AudioFlinger.cpp revision 99c99d00beb43b939dedc9ffb07adb89f6a85ba5
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 "SourceAudioBufferProvider.h" 83 84#ifdef HAVE_REQUEST_PRIORITY 85#include "SchedulingPolicyService.h" 86#endif 87 88#ifdef SOAKER 89#include "Soaker.h" 90#endif 91 92// ---------------------------------------------------------------------------- 93 94// Note: the following macro is used for extremely verbose logging message. In 95// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 96// 0; but one side effect of this is to turn all LOGV's as well. Some messages 97// are so verbose that we want to suppress them even when we have ALOG_ASSERT 98// turned on. Do not uncomment the #def below unless you really know what you 99// are doing and want to see all of the extremely verbose messages. 100//#define VERY_VERY_VERBOSE_LOGGING 101#ifdef VERY_VERY_VERBOSE_LOGGING 102#define ALOGVV ALOGV 103#else 104#define ALOGVV(a...) do { } while(0) 105#endif 106 107namespace android { 108 109static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 110static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 111 112static const float MAX_GAIN = 4096.0f; 113static const uint32_t MAX_GAIN_INT = 0x1000; 114 115// retry counts for buffer fill timeout 116// 50 * ~20msecs = 1 second 117static const int8_t kMaxTrackRetries = 50; 118static const int8_t kMaxTrackStartupRetries = 50; 119// allow less retry attempts on direct output thread. 120// direct outputs can be a scarce resource in audio hardware and should 121// be released as quickly as possible. 122static const int8_t kMaxTrackRetriesDirect = 2; 123 124static const int kDumpLockRetries = 50; 125static const int kDumpLockSleepUs = 20000; 126 127// don't warn about blocked writes or record buffer overflows more often than this 128static const nsecs_t kWarningThrottleNs = seconds(5); 129 130// RecordThread loop sleep time upon application overrun or audio HAL read error 131static const int kRecordThreadSleepUs = 5000; 132 133// maximum time to wait for setParameters to complete 134static const nsecs_t kSetParametersTimeoutNs = seconds(2); 135 136// minimum sleep time for the mixer thread loop when tracks are active but in underrun 137static const uint32_t kMinThreadSleepTimeUs = 5000; 138// maximum divider applied to the active sleep time in the mixer thread loop 139static const uint32_t kMaxThreadSleepTimeShift = 2; 140 141// minimum normal mix buffer size, expressed in milliseconds rather than frames 142static const uint32_t kMinNormalMixBufferSizeMs = 20; 143// maximum normal mix buffer size 144static const uint32_t kMaxNormalMixBufferSizeMs = 24; 145 146nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 147 148// Whether to use fast mixer 149static const enum { 150 FastMixer_Never, // never initialize or use: for debugging only 151 FastMixer_Always, // always initialize and use, even if not needed: for debugging only 152 // normal mixer multiplier is 1 153 FastMixer_Static, // initialize if needed, then use all the time if initialized, 154 // multiplier is calculated based on min & max normal mixer buffer size 155 FastMixer_Dynamic, // initialize if needed, then use dynamically depending on track load, 156 // multiplier is calculated based on min & max normal mixer buffer size 157 // FIXME for FastMixer_Dynamic: 158 // Supporting this option will require fixing HALs that can't handle large writes. 159 // For example, one HAL implementation returns an error from a large write, 160 // and another HAL implementation corrupts memory, possibly in the sample rate converter. 161 // We could either fix the HAL implementations, or provide a wrapper that breaks 162 // up large writes into smaller ones, and the wrapper would need to deal with scheduler. 163} kUseFastMixer = FastMixer_Static; 164 165// ---------------------------------------------------------------------------- 166 167#ifdef ADD_BATTERY_DATA 168// To collect the amplifier usage 169static void addBatteryData(uint32_t params) { 170 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 171 if (service == NULL) { 172 // it already logged 173 return; 174 } 175 176 service->addBatteryData(params); 177} 178#endif 179 180static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 181{ 182 const hw_module_t *mod; 183 int rc; 184 185 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 186 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 187 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 188 if (rc) { 189 goto out; 190 } 191 rc = audio_hw_device_open(mod, dev); 192 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 193 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 194 if (rc) { 195 goto out; 196 } 197 if ((*dev)->common.version != AUDIO_DEVICE_API_VERSION_CURRENT) { 198 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 199 rc = BAD_VALUE; 200 goto out; 201 } 202 return 0; 203 204out: 205 *dev = NULL; 206 return rc; 207} 208 209// ---------------------------------------------------------------------------- 210 211AudioFlinger::AudioFlinger() 212 : BnAudioFlinger(), 213 mPrimaryHardwareDev(NULL), 214 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 215 mMasterVolume(1.0f), 216 mMasterVolumeSupportLvl(MVS_NONE), 217 mMasterMute(false), 218 mNextUniqueId(1), 219 mMode(AUDIO_MODE_INVALID), 220 mBtNrecIsOff(false) 221{ 222} 223 224void AudioFlinger::onFirstRef() 225{ 226 int rc = 0; 227 228 Mutex::Autolock _l(mLock); 229 230 /* TODO: move all this work into an Init() function */ 231 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 232 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 233 uint32_t int_val; 234 if (1 == sscanf(val_str, "%u", &int_val)) { 235 mStandbyTimeInNsecs = milliseconds(int_val); 236 ALOGI("Using %u mSec as standby time.", int_val); 237 } else { 238 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 239 ALOGI("Using default %u mSec as standby time.", 240 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 241 } 242 } 243 244 mMode = AUDIO_MODE_NORMAL; 245 mMasterVolumeSW = 1.0; 246 mMasterVolume = 1.0; 247 mHardwareStatus = AUDIO_HW_IDLE; 248} 249 250AudioFlinger::~AudioFlinger() 251{ 252 253 while (!mRecordThreads.isEmpty()) { 254 // closeInput() will remove first entry from mRecordThreads 255 closeInput(mRecordThreads.keyAt(0)); 256 } 257 while (!mPlaybackThreads.isEmpty()) { 258 // closeOutput() will remove first entry from mPlaybackThreads 259 closeOutput(mPlaybackThreads.keyAt(0)); 260 } 261 262 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 263 // no mHardwareLock needed, as there are no other references to this 264 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 265 delete mAudioHwDevs.valueAt(i); 266 } 267} 268 269static const char * const audio_interfaces[] = { 270 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 271 AUDIO_HARDWARE_MODULE_ID_A2DP, 272 AUDIO_HARDWARE_MODULE_ID_USB, 273}; 274#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 275 276audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 277{ 278 // if module is 0, the request comes from an old policy manager and we should load 279 // well known modules 280 if (module == 0) { 281 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 282 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 283 loadHwModule_l(audio_interfaces[i]); 284 } 285 } else { 286 // check a match for the requested module handle 287 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 288 if (audioHwdevice != NULL) { 289 return audioHwdevice->hwDevice(); 290 } 291 } 292 // then try to find a module supporting the requested device. 293 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 294 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 295 if ((dev->get_supported_devices(dev) & devices) == devices) 296 return dev; 297 } 298 299 return NULL; 300} 301 302status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 303{ 304 const size_t SIZE = 256; 305 char buffer[SIZE]; 306 String8 result; 307 308 result.append("Clients:\n"); 309 for (size_t i = 0; i < mClients.size(); ++i) { 310 sp<Client> client = mClients.valueAt(i).promote(); 311 if (client != 0) { 312 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 313 result.append(buffer); 314 } 315 } 316 317 result.append("Global session refs:\n"); 318 result.append(" session pid count\n"); 319 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 320 AudioSessionRef *r = mAudioSessionRefs[i]; 321 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 322 result.append(buffer); 323 } 324 write(fd, result.string(), result.size()); 325 return NO_ERROR; 326} 327 328 329status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 330{ 331 const size_t SIZE = 256; 332 char buffer[SIZE]; 333 String8 result; 334 hardware_call_state hardwareStatus = mHardwareStatus; 335 336 snprintf(buffer, SIZE, "Hardware status: %d\n" 337 "Standby Time mSec: %u\n", 338 hardwareStatus, 339 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 340 result.append(buffer); 341 write(fd, result.string(), result.size()); 342 return NO_ERROR; 343} 344 345status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 346{ 347 const size_t SIZE = 256; 348 char buffer[SIZE]; 349 String8 result; 350 snprintf(buffer, SIZE, "Permission Denial: " 351 "can't dump AudioFlinger from pid=%d, uid=%d\n", 352 IPCThreadState::self()->getCallingPid(), 353 IPCThreadState::self()->getCallingUid()); 354 result.append(buffer); 355 write(fd, result.string(), result.size()); 356 return NO_ERROR; 357} 358 359static bool tryLock(Mutex& mutex) 360{ 361 bool locked = false; 362 for (int i = 0; i < kDumpLockRetries; ++i) { 363 if (mutex.tryLock() == NO_ERROR) { 364 locked = true; 365 break; 366 } 367 usleep(kDumpLockSleepUs); 368 } 369 return locked; 370} 371 372status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 373{ 374 if (!dumpAllowed()) { 375 dumpPermissionDenial(fd, args); 376 } else { 377 // get state of hardware lock 378 bool hardwareLocked = tryLock(mHardwareLock); 379 if (!hardwareLocked) { 380 String8 result(kHardwareLockedString); 381 write(fd, result.string(), result.size()); 382 } else { 383 mHardwareLock.unlock(); 384 } 385 386 bool locked = tryLock(mLock); 387 388 // failed to lock - AudioFlinger is probably deadlocked 389 if (!locked) { 390 String8 result(kDeadlockedString); 391 write(fd, result.string(), result.size()); 392 } 393 394 dumpClients(fd, args); 395 dumpInternals(fd, args); 396 397 // dump playback threads 398 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 399 mPlaybackThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump record threads 403 for (size_t i = 0; i < mRecordThreads.size(); i++) { 404 mRecordThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump all hardware devs 408 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 409 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 410 dev->dump(dev, fd); 411 } 412 if (locked) mLock.unlock(); 413 } 414 return NO_ERROR; 415} 416 417sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 418{ 419 // If pid is already in the mClients wp<> map, then use that entry 420 // (for which promote() is always != 0), otherwise create a new entry and Client. 