadBase> thread = mThread.promote(); 9185 if (thread == 0) { 9186 return NO_INIT; 9187 } 9188 effect->setThread(thread); 9189 9190 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 9191 // Auxiliary effects are inserted at the beginning of mEffects vector as 9192 // they are processed first and accumulated in chain input buffer 9193 mEffects.insertAt(effect, 0); 9194 9195 // the input buffer for auxiliary effect contains mono samples in 9196 // 32 bit format. This is to avoid saturation in AudoMixer 9197 // accumulation stage. Saturation is done in EffectModule::process() before 9198 // calling the process in effect engine 9199 size_t numSamples = thread->frameCount(); 9200 int32_t *buffer = new int32_t[numSamples]; 9201 memset(buffer, 0, numSamples * sizeof(int32_t)); 9202 effect->setInBuffer((int16_t *)buffer); 9203 // auxiliary effects output samples to chain input buffer for further processing 9204 // by insert effects 9205 effect->setOutBuffer(mInBuffer); 9206 } else { 9207 // Insert effects are inserted at the end of mEffects vector as they are processed 9208 // after track and auxiliary effects. 9209 // Insert effect order as a function of indicated preference: 9210 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 9211 // another effect is present 9212 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 9213 // last effect claiming first position 9214 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 9215 // first effect claiming last position 9216 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 9217 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 9218 // already present 9219 9220 size_t size = mEffects.size(); 9221 size_t idx_insert = size; 9222 ssize_t idx_insert_first = -1; 9223 ssize_t idx_insert_last = -1; 9224 9225 for (size_t i = 0; i < size; i++) { 9226 effect_descriptor_t d = mEffects[i]->desc(); 9227 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 9228 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 9229 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 9230 // check invalid effect chaining combinations 9231 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 9232 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 9233 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 9234 return INVALID_OPERATION; 9235 } 9236 // remember position of first insert effect and by default 9237 // select this as insert position for new effect 9238 if (idx_insert == size) { 9239 idx_insert = i; 9240 } 9241 // remember position of last insert effect claiming 9242 // first position 9243 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9244 idx_insert_first = i; 9245 } 9246 // remember position of first insert effect claiming 9247 // last position 9248 if (iPref == EFFECT_FLAG_INSERT_LAST && 9249 idx_insert_last == -1) { 9250 idx_insert_last = i; 9251 } 9252 } 9253 } 9254 9255 // modify idx_insert from first position if needed 9256 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9257 if (idx_insert_last != -1) { 9258 idx_insert = idx_insert_last; 9259 } else { 9260 idx_insert = size; 9261 } 9262 } else { 9263 if (idx_insert_first != -1) { 9264 idx_insert = idx_insert_first + 1; 9265 } 9266 } 9267 9268 // always read samples from chain input buffer 9269 effect->setInBuffer(mInBuffer); 9270 9271 // if last effect in the chain, output samples to chain 9272 // output buffer, otherwise to chain input buffer 9273 if (idx_insert == size) { 9274 if (idx_insert != 0) { 9275 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9276 mEffects[idx_insert-1]->configure(); 9277 } 9278 effect->setOutBuffer(mOutBuffer); 9279 } else { 9280 effect->setOutBuffer(mInBuffer); 9281 } 9282 mEffects.insertAt(effect, idx_insert); 9283 9284 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9285 } 9286 effect->configure(); 9287 return NO_ERROR; 9288} 9289 9290// removeEffect_l() must be called with PlaybackThread::mLock held 9291size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9292{ 9293 Mutex::Autolock _l(mLock); 9294 size_t size = mEffects.size(); 9295 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9296 9297 for (size_t i = 0; i < size; i++) { 9298 if (effect == mEffects[i]) { 9299 // calling stop here will remove pre-processing effect from the audio HAL. 9300 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9301 // the middle of a read from audio HAL 9302 if (mEffects[i]->state() == EffectModule::ACTIVE || 9303 mEffects[i]->state() == EffectModule::STOPPING) { 9304 mEffects[i]->stop(); 9305 } 9306 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9307 delete[] effect->inBuffer(); 9308 } else { 9309 if (i == size - 1 && i != 0) { 9310 