421 sp<Client> client = mClients.valueFor(pid).promote(); 422 if (client == 0) { 423 client = new Client(this, pid); 424 mClients.add(pid, client); 425 } 426 427 return client; 428} 429 430// IAudioFlinger interface 431 432 433sp<IAudioTrack> AudioFlinger::createTrack( 434 pid_t pid, 435 audio_stream_type_t streamType, 436 uint32_t sampleRate, 437 audio_format_t format, 438 uint32_t channelMask, 439 int frameCount, 440 IAudioFlinger::track_flags_t flags, 441 const sp<IMemory>& sharedBuffer, 442 audio_io_handle_t output, 443 pid_t tid, 444 int *sessionId, 445 status_t *status) 446{ 447 sp<PlaybackThread::Track> track; 448 sp<TrackHandle> trackHandle; 449 sp<Client> client; 450 status_t lStatus; 451 int lSessionId; 452 453 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 454 // but if someone uses binder directly they could bypass that and cause us to crash 455 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 456 ALOGE("createTrack() invalid stream type %d", streamType); 457 lStatus = BAD_VALUE; 458 goto Exit; 459 } 460 461 { 462 Mutex::Autolock _l(mLock); 463 PlaybackThread *thread = checkPlaybackThread_l(output); 464 PlaybackThread *effectThread = NULL; 465 if (thread == NULL) { 466 ALOGE("unknown output thread"); 467 lStatus = BAD_VALUE; 468 goto Exit; 469 } 470 471 client = registerPid_l(pid); 472 473 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 474 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 475 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 476 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 477 if (mPlaybackThreads.keyAt(i) != output) { 478 // prevent same audio session on different output threads 479 uint32_t sessions = t->hasAudioSession(*sessionId); 480 if (sessions & PlaybackThread::TRACK_SESSION) { 481 ALOGE("createTrack() session ID %d already in use", *sessionId); 482 lStatus = BAD_VALUE; 483 goto Exit; 484 } 485 // check if an effect with same session ID is waiting for a track to be created 486 if (sessions & PlaybackThread::EFFECT_SESSION) { 487 effectThread = t.get(); 488 } 489 } 490 } 491 lSessionId = *sessionId; 492 } else { 493 // if no audio session id is provided, create one here 494 lSessionId = nextUniqueId(); 495 if (sessionId != NULL) { 496 *sessionId = lSessionId; 497 } 498 } 499 ALOGV("createTrack() lSessionId: %d", lSessionId); 500 501 track = thread->createTrack_l(client, streamType, sampleRate, format, 502 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, &lStatus); 503 504 // move effect chain to this output thread if an effect on same session was waiting 505 // for a track to be created 506 if (lStatus == NO_ERROR && effectThread != NULL) { 507 Mutex::Autolock _dl(thread->mLock); 508 Mutex::Autolock _sl(effectThread->mLock); 509 moveEffectChain_l(lSessionId, effectThread, thread, true); 510 } 511 512 // Look for sync events awaiting for a session to be used. 513 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 514 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 515 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 516 if (lStatus == NO_ERROR) { 517 track->setSyncEvent(mPendingSyncEvents[i]); 518 } else { 519 mPendingSyncEvents[i]->cancel(); 520 } 521 mPendingSyncEvents.removeAt(i); 522 i--; 523 } 524 } 525 } 526 } 527 if (lStatus == NO_ERROR) { 528 trackHandle = new TrackHandle(track); 529 } else { 530 // remove local strong reference to Client before deleting the Track so that the Client 531 // destructor is called by the TrackBase destructor with mLock held 532 client.clear(); 533 track.clear(); 534 } 535 536Exit: 537 if (status != NULL) { 538 *status = lStatus; 539 } 540 return trackHandle; 541} 542 543uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 544{ 545 Mutex::Autolock _l(mLock); 546 PlaybackThread *thread = checkPlaybackThread_l(output); 547 if (thread == NULL) { 548 ALOGW("sampleRate() unknown thread %d", output); 549 return 0; 550 } 551 return thread->sampleRate(); 552} 553 554int AudioFlinger::channelCount(audio_io_handle_t output) const 555{ 556 Mutex::Autolock _l(mLock); 557 PlaybackThread *thread = checkPlaybackThread_l(output); 558 if (thread == NULL) { 559 ALOGW("channelCount() unknown thread %d", output); 560 return 0; 561 } 562 return thread->channelCount(); 563} 564 565audio_format_t AudioFlinger::format(audio_io_handle_t output) const 566{ 567 Mutex::Autolock _l(mLock); 568 PlaybackThread *thread = checkPlaybackThread_l(output); 569 if (thread == NULL) { 570 ALOGW("format() unknown thread %d", output); 571 return AUDIO_FORMAT_INVALID; 572 } 573 return thread->format(); 574} 575 576size_t AudioFlinger::frameCount(audio_io_handle_t output) const 577{ 578 Mutex::Autolock _l(mLock); 579 PlaybackThread *thread = checkPlaybackThread_l(output); 580 if (thread == NULL) { 581 ALOGW("frameCount() unknown thread %d", output); 582 return 0; 583 } 584 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 585 // should examine all callers and fix them to handle smaller counts 586 return thread->frameCount(); 587} 588 589uint32_t AudioFlinger::latency(audio_io_handle_t output) const 590{ 591 Mutex::Autolock _l(mLock); 592 PlaybackThread *thread = checkPlaybackThread_l(output); 593 if (thread == NULL) { 594 ALOGW("latency() unknown thread %d", output); 595 return 0; 596 } 597 return thread->latency(); 598} 599 600status_t AudioFlinger::setMasterVolume(float value) 601{ 602 status_t ret = initCheck(); 603 if (ret != NO_ERROR) { 604 return ret; 605 } 606 607 // check calling permissions 608 if (!settingsAllowed()) { 609 return PERMISSION_DENIED; 610 } 611 612 float swmv = value; 613 614 Mutex::Autolock _l(mLock); 615 616 // when hw supports master volume, don't scale in sw mixer 617 if (MVS_NONE != mMasterVolumeSupportLvl) { 618 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 619 AutoMutex lock(mHardwareLock); 620 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 621 622 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 623 if (NULL != dev->set_master_volume) { 624 dev->set_master_volume(dev, value); 625 } 626 mHardwareStatus = AUDIO_HW_IDLE; 627 } 628 629 swmv = 1.0; 630 } 631 632 mMasterVolume = value; 633 mMasterVolumeSW = swmv; 634 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 635 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 636 637 return NO_ERROR; 638} 639 640status_t AudioFlinger::setMode(audio_mode_t mode) 641{ 642 status_t ret = initCheck(); 643 if (ret != NO_ERROR) { 644 return ret; 645 } 646 647 // check calling permissions 648 if (!settingsAllowed()) { 649 return PERMISSION_DENIED; 650 } 651 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 652 ALOGW("Illegal value: setMode(%d)", mode); 653 return BAD_VALUE; 654 } 655 656 { // scope for the lock 657 AutoMutex lock(mHardwareLock); 658 mHardwareStatus = AUDIO_HW_SET_MODE; 659 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 660 mHardwareStatus = AUDIO_HW_IDLE; 661 } 662 663 if (NO_ERROR == ret) { 664 Mutex::Autolock _l(mLock); 665 mMode = mode; 666 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 667 mPlaybackThreads.valueAt(i)->setMode(mode); 668 } 669 670 return ret; 671} 672 673status_t AudioFlinger::setMicMute(bool state) 674{ 675 status_t ret = initCheck(); 676 if (ret != NO_ERROR) { 677 return ret; 678 } 679 680 // check calling permissions 681 if (!settingsAllowed()) { 682 return PERMISSION_DENIED; 683 } 684 685 AutoMutex lock(mHardwareLock); 686 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 687 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 688 mHardwareStatus = AUDIO_HW_IDLE; 689 return ret; 690} 691 692bool AudioFlinger::getMicMute() const 693{ 694 status_t ret = initCheck(); 695 if (ret != NO_ERROR) { 696 return false; 697 } 698 699 bool state = AUDIO_MODE_INVALID; 700 AutoMutex lock(mHardwareLock); 701 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 702 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 703 mHardwareStatus = AUDIO_HW_IDLE; 704 return state; 705} 706 707status_t AudioFlinger::setMasterMute(bool muted) 708{ 709 // check calling permissions 710 if (!settingsAllowed()) { 711 return PERMISSION_DENIED; 712 } 713 714 Mutex::Autolock _l(mLock); 715 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 716 mMasterMute = muted; 717 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 718 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 719 720 return NO_ERROR; 721} 722 723float AudioFlinger::masterVolume() const 724{ 725 Mutex::Autolock _l(mLock); 726 return masterVolume_l(); 727} 728 729float AudioFlinger::masterVolumeSW() const 730{ 731 Mutex::Autolock _l(mLock); 732 return masterVolumeSW_l(); 733} 734 735bool AudioFlinger::masterMute() const 736{ 737 Mutex::Autolock _l(mLock); 738 return masterMute_l(); 739} 740 741float AudioFlinger::masterVolume_l() const 742{ 743 if (MVS_FULL == mMasterVolumeSupportLvl) { 744 float ret_val; 745 AutoMutex lock(mHardwareLock); 746 747 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 748 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 749 (NULL != mPrimaryHardwareDev->get_master_volume), 750 "can't get master volume"); 751 752 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 753 mHardwareStatus = AUDIO_HW_IDLE; 754 return ret_val; 755 } 756 757 return mMasterVolume; 758} 759 760status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 761 audio_io_handle_t output) 762{ 763 // check calling permissions 764 if (!settingsAllowed()) { 765 return PERMISSION_DENIED; 766 } 767 768 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 769 ALOGE("setStreamVolume() invalid stream %d", stream); 770 return BAD_VALUE; 771 } 772 773 AutoMutex lock(mLock); 774 PlaybackThread *thread = NULL; 775 if (output) { 776 thread = checkPlaybackThread_l(output); 777 if (thread == NULL) { 778 return BAD_VALUE; 779 } 780 } 781 782 mStreamTypes[stream].volume = value; 783 784 if (thread == NULL) { 785 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 786 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 787 } 788 } else { 789 thread->setStreamVolume(stream, value); 790 } 791 792 return NO_ERROR; 793} 794 795status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 796{ 797 // check calling permissions 798 if (!settingsAllowed()) { 799 return PERMISSION_DENIED; 800 } 801 802 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 803 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 804 ALOGE("setStreamMute() invalid stream %d", stream); 805 return BAD_VALUE; 806 } 807 808 AutoMutex lock(mLock); 809 mStreamTypes[stream].mute = muted; 810 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 811 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 812 813 return NO_ERROR; 814} 815 816float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 817{ 818 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 819 return 0.0f; 820 } 821 822 AutoMutex lock(mLock); 823 float volume; 824 if (output) { 825 PlaybackThread *thread = checkPlaybackThread_l(output); 826 if (thread == NULL) { 827 return 0.0f; 828 } 829 volume = thread->streamVolume(stream); 830 } else { 831 volume = streamVolume_l(stream); 832 } 833 834 return volume; 835} 836 837bool AudioFlinger::streamMute(audio_stream_type_t stream) const 838{ 839 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 840 return true; 841 } 842 843 AutoMutex lock(mLock); 844 return streamMute_l(stream); 845} 846 847status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 848{ 849 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 850 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 851 // check calling permissions 852 if (!settingsAllowed()) { 853 return PERMISSION_DENIED; 854 } 855 856 // ioHandle == 0 means the parameters are global to the audio hardware interface 857 if (ioHandle == 0) { 858 Mutex::Autolock _l(mLock); 859 status_t final_result = NO_ERROR; 860 { 861 AutoMutex lock(mHardwareLock); 862 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 863 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 864 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 865 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 866 final_result = result ?: final_result; 867 } 868 mHardwareStatus = AUDIO_HW_IDLE; 869 } 870 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 871 AudioParameter param = AudioParameter(keyValuePairs); 872 String8 value; 873 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 874 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 875 if (mBtNrecIsOff != btNrecIsOff) { 876 for (size_t i = 0; i < mRecordThreads.size(); i++) { 877 sp<RecordThread> thread = mRecordThreads.valueAt(i); 878 RecordThread::RecordTrack *track = thread->track(); 879 if (track != NULL) { 880 audio_devices_t device = (audio_devices_t)( 881 thread->device() & AUDIO_DEVICE_IN_ALL); 882 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 883 thread->setEffectSuspended(FX_IID_AEC, 884 suspend, 885 track->sessionId()); 886 thread->setEffectSuspended(FX_IID_NS, 887 suspend, 888 track->sessionId()); 889 } 890 } 891 mBtNrecIsOff = btNrecIsOff; 892 } 893 } 894 return final_result; 895 } 896 897 // hold a strong ref on thread in case closeOutput() or closeInput() is called 898 // and the thread is exited once the lock is released 899 sp<ThreadBase> thread; 900 { 901 Mutex::Autolock _l(mLock); 902 thread = checkPlaybackThread_l(ioHandle); 903 if (thread == NULL) { 904 thread = checkRecordThread_l(ioHandle); 905 } else if (thread == primaryPlaybackThread_l()) { 906 // indicate output device change to all input threads for pre processing 907 AudioParameter param = AudioParameter(keyValuePairs); 908 int value; 909 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 910 (value != 0)) { 911 for (size_t i = 0; i < mRecordThreads.size(); i++) { 912 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 913 } 914 } 915 } 916 } 917 if (thread != 0) { 918 return thread->setParameters(keyValuePairs); 919 } 920 return BAD_VALUE; 921} 922 923String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 924{ 925// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 926// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 927 928 Mutex::Autolock _l(mLock); 929 930 if (ioHandle == 0) { 931 String8 out_s8; 932 933 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 934 char *s; 935 { 936 AutoMutex lock(mHardwareLock); 937 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 938 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 939 s = dev->get_parameters(dev, keys.string()); 940 mHardwareStatus = AUDIO_HW_IDLE; 941 } 942 out_s8 += String8(s ? s : ""); 943 free(s); 944 } 945 return out_s8; 946 } 947 948 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 949 if (playbackThread != NULL) { 950 return playbackThread->getParameters(keys); 951 } 952 RecordThread *recordThread = checkRecordThread_l(ioHandle); 953 if (recordThread != NULL) { 954 return recordThread->getParameters(keys); 955 } 956 return String8(""); 957} 958 959size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 960{ 961 status_t ret = initCheck(); 962 if (ret != NO_ERROR) { 963 return 0; 964 } 965 966 AutoMutex lock(mHardwareLock); 967 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 968 struct audio_config config = { 969 sample_rate: sampleRate, 970 channel_mask: audio_channel_in_mask_from_count(channelCount), 971 format: format, 972 }; 973 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, &config); 974 mHardwareStatus = AUDIO_HW_IDLE; 975 return size; 976} 977 978unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 979{ 980 if (ioHandle == 0) { 981 return 0; 982 } 983 984 Mutex::Autolock _l(mLock); 985 986 RecordThread *recordThread = checkRecordThread_l(ioHandle); 987 if (recordThread != NULL) { 988 return recordThread->getInputFramesLost(); 989 } 990 return 0; 991} 992 993status_t AudioFlinger::setVoiceVolume(float value) 994{ 995 status_t ret = initCheck(); 996 if (ret != NO_ERROR) { 997 return ret; 998 } 999 1000 // check calling permissions 1001 if (!settingsAllowed()) { 1002 return PERMISSION_DENIED; 1003 } 1004 1005 AutoMutex lock(mHardwareLock); 1006 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1007 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 1008 mHardwareStatus = AUDIO_HW_IDLE; 1009 1010 return ret; 1011} 1012 1013status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1014 audio_io_handle_t output) const 1015{ 1016 status_t status; 1017 1018 Mutex::Autolock _l(mLock); 1019 1020 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1021 if (playbackThread != NULL) { 1022 return playbackThread->getRenderPosition(halFrames, dspFrames); 1023 } 1024 1025 return BAD_VALUE; 1026} 1027 1028void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1029{ 1030 1031 Mutex::Autolock _l(mLock); 1032 1033 pid_t pid = IPCThreadState::self()->getCallingPid(); 1034 if (mNotificationClients.indexOfKey(pid) < 0) { 1035 sp<NotificationClient> notificationClient = new NotificationClient(this, 1036 client, 1037 pid); 1038 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1039 1040 mNotificationClients.add(pid, notificationClient); 1041 1042 sp<IBinder> binder = client->asBinder(); 1043 binder->linkToDeath(notificationClient); 1044 1045 // the config change is always sent from playback or record threads to avoid deadlock 1046 // with AudioSystem::gLock 1047 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1048 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1049 } 1050 1051 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1052 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1053 } 1054 } 1055} 1056 1057void AudioFlinger::removeNotificationClient(pid_t pid) 1058{ 1059 Mutex::Autolock _l(mLock); 1060 1061 mNotificationClients.removeItem(pid); 1062 1063 ALOGV("%d died, releasing its sessions", pid); 1064 size_t num = mAudioSessionRefs.size(); 1065 bool removed = false; 1066 for (size_t i = 0; i< num; ) { 1067 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1068 ALOGV(" pid %d @ %d", ref->mPid, i); 1069 if (ref->mPid == pid) { 1070 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1071 mAudioSessionRefs.removeAt(i); 1072 delete ref; 1073 removed = true; 1074 num--; 1075 } else { 1076 i++; 1077 } 1078 } 1079 if (removed) { 1080 purgeStaleEffects_l(); 1081 } 1082} 1083 1084// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1085void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1086{ 1087 size_t size = mNotificationClients.size(); 1088 for (size_t i = 0; i < size; i++) { 1089 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1090 param2); 1091 } 1092} 1093 1094// removeClient_l() must be called with AudioFlinger::mLock held 1095void AudioFlinger::removeClient_l(pid_t pid) 1096{ 1097 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1098 mClients.removeItem(pid); 1099} 1100 1101 1102// ---------------------------------------------------------------------------- 1103 1104AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1105 uint32_t device, type_t type) 1106 : Thread(false), 1107 mType(type), 1108 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mNormalFrameCount(0), 1109 // mChannelMask 1110 mChannelCount(0), 1111 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1112 mParamStatus(NO_ERROR), 1113 mStandby(false), mId(id), 1114 mDevice(device), 1115 mDeathRecipient(new PMDeathRecipient(this)) 1116{ 1117} 1118 1119AudioFlinger::ThreadBase::~ThreadBase() 1120{ 1121 mParamCond.broadcast(); 1122 // do not lock the mutex in destructor 1123 releaseWakeLock_l(); 1124 if (mPowerManager != 0) { 1125 sp<IBinder> binder = mPowerManager->asBinder(); 1126 binder->unlinkToDeath(mDeathRecipient); 1127 } 1128} 1129 1130void AudioFlinger::ThreadBase::exit() 1131{ 1132 ALOGV("ThreadBase::exit"); 1133 { 1134 // This lock prevents the following race in thread (uniprocessor for illustration): 1135 // if (!exitPending()) { 1136 // // context switch from here to exit() 1137 // // exit() calls requestExit(), what exitPending() observes 1138 // // exit() calls signal(), which is dropped since no waiters 1139 // // context switch back from exit() to here 1140 // mWaitWorkCV.wait(...); 1141 // // now thread is hung 1142 // } 1143 AutoMutex lock(mLock); 1144 requestExit(); 1145 mWaitWorkCV.signal(); 1146 } 1147 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1148 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1149 requestExitAndWait(); 1150} 1151 1152status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1153{ 1154 status_t status; 1155 1156 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1157 Mutex::Autolock _l(mLock); 1158 1159 mNewParameters.add(keyValuePairs); 1160 mWaitWorkCV.signal(); 1161 // wait condition with timeout in case the thread loop has exited 1162 // before the request could be processed 1163 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1164 status = mParamStatus; 1165 mWaitWorkCV.signal(); 1166 } else { 1167 status = TIMED_OUT; 1168 } 1169 return status; 1170} 1171 1172void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1173{ 1174 Mutex::Autolock _l(mLock); 1175 sendConfigEvent_l(event, param); 1176} 1177 1178// sendConfigEvent_l() must be called with ThreadBase::mLock held 1179void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1180{ 1181 ConfigEvent configEvent; 1182 configEvent.mEvent = event; 1183 configEvent.mParam = param; 1184 mConfigEvents.add(configEvent); 1185 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1186 mWaitWorkCV.signal(); 1187} 1188 1189void AudioFlinger::ThreadBase::processConfigEvents() 1190{ 1191 mLock.lock(); 1192 while (!mConfigEvents.isEmpty()) { 1193 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1194 ConfigEvent configEvent = mConfigEvents[0]; 1195 mConfigEvents.removeAt(0); 1196 // release mLock before locking AudioFlinger mLock: lock order is always 1197 // AudioFlinger then ThreadBase to avoid cross deadlock 1198 mLock.unlock(); 1199 mAudioFlinger->mLock.lock(); 1200 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1201 mAudioFlinger->mLock.unlock(); 1202 mLock.lock(); 1203 } 1204 mLock.unlock(); 1205} 1206 1207status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1208{ 1209 const size_t SIZE = 256; 1210 char buffer[SIZE]; 1211 String8 result; 1212 1213 bool locked = tryLock(mLock); 1214 if (!locked) { 1215 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1216 write(fd, buffer, strlen(buffer)); 1217 } 1218 1219 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1220 result.append(buffer); 1221 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1222 result.append(buffer); 1223 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1224 result.append(buffer); 1225 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1226 result.append(buffer); 1227 snprintf(buffer, SIZE, "HAL frame count: %d\n", mFrameCount); 1228 result.append(buffer); 1229 snprintf(buffer, SIZE, "Normal frame count: %d\n", mNormalFrameCount); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1232 result.append(buffer); 1233 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1234 result.append(buffer); 1235 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1236 result.append(buffer); 1237 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1238 result.append(buffer); 1239 1240 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1241 result.append(buffer); 1242 result.append(" Index Command"); 1243 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1244 snprintf(buffer, SIZE, "\n %02d ", i); 1245 result.append(buffer); 1246 result.append(mNewParameters[i]); 1247 } 1248 1249 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1250 result.append(buffer); 1251 snprintf(buffer, SIZE, " Index event param\n"); 1252 result.append(buffer); 1253 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1254 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1255 result.append(buffer); 1256 } 1257 result.append("\n"); 1258 1259 write(fd, result.string(), result.size()); 1260 1261 if (locked) { 1262 mLock.unlock(); 1263 } 1264 return NO_ERROR; 1265} 1266 1267status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1268{ 1269 const size_t SIZE = 256; 1270 char buffer[SIZE]; 1271 String8 result; 1272 1273 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1274 write(fd, buffer, strlen(buffer)); 1275 1276 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1277 sp<EffectChain> chain = mEffectChains[i]; 1278 if (chain != 0) { 1279 chain->dump(fd, args); 1280 } 1281 } 1282 return NO_ERROR; 1283} 1284 1285void AudioFlinger::ThreadBase::acquireWakeLock() 1286{ 1287 Mutex::Autolock _l(mLock); 1288 acquireWakeLock_l(); 1289} 1290 1291void AudioFlinger::ThreadBase::acquireWakeLock_l() 1292{ 1293 if (mPowerManager == 0) { 1294 // use checkService() to avoid blocking if power service is not up yet 1295 sp<IBinder> binder = 1296 defaultServiceManager()->checkService(String16("power")); 1297 if (binder == 0) { 1298 ALOGW("Thread %s cannot connect to the power manager service", mName); 1299 } else { 1300 mPowerManager = interface_cast<IPowerManager>(binder); 1301 binder->linkToDeath(mDeathRecipient); 1302 } 1303 } 1304 if (mPowerManager != 0) { 1305 sp<IBinder> binder = new BBinder(); 1306 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1307 binder, 1308 String16(mName)); 1309 if (status == NO_ERROR) { 1310 mWakeLockToken = binder; 1311 } 1312 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1313 } 1314} 1315 1316void AudioFlinger::ThreadBase::releaseWakeLock() 1317{ 1318 Mutex::Autolock _l(mLock); 1319 releaseWakeLock_l(); 1320} 1321 1322void AudioFlinger::ThreadBase::releaseWakeLock_l() 1323{ 1324 if (mWakeLockToken != 0) { 1325 ALOGV("releaseWakeLock_l() %s", mName); 1326 if (mPowerManager != 0) { 1327 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1328 } 1329 mWakeLockToken.clear(); 1330 } 1331} 1332 1333void AudioFlinger::ThreadBase::clearPowerManager() 1334{ 1335 Mutex::Autolock _l(mLock); 1336 releaseWakeLock_l(); 1337 mPowerManager.clear(); 1338} 1339 1340void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1341{ 1342 sp<ThreadBase> thread = mThread.promote(); 1343 if (thread != 0) { 1344 thread->clearPowerManager(); 1345 } 1346 ALOGW("power manager service died !!!"); 1347} 1348 1349void AudioFlinger::ThreadBase::setEffectSuspended( 1350 const effect_uuid_t *type, bool suspend, int sessionId) 1351{ 1352 Mutex::Autolock _l(mLock); 1353 setEffectSuspended_l(type, suspend, sessionId); 1354} 1355 1356void AudioFlinger::ThreadBase::setEffectSuspended_l( 1357 const effect_uuid_t *type, bool suspend, int sessionId) 1358{ 1359 sp<EffectChain> chain = getEffectChain_l(sessionId); 1360 if (chain != 0) { 1361 if (type != NULL) { 1362 chain->setEffectSuspended_l(type, suspend); 