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9311 mEffects[i - 1]->configure(); 9312 } 9313 } 9314 mEffects.removeAt(i); 9315 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9316 break; 9317 } 9318 } 9319 9320 return mEffects.size(); 9321} 9322 9323// setDevice_l() must be called with PlaybackThread::mLock held 9324void AudioFlinger::EffectChain::setDevice_l(audio_devices_t device) 9325{ 9326 size_t size = mEffects.size(); 9327 for (size_t i = 0; i < size; i++) { 9328 mEffects[i]->setDevice(device); 9329 } 9330} 9331 9332// setMode_l() must be called with PlaybackThread::mLock held 9333void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9334{ 9335 size_t size = mEffects.size(); 9336 for (size_t i = 0; i < size; i++) { 9337 mEffects[i]->setMode(mode); 9338 } 9339} 9340 9341// setVolume_l() must be called with PlaybackThread::mLock held 9342bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9343{ 9344 uint32_t newLeft = *left; 9345 uint32_t newRight = *right; 9346 bool hasControl = false; 9347 int ctrlIdx = -1; 9348 size_t size = mEffects.size(); 9349 9350 // first update volume controller 9351 for (size_t i = size; i > 0; i--) { 9352 if (mEffects[i - 1]->isProcessEnabled() && 9353 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9354 ctrlIdx = i - 1; 9355 hasControl = true; 9356 break; 9357 } 9358 } 9359 9360 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9361 if (hasControl) { 9362 *left = mNewLeftVolume; 9363 *right = mNewRightVolume; 9364 } 9365 return hasControl; 9366 } 9367 9368 mVolumeCtrlIdx = ctrlIdx; 9369 mLeftVolume = newLeft; 9370 mRightVolume = newRight; 9371 9372 // second get volume update from volume controller 9373 if (ctrlIdx >= 0) { 9374 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9375 mNewLeftVolume = newLeft; 9376 mNewRightVolume = newRight; 9377 } 9378 // then indicate volume to all other effects in chain. 9379 // Pass altered volume to effects before volume controller 9380 // and requested volume to effects after controller 9381 uint32_t lVol = newLeft; 9382 uint32_t rVol = newRight; 9383 9384 for (size_t i = 0; i < size; i++) { 9385 if ((int)i == ctrlIdx) continue; 9386 // this also works for ctrlIdx == -1 when there is no volume controller 9387 if ((int)i > ctrlIdx) { 9388 lVol = *left; 9389 rVol = *right; 9390 } 9391 mEffects[i]->setVolume(&lVol, &rVol, false); 9392 } 9393 *left = newLeft; 9394 *right = newRight; 9395 9396 return hasControl; 9397} 9398 9399void AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9400{ 9401 const size_t SIZE = 256; 9402 char buffer[SIZE]; 9403 String8 result; 9404 9405 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9406 result.append(buffer); 9407 9408 bool locked = tryLock(mLock); 9409 // failed to lock - AudioFlinger is probably deadlocked 9410 if (!locked) { 9411 result.append("\tCould not lock mutex:\n"); 9412 } 9413 9414 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9415 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9416 mEffects.size(), 9417 (uint32_t)mInBuffer, 9418 (uint32_t)mOutBuffer, 9419 mActiveTrackCnt); 9420 result.append(buffer); 9421 write(fd, result.string(), result.size()); 9422 9423 for (size_t i = 0; i < mEffects.size(); ++i) { 9424 sp<EffectModule> effect = mEffects[i]; 9425 if (effect != 0) { 9426 effect->dump(fd, args); 9427 } 9428 } 9429 9430 if (locked) { 9431 mLock.unlock(); 9432 } 9433} 9434 9435// must be called with ThreadBase::mLock held 9436void AudioFlinger::EffectChain::setEffectSuspended_l( 9437 const effect_uuid_t *type, bool suspend) 9438{ 9439 sp<SuspendedEffectDesc> desc; 9440 // use effect type UUID timelow as key as there is no real risk of identical 9441 // timeLow fields among effect type UUIDs. 9442 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9443 if (suspend) { 9444 if (index >= 0) { 9445 desc = mSuspendedEffects.valueAt(index); 9446 } else { 9447 desc = new SuspendedEffectDesc(); 9448 desc->mType = *type; 9449 mSuspendedEffects.add(type->timeLow, desc); 9450 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9451 } 9452 if (desc->mRefCount++ == 0) { 9453 sp<EffectModule> effect = getEffectIfEnabled(type); 9454 if (effect != 0) { 9455 desc->mEffect = effect; 9456 effect->setSuspended(true); 9457 effect->setEnabled(false); 9458 } 9459 } 9460 } else { 9461 if (index < 0) { 9462 return; 9463 } 9464 desc = mSuspendedEffects.valueAt(index); 9465 if (desc->mRefCount <= 0) { 9466 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9467 desc->mRefCount = 1; 9468 } 9469 if (--desc->mRefCount == 0) { 9470 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9471 if (desc->mEffect != 0) { 9472 sp<EffectModule> effect = desc->mEffect.promote(); 9473 if (effect != 0) { 9474 effect->setSuspended(false); 9475 effect->lock(); 9476 EffectHandle *handle = effect->controlHandle_l(); 9477 if (handle != NULL && !handle->destroyed_l()) { 9478 effect->setEnabled_l(handle->enabled()); 9479 } 9480 effect->unlock(); 9481 } 9482 desc->mEffect.clear(); 9483 } 9484 mSuspendedEffects.removeItemsAt(index); 9485 } 9486 } 9487} 9488 9489// must be called with ThreadBase::mLock held 9490void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9491{ 9492 sp<SuspendedEffectDesc> desc; 9493 9494 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9495 if (suspend) { 9496 if (index >= 0) { 9497 desc = mSuspendedEffects.valueAt(index); 9498 } else { 9499 desc = new SuspendedEffectDesc(); 9500 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9501 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9502 } 9503 if (desc->mRefCount++ == 0) { 9504 Vector< sp<EffectModule> > effects; 9505 getSuspendEligibleEffects(effects); 9506 for (size_t i = 0; i < effects.size(); i++) { 9507 setEffectSuspended_l(&effects[i]->desc().type, true); 9508 } 9509 } 9510 } else { 9511 if (index < 0) { 9512 return; 9513 } 9514 desc = mSuspendedEffects.valueAt(index); 9515 if (desc->mRefCount <= 0) { 9516 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9517 desc->mRefCount = 1; 9518 } 9519 if (--desc->mRefCount == 0) { 9520 Vector<const effect_uuid_t *> types; 9521 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9522 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9523 continue; 9524 } 9525 types.add(&mSuspendedEffects.valueAt(i)->mType); 9526 } 9527 for (size_t i = 0; i < types.size(); i++) { 9528 setEffectSuspended_l(types[i], false); 9529 } 9530 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9531 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9532 } 9533 } 9534} 9535 9536 9537// The volume effect is used for automated tests only 9538#ifndef OPENSL_ES_H_ 9539static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9540 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9541const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9542#endif //OPENSL_ES_H_ 9543 9544bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9545{ 9546 // auxiliary effects and visualizer are never suspended on output mix 9547 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9548 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9549 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9550 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9551 return false; 9552 } 9553 return true; 9554} 9555 9556void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9557{ 9558 effects.clear(); 9559 for (size_t i = 0; i < mEffects.size(); i++) { 9560 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9561 effects.add(mEffects[i]); 9562 } 9563 } 9564} 9565 9566sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9567 const effect_uuid_t *type) 9568{ 9569 sp<EffectModule> effect = getEffectFromType_l(type); 9570 return effect != 0 && effect->isEnabled() ? effect : 0; 9571} 9572 9573void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9574 bool enabled) 9575{ 9576 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9577 if (enabled) { 9578 if (index < 0) { 9579 // if the effect is not suspend check if all effects are suspended 9580 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9581 if (index < 0) { 9582 return; 9583 } 9584 if (!isEffectEligibleForSuspend(effect->desc())) { 9585 return; 9586 } 9587 setEffectSuspended_l(&effect->desc().type, enabled); 9588 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9589 if (index < 0) { 9590 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9591 return; 9592 } 9593 } 9594 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9595 effect->desc().type.timeLow); 9596 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9597 // if effect is requested to suspended but was not yet enabled, supend it now. 9598 if (desc->mEffect == 0) { 9599 desc->mEffect = effect; 9600 effect->setEnabled(false); 9601 effect->setSuspended(true); 9602 } 9603 } else { 9604 if (index < 0) { 9605 return; 9606 } 9607 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9608 effect->desc().type.timeLow); 9609 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9610 desc->mEffect.clear(); 9611 effect->setSuspended(false); 9612 } 9613} 9614 9615#undef LOG_TAG 9616#define LOG_TAG "AudioFlinger" 9617 9618// ---------------------------------------------------------------------------- 9619 9620status_t AudioFlinger::onTransact( 9621 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9622{ 9623 return BnAudioFlinger::onTransact(code, data, reply, flags); 9624} 9625 9626}; // namespace android 9627