1363 } else { 1364 chain->setEffectSuspendedAll_l(suspend); 1365 } 1366 } 1367 1368 updateSuspendedSessions_l(type, suspend, sessionId); 1369} 1370 1371void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1372{ 1373 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1374 if (index < 0) { 1375 return; 1376 } 1377 1378 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1379 mSuspendedSessions.editValueAt(index); 1380 1381 for (size_t i = 0; i < sessionEffects.size(); i++) { 1382 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1383 for (int j = 0; j < desc->mRefCount; j++) { 1384 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1385 chain->setEffectSuspendedAll_l(true); 1386 } else { 1387 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1388 desc->mType.timeLow); 1389 chain->setEffectSuspended_l(&desc->mType, true); 1390 } 1391 } 1392 } 1393} 1394 1395void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1396 bool suspend, 1397 int sessionId) 1398{ 1399 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1400 1401 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1402 1403 if (suspend) { 1404 if (index >= 0) { 1405 sessionEffects = mSuspendedSessions.editValueAt(index); 1406 } else { 1407 mSuspendedSessions.add(sessionId, sessionEffects); 1408 } 1409 } else { 1410 if (index < 0) { 1411 return; 1412 } 1413 sessionEffects = mSuspendedSessions.editValueAt(index); 1414 } 1415 1416 1417 int key = EffectChain::kKeyForSuspendAll; 1418 if (type != NULL) { 1419 key = type->timeLow; 1420 } 1421 index = sessionEffects.indexOfKey(key); 1422 1423 sp<SuspendedSessionDesc> desc; 1424 if (suspend) { 1425 if (index >= 0) { 1426 desc = sessionEffects.valueAt(index); 1427 } else { 1428 desc = new SuspendedSessionDesc(); 1429 if (type != NULL) { 1430 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1431 } 1432 sessionEffects.add(key, desc); 1433 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1434 } 1435 desc->mRefCount++; 1436 } else { 1437 if (index < 0) { 1438 return; 1439 } 1440 desc = sessionEffects.valueAt(index); 1441 if (--desc->mRefCount == 0) { 1442 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1443 sessionEffects.removeItemsAt(index); 1444 if (sessionEffects.isEmpty()) { 1445 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1446 sessionId); 1447 mSuspendedSessions.removeItem(sessionId); 1448 } 1449 } 1450 } 1451 if (!sessionEffects.isEmpty()) { 1452 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1453 } 1454} 1455 1456void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1457 bool enabled, 1458 int sessionId) 1459{ 1460 Mutex::Autolock _l(mLock); 1461 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1462} 1463 1464void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1465 bool enabled, 1466 int sessionId) 1467{ 1468 if (mType != RECORD) { 1469 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1470 // another session. This gives the priority to well behaved effect control panels 1471 // and applications not using global effects. 1472 // Enabling post processing in AUDIO_SESSION_OUTPUT_STAGE session does not affect 1473 // global effects 1474 if ((sessionId != AUDIO_SESSION_OUTPUT_MIX) && (sessionId != AUDIO_SESSION_OUTPUT_STAGE)) { 1475 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1476 } 1477 } 1478 1479 sp<EffectChain> chain = getEffectChain_l(sessionId); 1480 if (chain != 0) { 1481 chain->checkSuspendOnEffectEnabled(effect, enabled); 1482 } 1483} 1484 1485// ---------------------------------------------------------------------------- 1486 1487AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1488 AudioStreamOut* output, 1489 audio_io_handle_t id, 1490 uint32_t device, 1491 type_t type) 1492 : ThreadBase(audioFlinger, id, device, type), 1493 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1494 // Assumes constructor is called by AudioFlinger with it's mLock held, 1495 // but it would be safer to explicitly pass initial masterMute as parameter 1496 mMasterMute(audioFlinger->masterMute_l()), 1497 // mStreamTypes[] initialized in constructor body 1498 mOutput(output), 1499 // Assumes constructor is called by AudioFlinger with it's mLock held, 1500 // but it would be safer to explicitly pass initial masterVolume as parameter 1501 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1502 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1503 mMixerStatus(MIXER_IDLE), 1504 mMixerStatusIgnoringFastTracks(MIXER_IDLE), 1505 standbyDelay(AudioFlinger::mStandbyTimeInNsecs), 1506 // index 0 is reserved for normal mixer's submix 1507 mFastTrackAvailMask(((1 << FastMixerState::kMaxFastTracks) - 1) & ~1) 1508{ 1509 snprintf(mName, kNameLength, "AudioOut_%X", id); 1510 1511 readOutputParameters(); 1512 1513 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1514 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1515 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1516 stream = (audio_stream_type_t) (stream + 1)) { 1517 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1518 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1519 } 1520 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1521 // because mAudioFlinger doesn't have one to copy from 1522} 1523 1524AudioFlinger::PlaybackThread::~PlaybackThread() 1525{ 1526 delete [] mMixBuffer; 1527} 1528 1529status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1530{ 1531 dumpInternals(fd, args); 1532 dumpTracks(fd, args); 1533 dumpEffectChains(fd, args); 1534 return NO_ERROR; 1535} 1536 1537status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1538{ 1539 const size_t SIZE = 256; 1540 char buffer[SIZE]; 1541 String8 result; 1542 1543 result.appendFormat("Output thread %p stream volumes in dB:\n ", this); 1544 for (int i = 0; i < AUDIO_STREAM_CNT; ++i) { 1545 const stream_type_t *st = &mStreamTypes[i]; 1546 if (i > 0) { 1547 result.appendFormat(", "); 1548 } 1549 result.appendFormat("%d:%.2g", i, 20.0 * log10(st->volume)); 1550 if (st->mute) { 1551 result.append("M"); 1552 } 1553 } 1554 result.append("\n"); 1555 write(fd, result.string(), result.length()); 1556 result.clear(); 1557 1558 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1559 result.append(buffer); 1560 Track::appendDumpHeader(result); 1561 for (size_t i = 0; i < mTracks.size(); ++i) { 1562 sp<Track> track = mTracks[i]; 1563 if (track != 0) { 1564 track->dump(buffer, SIZE); 1565 result.append(buffer); 1566 } 1567 } 1568 1569 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1570 result.append(buffer); 1571 Track::appendDumpHeader(result); 1572 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1573 sp<Track> track = mActiveTracks[i].promote(); 1574 if (track != 0) { 1575 track->dump(buffer, SIZE); 1576 result.append(buffer); 1577 } 1578 } 1579 write(fd, result.string(), result.size()); 1580 1581 // These values are "raw"; they will wrap around. See prepareTracks_l() for a better way. 1582 FastTrackUnderruns underruns = getFastTrackUnderruns(0); 1583 fdprintf(fd, "Normal mixer raw underrun counters: partial=%u empty=%u\n", 1584 underruns.mBitFields.mPartial, underruns.mBitFields.mEmpty); 1585 1586 return NO_ERROR; 1587} 1588 1589status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1590{ 1591 const size_t SIZE = 256; 1592 char buffer[SIZE]; 1593 String8 result; 1594 1595 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1596 result.append(buffer); 1597 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1598 result.append(buffer); 1599 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1600 result.append(buffer); 1601 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1602 result.append(buffer); 1603 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1604 result.append(buffer); 1605 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1606 result.append(buffer); 1607 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1608 result.append(buffer); 1609 write(fd, result.string(), result.size()); 1610 1611 dumpBase(fd, args); 1612 1613 return NO_ERROR; 1614} 1615 1616// Thread virtuals 1617status_t AudioFlinger::PlaybackThread::readyToRun() 1618{ 1619 status_t status = initCheck(); 1620 if (status == NO_ERROR) { 1621 ALOGI("AudioFlinger's thread %p ready to run", this); 1622 } else { 1623 ALOGE("No working audio driver found."); 1624 } 1625 return status; 1626} 1627 1628void AudioFlinger::PlaybackThread::onFirstRef() 1629{ 1630 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1631} 1632 1633// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1634sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1635 const sp<AudioFlinger::Client>& client, 1636 audio_stream_type_t streamType, 1637 uint32_t sampleRate, 1638 audio_format_t format, 1639 uint32_t channelMask, 1640 int frameCount, 1641 const sp<IMemory>& sharedBuffer, 1642 int sessionId, 1643 IAudioFlinger::track_flags_t flags, 1644 pid_t tid, 1645 status_t *status) 1646{ 1647 sp<Track> track; 1648 status_t lStatus; 1649 1650 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1651 1652 // client expresses a preference for FAST, but we get the final say 1653 if (flags & IAudioFlinger::TRACK_FAST) { 1654 if ( 1655 // not timed 1656 (!isTimed) && 1657 // either of these use cases: 1658 ( 1659 // use case 1: shared buffer with any frame count 1660 ( 1661 (sharedBuffer != 0) 1662 ) || 1663 // use case 2: callback handler and frame count is default or at least as large as HAL 1664 ( 1665 (tid != -1) && 1666 ((frameCount == 0) || 1667 (frameCount >= (int) (mFrameCount * 2))) // * 2 is due to SRC jitter, see below 1668 ) 1669 ) && 1670 // PCM data 1671 audio_is_linear_pcm(format) && 1672 // mono or stereo 1673 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1674 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1675#ifndef FAST_TRACKS_AT_NON_NATIVE_SAMPLE_RATE 1676 // hardware sample rate 1677 (sampleRate == mSampleRate) && 1678#endif 1679 // normal mixer has an associated fast mixer 1680 hasFastMixer() && 1681 // there are sufficient fast track slots available 1682 (mFastTrackAvailMask != 0) 1683 // FIXME test that MixerThread for this fast track has a capable output HAL 1684 // FIXME add a permission test also? 1685 ) { 1686 // if frameCount not specified, then it defaults to fast mixer (HAL) frame count 1687 if (frameCount == 0) { 1688 frameCount = mFrameCount * 2; // FIXME * 2 is due to SRC jitter, should be computed 1689 } 1690 ALOGV("AUDIO_OUTPUT_FLAG_FAST accepted: frameCount=%d mFrameCount=%d", 1691 frameCount, mFrameCount); 1692 } else { 1693 ALOGW("AUDIO_OUTPUT_FLAG_FAST denied: isTimed=%d sharedBuffer=%p frameCount=%d " 1694 "mFrameCount=%d format=%d isLinear=%d channelMask=%d sampleRate=%d mSampleRate=%d " 1695 "hasFastMixer=%d tid=%d fastTrackAvailMask=%#x", 1696 isTimed, sharedBuffer.get(), frameCount, mFrameCount, format, 1697 audio_is_linear_pcm(format), 1698 channelMask, sampleRate, mSampleRate, hasFastMixer(), tid, mFastTrackAvailMask); 1699 flags &= ~IAudioFlinger::TRACK_FAST; 1700 // For compatibility with AudioTrack calculation, buffer depth is forced 1701 // to be at least 2 x the normal mixer frame count and cover audio hardware latency. 1702 // This is probably too conservative, but legacy application code may depend on it. 1703 // If you change this calculation, also review the start threshold which is related. 1704 uint32_t latencyMs = mOutput->stream->get_latency(mOutput->stream); 1705 uint32_t minBufCount = latencyMs / ((1000 * mNormalFrameCount) / mSampleRate); 1706 if (minBufCount < 2) { 1707 minBufCount = 2; 1708 } 1709 int minFrameCount = mNormalFrameCount * minBufCount; 1710 if (frameCount < minFrameCount) { 1711 frameCount = minFrameCount; 1712 } 1713 } 1714 } 1715 1716 if (mType == DIRECT) { 1717 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1718 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1719 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1720 "for output %p with format %d", 1721 sampleRate, format, channelMask, mOutput, mFormat); 1722 lStatus = BAD_VALUE; 1723 goto Exit; 1724 } 1725 } 1726 } else { 1727 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1728 if (sampleRate > mSampleRate*2) { 1729 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1730 lStatus = BAD_VALUE; 1731 goto Exit; 1732 } 1733 } 1734 1735 lStatus = initCheck(); 1736 if (lStatus != NO_ERROR) { 1737 ALOGE("Audio driver not initialized."); 1738 goto Exit; 1739 } 1740 1741 { // scope for mLock 1742 Mutex::Autolock _l(mLock); 1743 1744 // all tracks in same audio session must share the same routing strategy otherwise 1745 // conflicts will happen when tracks are moved from one output to another by audio policy 1746 // manager 1747 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1748 for (size_t i = 0; i < mTracks.size(); ++i) { 1749 sp<Track> t = mTracks[i]; 1750 if (t != 0 && !t->isOutputTrack()) { 1751 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1752 if (sessionId == t->sessionId() && strategy != actual) { 1753 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1754 strategy, actual); 1755 lStatus = BAD_VALUE; 1756 goto Exit; 1757 } 1758 } 1759 } 1760 1761 if (!isTimed) { 1762 track = new Track(this, client, streamType, sampleRate, format, 1763 channelMask, frameCount, sharedBuffer, sessionId, flags); 1764 } else { 1765 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1766 channelMask, frameCount, sharedBuffer, sessionId); 1767 } 1768 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1769 lStatus = NO_MEMORY; 1770 goto Exit; 1771 } 1772 mTracks.add(track); 1773 1774 sp<EffectChain> chain = getEffectChain_l(sessionId); 1775 if (chain != 0) { 1776 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1777 track->setMainBuffer(chain->inBuffer()); 1778 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1779 chain->incTrackCnt(); 1780 } 1781 } 1782 1783#ifdef HAVE_REQUEST_PRIORITY 1784 if ((flags & IAudioFlinger::TRACK_FAST) && (tid != -1)) { 1785 pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1786 // we don't have CAP_SYS_NICE, nor do we want to have it as it's too powerful, 1787 // so ask activity manager to do this on our behalf 1788 int err = requestPriority(callingPid, tid, 1); 1789 if (err != 0) { 1790 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 1791 1, callingPid, tid, err); 1792 } 1793 } 1794#endif 1795 1796 lStatus = NO_ERROR; 1797 1798Exit: 1799 if (status) { 1800 *status = lStatus; 1801 } 1802 return track; 1803} 1804 1805uint32_t AudioFlinger::PlaybackThread::latency() const 1806{ 1807 Mutex::Autolock _l(mLock); 1808 if (initCheck() == NO_ERROR) { 1809 return mOutput->stream->get_latency(mOutput->stream); 1810 } else { 1811 return 0; 1812 } 1813} 1814 1815void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1816{ 1817 Mutex::Autolock _l(mLock); 1818 mMasterVolume = value; 1819} 1820 1821void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1822{ 1823 Mutex::Autolock _l(mLock); 1824 setMasterMute_l(muted); 1825} 1826 1827void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1828{ 1829 Mutex::Autolock _l(mLock); 1830 mStreamTypes[stream].volume = value; 1831} 1832 1833void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1834{ 1835 Mutex::Autolock _l(mLock); 1836 mStreamTypes[stream].mute = muted; 1837} 1838 1839float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1840{ 1841 Mutex::Autolock _l(mLock); 1842 return mStreamTypes[stream].volume; 1843} 1844 1845// addTrack_l() must be called with ThreadBase::mLock held 1846status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1847{ 1848 status_t status = ALREADY_EXISTS; 1849 1850 // set retry count for buffer fill 1851 track->mRetryCount = kMaxTrackStartupRetries; 1852 if (mActiveTracks.indexOf(track) < 0) { 1853 // the track is newly added, make sure it fills up all its 1854 // buffers before playing. This is to ensure the client will 1855 // effectively get the latency it requested. 1856 track->mFillingUpStatus = Track::FS_FILLING; 1857 track->mResetDone = false; 1858 track->mPresentationCompleteFrames = 0; 1859 mActiveTracks.add(track); 1860 if (track->mainBuffer() != mMixBuffer) { 1861 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1862 if (chain != 0) { 1863 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1864 chain->incActiveTrackCnt(); 1865 } 1866 } 1867 1868 status = NO_ERROR; 1869 } 1870 1871 ALOGV("mWaitWorkCV.broadcast"); 1872 mWaitWorkCV.broadcast(); 1873 1874 return status; 1875} 1876 1877// destroyTrack_l() must be called with ThreadBase::mLock held 1878void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1879{ 1880 track->mState = TrackBase::TERMINATED; 1881 // active tracks are removed by threadLoop() 1882 if (mActiveTracks.indexOf(track) < 0) { 1883 removeTrack_l(track); 1884 } 1885} 1886 1887void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1888{ 1889 track->triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 1890 mTracks.remove(track); 1891 deleteTrackName_l(track->name()); 1892 // redundant as track is about to be destroyed, for dumpsys only 1893 track->mName = -1; 1894 if (track->isFastTrack()) { 1895 int index = track->mFastIndex; 1896 ALOG_ASSERT(0 < index && index < (int)FastMixerState::kMaxFastTracks); 1897 ALOG_ASSERT(!(mFastTrackAvailMask & (1 << index))); 1898 mFastTrackAvailMask |= 1 << index; 1899 // redundant as track is about to be destroyed, for dumpsys only 1900 track->mFastIndex = -1; 1901 } 1902 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1903 if (chain != 0) { 1904 chain->decTrackCnt(); 1905 } 1906} 1907 1908String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1909{ 1910 String8 out_s8 = String8(""); 1911 char *s; 1912 1913 Mutex::Autolock _l(mLock); 1914 if (initCheck() != NO_ERROR) { 1915 return out_s8; 1916 } 1917 1918 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1919 out_s8 = String8(s); 1920 free(s); 1921 return out_s8; 1922} 1923 1924// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1925void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1926 AudioSystem::OutputDescriptor desc; 1927 void *param2 = NULL; 1928 1929 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1930 1931 switch (event) { 1932 case AudioSystem::OUTPUT_OPENED: 1933 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1934 desc.channels = mChannelMask; 1935 desc.samplingRate = mSampleRate; 1936 desc.format = mFormat; 1937 desc.frameCount = mNormalFrameCount; // FIXME see AudioFlinger::frameCount(audio_io_handle_t) 1938 desc.latency = latency(); 1939 param2 = &desc; 1940 break; 1941 1942 case AudioSystem::STREAM_CONFIG_CHANGED: 1943 param2 = ¶m; 1944 case AudioSystem::OUTPUT_CLOSED: 1945 default: 1946 break; 1947 } 1948 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1949} 1950 1951void AudioFlinger::PlaybackThread::readOutputParameters() 1952{ 1953 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1954 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1955 mChannelCount = (uint16_t)popcount(mChannelMask); 1956 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1957 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1958 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1959 if (mFrameCount & 15) { 1960 ALOGW("HAL output buffer size is %u frames but AudioMixer requires multiples of 16 frames", 1961 mFrameCount); 1962 } 1963 1964 // Calculate size of normal mix buffer relative to the HAL output buffer size 1965 double multiplier = 1.0; 1966 if (mType == MIXER && (kUseFastMixer == FastMixer_Static || kUseFastMixer == FastMixer_Dynamic)) { 1967 size_t minNormalFrameCount = (kMinNormalMixBufferSizeMs * mSampleRate) / 1000; 1968 size_t maxNormalFrameCount = (kMaxNormalMixBufferSizeMs * mSampleRate) / 1000; 1969 // round up minimum and round down maximum to nearest 16 frames to satisfy AudioMixer 1970 minNormalFrameCount = (minNormalFrameCount + 15) & ~15; 1971 maxNormalFrameCount = maxNormalFrameCount & ~15; 1972 if (maxNormalFrameCount < minNormalFrameCount) { 1973 maxNormalFrameCount = minNormalFrameCount; 1974 } 1975 multiplier = (double) minNormalFrameCount / (double) mFrameCount; 1976 if (multiplier <= 1.0) { 1977 multiplier = 1.0; 1978 } else if (multiplier <= 2.0) { 1979 if (2 * mFrameCount <= maxNormalFrameCount) { 1980 multiplier = 2.0; 1981 } else { 1982 multiplier = (double) maxNormalFrameCount / (double) mFrameCount; 1983 } 1984 } else { 1985 // prefer an even multiplier, for compatibility with doubling of fast tracks due to HAL SRC 1986 // (it would be unusual for the normal mix buffer size to not be a multiple of fast 1987 // track, but we sometimes have to do this to satisfy the maximum frame count constraint) 1988 // FIXME this rounding up should not be done if no HAL SRC 1989 uint32_t truncMult = (uint32_t) multiplier; 1990 if ((truncMult & 1)) { 1991 if ((truncMult + 1) * mFrameCount <= maxNormalFrameCount) { 1992 ++truncMult; 1993 } 1994 } 1995 multiplier = (double) truncMult; 1996 } 1997 } 1998 mNormalFrameCount = multiplier * mFrameCount; 1999 // round up to nearest 16 frames to satisfy AudioMixer 2000 mNormalFrameCount = (mNormalFrameCount + 15) & ~15; 2001 ALOGI("HAL output buffer size %u frames, normal mix buffer size %u frames", mFrameCount, mNormalFrameCount); 2002 2003 // FIXME - Current mixer implementation only supports stereo output: Always 2004 // Allocate a stereo buffer even if HW output is mono. 2005 delete[] mMixBuffer; 2006 mMixBuffer = new int16_t[mNormalFrameCount * 2]; 2007 memset(mMixBuffer, 0, mNormalFrameCount * 2 * sizeof(int16_t)); 2008 2009 // force reconfiguration of effect chains and engines to take new buffer size and audio 2010 // parameters into account 2011 // Note that mLock is not held when readOutputParameters() is called from the constructor 2012 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 2013 // matter. 2014 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 2015 Vector< sp<EffectChain> > effectChains = mEffectChains; 2016 for (size_t i = 0; i < effectChains.size(); i ++) { 2017 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 2018 } 2019} 2020 2021status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 2022{ 2023 if (halFrames == NULL || dspFrames == NULL) { 2024 return BAD_VALUE; 2025 } 2026 Mutex::Autolock _l(mLock); 2027 if (initCheck() != NO_ERROR) { 2028 return INVALID_OPERATION; 2029 } 2030 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2031 2032 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 2033} 2034 2035uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 2036{ 2037 Mutex::Autolock _l(mLock); 2038 uint32_t result = 0; 2039 if (getEffectChain_l(sessionId) != 0) { 2040 result = EFFECT_SESSION; 2041 } 2042 2043 for (size_t i = 0; i < mTracks.size(); ++i) { 2044 sp<Track> track = mTracks[i]; 2045 if (sessionId == track->sessionId() && 2046 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2047 result |= TRACK_SESSION; 2048 break; 2049 } 2050 } 2051 2052 return result; 2053} 2054 2055uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 2056{ 2057 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 2058 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 2059 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2060 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2061 } 2062 for (size_t i = 0; i < mTracks.size(); i++) { 2063 sp<Track> track = mTracks[i]; 2064 if (sessionId == track->sessionId() && 2065 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 2066 return AudioSystem::getStrategyForStream(track->streamType()); 2067 } 2068 } 2069 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 2070} 2071 2072 2073AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 2074{ 2075 Mutex::Autolock _l(mLock); 2076 return mOutput; 2077} 2078 2079AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 2080{ 2081 Mutex::Autolock _l(mLock); 2082 AudioStreamOut *output = mOutput; 2083 mOutput = NULL; 2084 // FIXME FastMixer might also have a raw ptr to mOutputSink; 2085 // must push a NULL and wait for ack 2086 mOutputSink.clear(); 2087 mPipeSink.clear(); 2088 mNormalSink.clear(); 2089 return output; 2090} 2091 2092// this method must always be called either with ThreadBase mLock held or inside the thread loop 2093audio_stream_t* AudioFlinger::PlaybackThread::stream() const 2094{ 2095 if (mOutput == NULL) { 2096 return NULL; 2097 } 2098 return &mOutput->stream->common; 2099} 2100 2101uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 2102{ 2103 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 2104 // decoding and transfer time. So sleeping for half of the latency would likely cause 2105 // underruns 2106 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 2107 return (uint32_t)((uint32_t)((mNormalFrameCount * 1000) / mSampleRate) * 1000); 2108 } else { 2109 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2110 } 2111} 2112 2113status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 2114{ 2115 if (!isValidSyncEvent(event)) { 2116 return BAD_VALUE; 2117 } 2118 2119 Mutex::Autolock _l(mLock); 2120 2121 for (size_t i = 0; i < mTracks.size(); ++i) { 2122 sp<Track> track = mTracks[i]; 2123 if (event->triggerSession() == track->sessionId()) { 2124 track->setSyncEvent(event); 2125 return NO_ERROR; 2126 } 2127 } 2128 2129 return NAME_NOT_FOUND; 2130} 2131 2132bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 2133{ 2134 switch (event->type()) { 2135 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 2136 return true; 2137 default: 2138 break; 2139 } 2140 return false; 2141} 2142 2143void AudioFlinger::PlaybackThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2144{ 2145 size_t count = tracksToRemove.size(); 2146 if (CC_UNLIKELY(count)) { 2147 for (size_t i = 0 ; i < count ; i++) { 2148 const sp<Track>& track = tracksToRemove.itemAt(i); 2149 if ((track->sharedBuffer() != 0) && 2150 (track->mState == TrackBase::ACTIVE || track->mState == TrackBase::RESUMING)) { 2151 AudioSystem::stopOutput(mId, track->streamType(), track->sessionId()); 2152 } 2153 } 2154 } 2155 2156} 2157 2158// ---------------------------------------------------------------------------- 2159 2160AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 2161 audio_io_handle_t id, uint32_t device, type_t type) 2162 : PlaybackThread(audioFlinger, output, id, device, type), 2163 // mAudioMixer below 2164#ifdef SOAKER 2165 mSoaker(NULL), 2166#endif 2167 // mFastMixer below 2168 mFastMixerFutex(0) 2169 // mOutputSink below 2170 // mPipeSink below 2171 // mNormalSink below 2172{ 2173 ALOGV("MixerThread() id=%d device=%d type=%d", id, device, type); 2174 ALOGV("mSampleRate=%d, mChannelMask=%d, mChannelCount=%d, mFormat=%d, mFrameSize=%d, " 2175 "mFrameCount=%d, mNormalFrameCount=%d", 2176 mSampleRate, mChannelMask, mChannelCount, mFormat, mFrameSize, mFrameCount, 2177 mNormalFrameCount); 2178 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 2179 2180 // FIXME - Current mixer implementation only supports stereo output 2181 if (mChannelCount == 1) { 2182 ALOGE("Invalid audio hardware channel count"); 2183 } 2184 2185 // create an NBAIO sink for the HAL output stream, and negotiate 2186 mOutputSink = new AudioStreamOutSink(output->stream); 2187 size_t numCounterOffers = 0; 2188 const NBAIO_Format offers[1] = {Format_from_SR_C(mSampleRate, mChannelCount)}; 2189 ssize_t index = mOutputSink->negotiate(offers, 1, NULL, numCounterOffers); 2190 ALOG_ASSERT(index == 0); 2191 2192 // initialize fast mixer depending on configuration 2193 bool initFastMixer; 2194 switch (kUseFastMixer) { 2195 case FastMixer_Never: 2196 initFastMixer = false; 2197 break; 2198 case FastMixer_Always: 2199 initFastMixer = true; 2200 break; 2201 case FastMixer_Static: 2202 case FastMixer_Dynamic: 2203 initFastMixer = mFrameCount < mNormalFrameCount; 2204 break; 2205 } 2206 if (initFastMixer) { 2207 2208 // create a MonoPipe to connect our submix to FastMixer 2209 NBAIO_Format format = mOutputSink->format(); 2210 // This pipe depth compensates for scheduling latency of the normal mixer thread. 2211 // When it wakes up after a maximum latency, it runs a few cycles quickly before 2212 // finally blocking. Note the pipe implementation rounds up the request to a power of 2. 2213 MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/); 2214 const NBAIO_Format offers[1] = {format}; 2215 size_t numCounterOffers = 0; 2216 ssize_t index = monoPipe->negotiate(offers, 1, NULL, numCounterOffers); 2217 ALOG_ASSERT(index == 0); 2218 mPipeSink = monoPipe; 2219 2220#ifdef SOAKER 2221 // create a soaker as workaround for governor issues 2222 mSoaker = new Soaker(); 2223 // FIXME Soaker should only run when needed, i.e. when FastMixer is not in COLD_IDLE 2224 mSoaker->run("Soaker", PRIORITY_LOWEST); 2225#endif 2226 2227 // create fast mixer and configure it initially with just one fast track for our submix 2228 mFastMixer = new FastMixer(); 2229 FastMixerStateQueue *sq = mFastMixer->sq(); 2230 FastMixerState *state = sq->begin(); 2231 FastTrack *fastTrack = &state->mFastTracks[0]; 2232 // wrap the source side of the MonoPipe to make it an AudioBufferProvider 2233 fastTrack->mBufferProvider = new SourceAudioBufferProvider(new MonoPipeReader(monoPipe)); 2234 fastTrack->mVolumeProvider = NULL; 2235 fastTrack->mGeneration++; 2236 state->mFastTracksGen++; 2237 state->mTrackMask = 1; 2238 // fast mixer will use the HAL output sink 2239 state->mOutputSink = mOutputSink.get(); 2240 state->mOutputSinkGen++; 2241 state->mFrameCount = mFrameCount; 2242 state->mCommand = FastMixerState::COLD_IDLE; 2243 // already done in constructor initialization list 2244 //mFastMixerFutex = 0; 2245 state->mColdFutexAddr = &mFastMixerFutex; 2246 state->mColdGen++; 2247 state->mDumpState = &mFastMixerDumpState; 2248 sq->end(); 2249 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2250 2251 // start the fast mixer 2252 mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO); 2253#ifdef HAVE_REQUEST_PRIORITY 2254 pid_t tid = mFastMixer->getTid(); 2255 int err = requestPriority(getpid_cached, tid, 2); 2256 if (err != 0) { 2257 ALOGW("Policy SCHED_FIFO priority %d is unavailable for pid %d tid %d; error %d", 2258 2, getpid_cached, tid, err); 2259 } 2260#endif 2261 2262 } else { 2263 mFastMixer = NULL; 2264 } 2265 2266 switch (kUseFastMixer) { 2267 case FastMixer_Never: 2268 case FastMixer_Dynamic: 2269 mNormalSink = mOutputSink; 2270 break; 2271 case FastMixer_Always: 2272 mNormalSink = mPipeSink; 2273 break; 2274 case FastMixer_Static: 2275 mNormalSink = initFastMixer ? mPipeSink : mOutputSink; 2276 break; 2277 } 2278} 2279 2280AudioFlinger::MixerThread::~MixerThread() 2281{ 2282 if (mFastMixer != NULL) { 2283 FastMixerStateQueue *sq = mFastMixer->sq(); 2284 FastMixerState *state = sq->begin(); 2285 if (state->mCommand == FastMixerState::COLD_IDLE) { 2286 int32_t old = android_atomic_inc(&mFastMixerFutex); 2287 if (old == -1) { 2288 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2289 } 2290 } 2291 state->mCommand = FastMixerState::EXIT; 2292 sq->end(); 2293 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2294 mFastMixer->join(); 2295 // Though the fast mixer thread has exited, it's state queue is still valid. 2296 // We'll use that extract the final state which contains one remaining fast track 2297 // corresponding to our sub-mix. 2298 state = sq->begin(); 2299 ALOG_ASSERT(state->mTrackMask == 1); 2300 FastTrack *fastTrack = &state->mFastTracks[0]; 2301 ALOG_ASSERT(fastTrack->mBufferProvider != NULL); 2302 delete fastTrack->mBufferProvider; 2303 sq->end(false /*didModify*/); 2304 delete mFastMixer; 2305#ifdef SOAKER 2306 if (mSoaker != NULL) { 2307 mSoaker->requestExitAndWait(); 2308 } 2309 delete mSoaker; 2310#endif 2311 } 2312 delete mAudioMixer; 2313} 2314 2315class CpuStats { 2316public: 2317 CpuStats(); 2318 void sample(const String8 &title); 2319#ifdef DEBUG_CPU_USAGE 2320private: 2321 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 2322 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 2323 2324 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 2325 2326 int mCpuNum; // thread's current CPU number 2327 int mCpukHz; // frequency of thread's current CPU in kHz 2328#endif 2329}; 2330 2331CpuStats::CpuStats() 2332#ifdef DEBUG_CPU_USAGE 2333 : mCpuNum(-1), mCpukHz(-1) 2334#endif 2335{ 2336} 2337 2338void CpuStats::sample(const String8 &title) { 2339#ifdef DEBUG_CPU_USAGE 2340 // get current thread's delta CPU time in wall clock ns 2341 double wcNs; 2342 bool valid = mCpuUsage.sampleAndEnable(wcNs); 2343 2344 // record sample for wall clock statistics 2345 if (valid) { 2346 mWcStats.sample(wcNs); 2347 } 2348 2349 // get the current CPU number 2350 int cpuNum = sched_getcpu(); 2351 2352 // get the current CPU frequency in kHz 2353 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2354 2355 // check if either CPU number or frequency changed 2356 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2357 mCpuNum = cpuNum; 2358 mCpukHz = cpukHz; 2359 // ignore sample for purposes of cycles 2360 valid = false; 2361 } 2362 2363 // if no change in CPU number or frequency, then record sample for cycle statistics 2364 if (valid && mCpukHz > 0) { 2365 double cycles = wcNs * cpukHz * 0.000001; 2366 mHzStats.sample(cycles); 2367 } 2368 2369 unsigned n = mWcStats.n(); 2370 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2371 if ((n & 127) == 1) { 2372 long long elapsed = mCpuUsage.elapsed(); 2373 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2374 double perLoop = elapsed / (double) n; 2375 double perLoop100 = perLoop * 0.01; 2376 double perLoop1k = perLoop * 0.001; 2377 double mean = mWcStats.mean(); 2378 double stddev = mWcStats.stddev(); 2379 double minimum = mWcStats.minimum(); 2380 double maximum = mWcStats.maximum(); 2381 double meanCycles = mHzStats.mean(); 2382 double stddevCycles = mHzStats.stddev(); 2383 double minCycles = mHzStats.minimum(); 2384 double maxCycles = mHzStats.maximum(); 2385 mCpuUsage.resetElapsed(); 2386 mWcStats.reset(); 2387 mHzStats.reset(); 2388 ALOGD("CPU usage for %s over past %.1f secs\n" 2389 " (%u mixer loops at %.1f mean ms per loop):\n" 2390 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2391 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2392 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2393 title.string(), 2394 elapsed * .000000001, n, perLoop * .000001, 2395 mean * .001, 2396 stddev * .001, 2397 minimum * .001, 2398 maximum * .001, 2399 mean / perLoop100, 2400 stddev / perLoop100, 2401 minimum / perLoop100, 2402 maximum / perLoop100, 2403 meanCycles / perLoop1k, 2404 stddevCycles / perLoop1k, 2405 minCycles / perLoop1k, 2406 maxCycles / perLoop1k); 2407 2408 } 2409 } 2410#endif 2411}; 2412 2413void AudioFlinger::PlaybackThread::checkSilentMode_l() 2414{ 2415 if (!mMasterMute) { 2416 char value[PROPERTY_VALUE_MAX]; 2417 if (property_get("ro.audio.silent", value, "0") > 0) { 2418 char *endptr; 2419 unsigned long ul = strtoul(value, &endptr, 0); 2420 if (*endptr == '\0' && ul != 0) { 2421 ALOGD("Silence is golden"); 2422 // The setprop command will not allow a property to be changed after 2423 // the first time it is set, so we don't have to worry about un-muting. 2424 setMasterMute_l(true); 2425 } 2426 } 2427 } 2428} 2429 2430bool AudioFlinger::PlaybackThread::threadLoop() 2431{ 2432 Vector< sp<Track> > tracksToRemove; 2433 2434 standbyTime = systemTime(); 2435 2436 // MIXER 2437 nsecs_t lastWarning = 0; 2438if (mType == MIXER) { 2439 longStandbyExit = false; 2440} 2441 2442 // DUPLICATING 2443 // FIXME could this be made local to while loop? 2444 writeFrames = 0; 2445 2446 cacheParameters_l(); 2447 sleepTime = idleSleepTime; 2448 2449if (mType == MIXER) { 2450 sleepTimeShift = 0; 2451} 2452 2453 CpuStats cpuStats; 2454 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2455 2456 acquireWakeLock(); 2457 2458 while (!exitPending()) 2459 { 2460 cpuStats.sample(myName); 2461 2462 Vector< sp<EffectChain> > effectChains; 2463 2464 processConfigEvents(); 2465 2466 { // scope for mLock 2467 2468 Mutex::Autolock _l(mLock); 2469 2470 if (checkForNewParameters_l()) { 2471 cacheParameters_l(); 2472 } 2473 2474 saveOutputTracks(); 2475 2476 // put audio hardware into standby after short delay 2477 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2478 mSuspended > 0)) { 2479 if (!mStandby) { 2480 2481 threadLoop_standby(); 2482 2483 mStandby = true; 2484 mBytesWritten = 0; 2485 } 2486 2487 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2488 // we're about to wait, flush the binder command buffer 2489 IPCThreadState::self()->flushCommands(); 2490 2491 clearOutputTracks(); 2492 2493 if (exitPending()) break; 2494 2495 releaseWakeLock_l(); 2496 // wait until we have something to do... 2497 ALOGV("%s going to sleep", myName.string()); 2498 mWaitWorkCV.wait(mLock); 2499 ALOGV("%s waking up", myName.string()); 2500 acquireWakeLock_l(); 2501 2502 mMixerStatus = MIXER_IDLE; 2503 mMixerStatusIgnoringFastTracks = MIXER_IDLE; 2504 2505 checkSilentMode_l(); 2506 2507 standbyTime = systemTime() + standbyDelay; 2508 sleepTime = idleSleepTime; 2509 if (mType == MIXER) { 2510 sleepTimeShift = 0; 2511 } 2512 2513 continue; 2514 } 2515 } 2516 2517 // mMixerStatusIgnoringFastTracks is also updated internally 2518 mMixerStatus = prepareTracks_l(&tracksToRemove); 2519 2520 // prevent any changes in effect chain list and in each effect chain 2521 // during mixing and effect process as the audio buffers could be deleted 2522 // or modified if an effect is created or deleted 2523 lockEffectChains_l(effectChains); 2524 } 2525 2526 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2527 threadLoop_mix(); 2528 } else { 2529 threadLoop_sleepTime(); 2530 } 2531 2532 if (mSuspended > 0) { 2533 sleepTime = suspendSleepTimeUs(); 2534 } 2535 2536 // only process effects if we're going to write 2537 if (sleepTime == 0) { 2538 for (size_t i = 0; i < effectChains.size(); i ++) { 2539 effectChains[i]->process_l(); 2540 } 2541 } 2542 2543 // enable changes in effect chain 2544 unlockEffectChains(effectChains); 2545 2546 // sleepTime == 0 means we must write to audio hardware 2547 if (sleepTime == 0) { 2548 2549 threadLoop_write(); 2550 2551if (mType == MIXER) { 2552 // write blocked detection 2553 nsecs_t now = systemTime(); 2554 nsecs_t delta = now - mLastWriteTime; 2555 if (!mStandby && delta > maxPeriod) { 2556 mNumDelayedWrites++; 2557 if ((now - lastWarning) > kWarningThrottleNs) { 2558#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2559 ScopedTrace st(ATRACE_TAG, "underrun"); 2560#endif 2561 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2562 ns2ms(delta), mNumDelayedWrites, this); 2563 lastWarning = now; 2564 } 2565 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2566 // a different threshold. Or completely removed for what it is worth anyway... 2567 if (mStandby) { 2568 longStandbyExit = true; 2569 } 2570 } 2571} 2572 2573 mStandby = false; 2574 } else { 2575 usleep(sleepTime); 2576 } 2577 2578 // Finally let go of removed track(s), without the lock held 2579 // since we can't guarantee the destructors won't acquire that 2580 // same lock. This will also mutate and push a new fast mixer state. 2581 threadLoop_removeTracks(tracksToRemove); 2582 tracksToRemove.clear(); 2583 2584 // FIXME I don't understand the need for this here; 2585 // it was in the original code but maybe the 2586 // assignment in saveOutputTracks() makes this unnecessary? 2587 clearOutputTracks(); 2588 2589 // Effect chains will be actually deleted here if they were removed from 2590 // mEffectChains list during mixing or effects processing 2591 effectChains.clear(); 2592 2593 // FIXME Note that the above .clear() is no longer necessary since effectChains 2594 // is now local to this block, but will keep it for now (at least until merge done). 2595 } 2596 2597if (mType == MIXER || mType == DIRECT) { 2598 // put output stream into standby mode 2599 if (!mStandby) { 2600 mOutput->stream->common.standby(&mOutput->stream->common); 2601 } 2602} 2603if (mType == DUPLICATING) { 2604 // for DuplicatingThread, standby mode is handled by the outputTracks 2605} 2606 2607 releaseWakeLock(); 2608 2609 ALOGV("Thread %p type %d exiting", this, mType); 2610 return false; 2611} 2612 2613void AudioFlinger::MixerThread::threadLoop_removeTracks(const Vector< sp<Track> >& tracksToRemove) 2614{ 2615 PlaybackThread::threadLoop_removeTracks(tracksToRemove); 2616} 2617 2618void AudioFlinger::MixerThread::threadLoop_write() 2619{ 2620 // FIXME we should only do one push per cycle; confirm this is true 2621 // Start the fast mixer if it's not already running 2622 if (mFastMixer != NULL) { 2623 FastMixerStateQueue *sq = mFastMixer->sq(); 2624 FastMixerState *state = sq->begin(); 2625 if (state->mCommand != FastMixerState::MIX_WRITE && 2626 (kUseFastMixer != FastMixer_Dynamic || state->mTrackMask > 1)) { 2627 if (state->mCommand == FastMixerState::COLD_IDLE) { 2628 int32_t old = android_atomic_inc(&mFastMixerFutex); 2629 if (old == -1) { 2630 __futex_syscall3(&mFastMixerFutex, FUTEX_WAKE_PRIVATE, 1); 2631 } 2632 } 2633 state->mCommand = FastMixerState::MIX_WRITE; 2634 sq->end(); 2635 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 2636 if (kUseFastMixer == FastMixer_Dynamic) { 2637 mNormalSink = mPipeSink; 2638 } 2639 } else { 2640 sq->end(false /*didModify*/); 2641 } 2642 } 2643 PlaybackThread::threadLoop_write(); 2644} 2645 2646// shared by MIXER and DIRECT, overridden by DUPLICATING 2647void AudioFlinger::PlaybackThread::threadLoop_write() 2648{ 2649 // FIXME rewrite to reduce number of system calls 2650 mLastWriteTime = systemTime(); 2651 mInWrite = true; 2652 2653#define mBitShift 2 // FIXME 2654 size_t count = mixBufferSize >> mBitShift; 2655#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2656 Tracer::traceBegin(ATRACE_TAG, "write"); 2657#endif 2658 ssize_t framesWritten = mNormalSink->write(mMixBuffer, count); 2659#if defined(ATRACE_TAG) && (ATRACE_TAG != ATRACE_TAG_NEVER) 2660 Tracer::traceEnd(ATRACE_TAG); 2661#endif 2662 if (framesWritten > 0) { 2663 size_t bytesWritten = framesWritten << mBitShift; 2664 mBytesWritten += bytesWritten; 2665 } 2666 2667 mNumWrites++; 2668 mInWrite = false; 2669} 2670 2671void AudioFlinger::MixerThread::threadLoop_standby() 2672{ 2673 // Idle the fast mixer if it's currently running 2674 if (mFastMixer != NULL) { 2675 FastMixerStateQueue *sq = mFastMixer->sq(); 2676 FastMixerState *state = sq->begin(); 2677 if (!(state->mCommand & FastMixerState::IDLE)) { 2678 state->mCommand = FastMixerState::COLD_IDLE; 2679 state->mColdFutexAddr = &mFastMixerFutex; 2680 state->mColdGen++; 2681 mFastMixerFutex = 0; 2682 sq->end(); 2683 // BLOCK_UNTIL_PUSHED would be insufficient, as we need it to stop doing I/O now 2684 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 2685 if (kUseFastMixer == FastMixer_Dynamic) { 2686 mNormalSink = mOutputSink; 2687 } 2688 } else { 2689 sq->end(false /*didModify*/); 2690 } 2691 } 2692 PlaybackThread::threadLoop_standby(); 2693} 2694 2695// shared by MIXER and DIRECT, overridden by DUPLICATING 2696void AudioFlinger::PlaybackThread::threadLoop_standby() 2697{ 2698 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2699 mOutput->stream->common.standby(&mOutput->stream->common); 2700} 2701 2702void AudioFlinger::MixerThread::threadLoop_mix() 2703{ 2704 // obtain the presentation timestamp of the next output buffer 2705 int64_t pts; 2706 status_t status = INVALID_OPERATION; 2707 2708 if (NULL != mOutput->stream->get_next_write_timestamp) { 2709 status = mOutput->stream->get_next_write_timestamp( 2710 mOutput->stream, &pts); 2711 } 2712 2713 if (status != NO_ERROR) { 2714 pts = AudioBufferProvider::kInvalidPTS; 2715 } 2716 2717 // mix buffers... 2718 mAudioMixer->process(pts); 2719 // increase sleep time progressively when application underrun condition clears. 2720 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2721 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2722 // such that we would underrun the audio HAL. 2723 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2724 sleepTimeShift--; 2725 } 2726 sleepTime = 0; 2727 standbyTime = systemTime() + standbyDelay; 2728 //TODO: delay standby when effects have a tail 2729} 2730 2731void AudioFlinger::MixerThread::threadLoop_sleepTime() 2732{ 2733 // If no tracks are ready, sleep once for the duration of an output 2734 // buffer size, then write 0s to the output 2735 if (sleepTime == 0) { 2736 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2737 sleepTime = activeSleepTime >> sleepTimeShift; 2738 if (sleepTime < kMinThreadSleepTimeUs) { 2739 sleepTime = kMinThreadSleepTimeUs; 2740 } 2741 // reduce sleep time in case of consecutive application underruns to avoid 2742 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2743 // duration we would end up writing less data than needed by the audio HAL if 2744 // the condition persists. 2745 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2746 sleepTimeShift++; 2747 } 2748 } else { 2749 sleepTime = idleSleepTime; 2750 } 2751 } else if (mBytesWritten != 0 || 2752 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2753 memset (mMixBuffer, 0, mixBufferSize); 2754 sleepTime = 0; 2755 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2756 } 2757 // TODO add standby time extension fct of effect tail 2758} 2759 2760// prepareTracks_l() must be called with ThreadBase::mLock held 2761AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2762 Vector< sp<Track> > *tracksToRemove) 2763{ 2764 2765 mixer_state mixerStatus = MIXER_IDLE; 2766 // find out which tracks need to be processed 2767 size_t count = mActiveTracks.size(); 2768 size_t mixedTracks = 0; 2769 size_t tracksWithEffect = 0; 2770 // counts only _active_ fast tracks 2771 size_t fastTracks = 0; 2772 uint32_t resetMask = 0; // bit mask of fast tracks that need to be reset 2773 2774 float masterVolume = mMasterVolume; 2775 bool masterMute = mMasterMute; 2776 2777 if (masterMute) { 2778 masterVolume = 0; 2779 } 2780 // Delegate master volume control to effect in output mix effect chain if needed 2781 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2782 if (chain != 0) { 2783 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2784 chain->setVolume_l(&v, &v); 2785 masterVolume = (float)((v + (1 << 23)) >> 24); 2786 chain.clear(); 2787 } 2788 2789 // prepare a new state to push 2790 FastMixerStateQueue *sq = NULL; 2791 FastMixerState *state = NULL; 2792 bool didModify = false; 2793 FastMixerStateQueue::block_t block = FastMixerStateQueue::BLOCK_UNTIL_PUSHED; 2794 if (mFastMixer != NULL) { 2795 sq = mFastMixer->sq(); 2796 state = sq->begin(); 2797 } 2798 2799 for (size_t i=0 ; i<count ; i++) { 2800 sp<Track> t = mActiveTracks[i].promote(); 2801 if (t == 0) continue; 2802 2803 // this const just means the local variable doesn't change 2804 Track* const track = t.get(); 2805 2806 // process fast tracks 2807 if (track->isFastTrack()) { 2808 2809 // It's theoretically possible (though unlikely) for a fast track to be created 2810 // and then removed within the same normal mix cycle. This is not a problem, as 2811 // the track never becomes active so it's fast mixer slot is never touched. 2812 // The converse, of removing an (active) track and then creating a new track 2813 // at the identical fast mixer slot within the same normal mix cycle, 2814 // is impossible because the slot isn't marked available until the end of each cycle. 2815 int j = track->mFastIndex; 2816 FastTrack *fastTrack = &state->mFastTracks[j]; 2817 2818 // Determine whether the track is currently in underrun condition, 2819 // and whether it had a recent underrun. 2820 FastTrackUnderruns underruns = mFastMixerDumpState.mTracks[j].mUnderruns; 2821 uint32_t recentFull = (underruns.mBitFields.mFull - 2822 track->mObservedUnderruns.mBitFields.mFull) & UNDERRUN_MASK; 2823 uint32_t recentPartial = (underruns.mBitFields.mPartial - 2824 track->mObservedUnderruns.mBitFields.mPartial) & UNDERRUN_MASK; 2825 uint32_t recentEmpty = (underruns.mBitFields.mEmpty - 2826 track->mObservedUnderruns.mBitFields.mEmpty) & UNDERRUN_MASK; 2827 uint32_t recentUnderruns = recentPartial + recentEmpty; 2828 track->mObservedUnderruns = underruns; 2829 // don't count underruns that occur while stopping or pausing 2830 // or stopped which can occur when flush() is called while active 2831 if (!(track->isStopping() || track->isPausing() || track->isStopped())) { 2832 track->mUnderrunCount += recentUnderruns; 2833 } 2834 2835 // This is similar to the state machine for normal tracks, 2836 // with a few modifications for fast tracks. 2837 bool isActive = true; 2838 switch (track->mState) { 2839 case TrackBase::STOPPING_1: 2840 // track stays active in STOPPING_1 state until first underrun 2841 if (recentUnderruns > 0) { 2842 track->mState = TrackBase::STOPPING_2; 2843 } 2844 break; 2845 case TrackBase::PAUSING: 2846 // ramp down is not yet implemented 2847 track->setPaused(); 2848 break; 2849 case TrackBase::RESUMING: 2850 // ramp up is not yet implemented 2851 track->mState = TrackBase::ACTIVE; 2852 break; 2853 case TrackBase::ACTIVE: 2854 if (recentFull > 0 || recentPartial > 0) { 2855 // track has provided at least some frames recently: reset retry count 2856 track->mRetryCount = kMaxTrackRetries; 2857 } 2858 if (recentUnderruns == 0) { 2859 // no recent underruns: stay active 2860 break; 2861 } 2862 // there has recently been an underrun of some kind 2863 if (track->sharedBuffer() == 0) { 2864 // were any of the recent underruns "empty" (no frames available)? 2865 if (recentEmpty == 0) { 2866 // no, then ignore the partial underruns as they are allowed indefinitely 2867 break; 2868 } 2869 // there has recently been an "empty" underrun: decrement the retry counter 2870 if (--(track->mRetryCount) > 0) { 2871 break; 2872 } 2873 // indicate to client process that the track was disabled because of underrun; 2874 // it will then automatically call start() when data is available 2875 android_atomic_or(CBLK_DISABLED_ON, &track->mCblk->flags); 2876 // remove from active list, but state remains ACTIVE [confusing but true] 2877 isActive = false; 2878 break; 2879 } 2880 // fall through 2881 case TrackBase::STOPPING_2: 2882 case TrackBase::PAUSED: 2883 case TrackBase::TERMINATED: 2884 case TrackBase::STOPPED: 2885 case TrackBase::FLUSHED: // flush() while active 2886 // Check for presentation complete if track is inactive 2887 // We have consumed all the buffers of this track. 2888 // This would be incomplete if we auto-paused on underrun 2889 { 2890 size_t audioHALFrames = 2891 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2892 size_t framesWritten = 2893 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2894 if (!track->presentationComplete(framesWritten, audioHALFrames)) { 2895 // track stays in active list until presentation is complete 2896 break; 2897 } 2898 } 2899 if (track->isStopping_2()) { 2900 track->mState = TrackBase::STOPPED; 2901 } 2902 if (track->isStopped()) { 2903 // Can't reset directly, as fast mixer is still polling this track 2904 // track->reset(); 2905 // So instead mark this track as needing to be reset after push with ack 2906 resetMask |= 1 << i; 2907 } 2908 isActive = false; 2909 break; 2910 case TrackBase::IDLE: 2911 default: 2912 LOG_FATAL("unexpected track state %d", track->mState); 2913 } 2914 2915 if (isActive) { 2916 // was it previously inactive? 2917 if (!(state->mTrackMask & (1 << j))) { 2918 ExtendedAudioBufferProvider *eabp = track; 2919 VolumeProvider *vp = track; 2920 fastTrack->mBufferProvider = eabp; 2921 fastTrack->mVolumeProvider = vp; 2922 fastTrack->mSampleRate = track->mSampleRate; 2923 fastTrack->mChannelMask = track->mChannelMask; 2924 fastTrack->mGeneration++; 2925 state->mTrackMask |= 1 << j; 2926 didModify = true; 2927 // no acknowledgement required for newly active tracks 2928 } 2929 // cache the combined master volume and stream type volume for fast mixer; this 2930 // lacks any synchronization or barrier so VolumeProvider may read a stale value 2931 track->mCachedVolume = track->isMuted() ? 2932 0 : masterVolume * mStreamTypes[track->streamType()].volume; 2933 ++fastTracks; 2934 } else { 2935 // was it previously active? 2936 if (state->mTrackMask & (1 << j)) { 2937 fastTrack->mBufferProvider = NULL; 2938 fastTrack->mGeneration++; 2939 state->mTrackMask &= ~(1 << j); 2940 didModify = true; 2941 // If any fast tracks were removed, we must wait for acknowledgement 2942 // because we're about to decrement the last sp<> on those tracks. 2943 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 2944 } else { 2945 LOG_FATAL("fast track %d should have been active", j); 2946 } 2947 tracksToRemove->add(track); 2948 // Avoids a misleading display in dumpsys 2949 track->mObservedUnderruns.mBitFields.mMostRecent = UNDERRUN_FULL; 2950 } 2951 continue; 2952 } 2953 2954 { // local variable scope to avoid goto warning 2955 2956 audio_track_cblk_t* cblk = track->cblk(); 2957 2958 // The first time a track is added we wait 2959 // for all its buffers to be filled before processing it 2960 int name = track->name(); 2961 // make sure that we have enough frames to mix one full buffer. 2962 // enforce this condition only once to enable draining the buffer in case the client 2963 // app does not call stop() and relies on underrun to stop: 2964 // hence the test on (mMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2965 // during last round 2966 uint32_t minFrames = 1; 2967 if ((track->sharedBuffer() == 0) && !track->isStopped() && !track->isPausing() && 2968 (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY)) { 2969 if (t->sampleRate() == (int)mSampleRate) { 2970 minFrames = mNormalFrameCount; 2971 } else { 2972 // +1 for rounding and +1 for additional sample needed for interpolation 2973 minFrames = (mNormalFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2974 // add frames already consumed but not yet released by the resampler 2975 // because cblk->framesReady() will include these frames 2976 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2977 // the minimum track buffer size is normally twice the number of frames necessary 2978 // to fill one buffer and the resampler should not leave more than one buffer worth 2979 // of unreleased frames after each pass, but just in case... 2980 ALOG_ASSERT(minFrames <= cblk->frameCount); 2981 } 2982 } 2983 if ((track->framesReady() >= minFrames) && track->isReady() && 2984 !track->isPaused() && !track->isTerminated()) 2985 { 2986 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2987 2988 mixedTracks++; 2989 2990 // track->mainBuffer() != mMixBuffer means there is an effect chain 2991 // connected to the track 2992 chain.clear(); 2993 if (track->mainBuffer() != mMixBuffer) { 2994 chain = getEffectChain_l(track->sessionId()); 2995 // Delegate volume control to effect in track effect chain if needed 2996 if (chain != 0) { 2997 tracksWithEffect++; 2998 } else { 2999 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 3000 name, track->sessionId()); 3001 } 3002 } 3003 3004 3005 int param = AudioMixer::VOLUME; 3006 if (track->mFillingUpStatus == Track::FS_FILLED) { 3007 // no ramp for the first volume setting 3008 track->mFillingUpStatus = Track::FS_ACTIVE; 3009 if (track->mState == TrackBase::RESUMING) { 3010 track->mState = TrackBase::ACTIVE; 3011 param = AudioMixer::RAMP_VOLUME; 3012 } 3013 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 3014 } else if (cblk->server != 0) { 3015 // If the track is stopped before the first frame was mixed, 3016 // do not apply ramp 3017 param = AudioMixer::RAMP_VOLUME; 3018 } 3019 3020 // compute volume for this track 3021 uint32_t vl, vr, va; 3022 if (track->isMuted() || track->isPausing() || 3023 mStreamTypes[track->streamType()].mute) { 3024 vl = vr = va = 0; 3025 if (track->isPausing()) { 3026 track->setPaused(); 3027 } 3028 } else { 3029 3030 // read original volumes with volume control 3031 float typeVolume = mStreamTypes[track->streamType()].volume; 3032 float v = masterVolume * typeVolume; 3033 uint32_t vlr = cblk->getVolumeLR(); 3034 vl = vlr & 0xFFFF; 3035 vr = vlr >> 16; 3036 // track volumes come from shared memory, so can't be trusted and must be clamped 3037 if (vl > MAX_GAIN_INT) { 3038 ALOGV("Track left volume out of range: %04X", vl); 3039 vl = MAX_GAIN_INT; 3040 } 3041 if (vr > MAX_GAIN_INT) { 3042 ALOGV("Track right volume out of range: %04X", vr); 3043 vr = MAX_GAIN_INT; 3044 } 3045 // now apply the master volume and stream type volume 3046 vl = (uint32_t)(v * vl) << 12; 3047 vr = (uint32_t)(v * vr) << 12; 3048 // assuming master volume and stream type volume each go up to 1.0, 3049 // vl and vr are now in 8.24 format 3050 3051 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 3052 // send level comes from shared memory and so may be corrupt 3053 if (sendLevel > MAX_GAIN_INT) { 3054 ALOGV("Track send level out of range: %04X", sendLevel); 3055 sendLevel = MAX_GAIN_INT; 3056 } 3057 va = (uint32_t)(v * sendLevel); 3058 } 3059 // Delegate volume control to effect in track effect chain if needed 3060 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 3061 // Do not ramp volume if volume is controlled by effect 3062 param = AudioMixer::VOLUME; 3063 track->mHasVolumeController = true; 3064 } else { 3065 // force no volume ramp when volume controller was just disabled or removed 3066 // from effect chain to avoid volume spike 3067 if (track->mHasVolumeController) { 3068 param = AudioMixer::VOLUME; 3069 } 3070 track->mHasVolumeController = false; 3071 } 3072 3073 // Convert volumes from 8.24 to 4.12 format 3074 // This additional clamping is needed in case chain->setVolume_l() overshot 3075 vl = (vl + (1 << 11)) >> 12; 3076 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 3077 vr = (vr + (1 << 11)) >> 12; 3078 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 3079 3080 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 3081 3082 // XXX: these things DON'T need to be done each time 3083 mAudioMixer->setBufferProvider(name, track); 3084 mAudioMixer->enable(name); 3085 3086 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 3087 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 3088 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 3089 mAudioMixer->setParameter( 3090 name, 3091 AudioMixer::TRACK, 3092 AudioMixer::FORMAT, (void *)track->format()); 3093 mAudioMixer->setParameter( 3094 name, 3095 AudioMixer::TRACK, 3096 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 3097 mAudioMixer->setParameter( 3098 name, 3099 AudioMixer::RESAMPLE, 3100 AudioMixer::SAMPLE_RATE, 3101 (void *)(cblk->sampleRate)); 3102 mAudioMixer->setParameter( 3103 name, 3104 AudioMixer::TRACK, 3105 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 3106 mAudioMixer->setParameter( 3107 name, 3108 AudioMixer::TRACK, 3109 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 3110 3111 // reset retry count 3112 track->mRetryCount = kMaxTrackRetries; 3113 3114 // If one track is ready, set the mixer ready if: 3115 // - the mixer was not ready during previous round OR 3116 // - no other track is not ready 3117 if (mMixerStatusIgnoringFastTracks != MIXER_TRACKS_READY || 3118 mixerStatus != MIXER_TRACKS_ENABLED) { 3119 mixerStatus = MIXER_TRACKS_READY; 3120 } 3121 } else { 3122 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 3123 if ((track->sharedBuffer() != 0) || track->isTerminated() || 3124 track->isStopped() || track->isPaused()) { 3125 // We have consumed all the buffers of this track. 3126 // Remove it from the list of active tracks. 3127 // TODO: use actual buffer filling status instead of latency when available from 3128 // audio HAL 3129 size_t audioHALFrames = 3130 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3131 size_t framesWritten = 3132 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3133 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3134 if (track->isStopped()) { 3135 track->reset(); 3136 } 3137 tracksToRemove->add(track); 3138 } 3139 } else { 3140 // No buffers for this track. Give it a few chances to 3141 // fill a buffer, then remove it from active list. 3142 if (--(track->mRetryCount) <= 0) { 3143 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 3144 tracksToRemove->add(track); 3145 // indicate to client process that the track was disabled because of underrun; 3146 // it will then automatically call start() when data is available 3147 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 3148 // If one track is not ready, mark the mixer also not ready if: 3149 // - the mixer was ready during previous round OR 3150 // - no other track is ready 3151 } else if (mMixerStatusIgnoringFastTracks == MIXER_TRACKS_READY || 3152 mixerStatus != MIXER_TRACKS_READY) { 3153 mixerStatus = MIXER_TRACKS_ENABLED; 3154 } 3155 } 3156 mAudioMixer->disable(name); 3157 } 3158 3159 } // local variable scope to avoid goto warning 3160track_is_ready: ; 3161 3162 } 3163 3164 // Push the new FastMixer state if necessary 3165 if (didModify) { 3166 state->mFastTracksGen++; 3167 // if the fast mixer was active, but now there are no fast tracks, then put it in cold idle 3168 if (kUseFastMixer == FastMixer_Dynamic && 3169 state->mCommand == FastMixerState::MIX_WRITE && state->mTrackMask <= 1) { 3170 state->mCommand = FastMixerState::COLD_IDLE; 3171 state->mColdFutexAddr = &mFastMixerFutex; 3172 state->mColdGen++; 3173 mFastMixerFutex = 0; 3174 if (kUseFastMixer == FastMixer_Dynamic) { 3175 mNormalSink = mOutputSink; 3176 } 3177 // If we go into cold idle, need to wait for acknowledgement 3178 // so that fast mixer stops doing I/O. 3179 block = FastMixerStateQueue::BLOCK_UNTIL_ACKED; 3180 } 3181 sq->end(); 3182 } 3183 if (sq != NULL) { 3184 sq->end(didModify); 3185 sq->push(block); 3186 } 3187 3188 // Now perform the deferred reset on fast tracks that have stopped 3189 while (resetMask != 0) { 3190 size_t i = __builtin_ctz(resetMask); 3191 ALOG_ASSERT(i < count); 3192 resetMask &= ~(1 << i); 3193 sp<Track> t = mActiveTracks[i].promote(); 3194 if (t == 0) continue; 3195 Track* track = t.get(); 3196 ALOG_ASSERT(track->isFastTrack() && track->isStopped()); 3197 track->reset(); 3198 } 3199 3200 // remove all the tracks that need to be... 3201 count = tracksToRemove->size(); 3202 if (CC_UNLIKELY(count)) { 3203 for (size_t i=0 ; i<count ; i++) { 3204 const sp<Track>& track = tracksToRemove->itemAt(i); 3205 mActiveTracks.remove(track); 3206 if (track->mainBuffer() != mMixBuffer) { 3207 chain = getEffectChain_l(track->sessionId()); 3208 if (chain != 0) { 3209 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 3210 chain->decActiveTrackCnt(); 3211 } 3212 } 3213 if (track->isTerminated()) { 3214 removeTrack_l(track); 3215 } 3216 } 3217 } 3218 3219 // mix buffer must be cleared if all tracks are connected to an 3220 // effect chain as in this case the mixer will not write to 3221 // mix buffer and track effects will accumulate into it 3222 if ((mixedTracks != 0 && mixedTracks == tracksWithEffect) || (mixedTracks == 0 && fastTracks > 0)) { 3223 // FIXME as a performance optimization, should remember previous zero status 3224 memset(mMixBuffer, 0, mNormalFrameCount * mChannelCount * sizeof(int16_t)); 3225 } 3226 3227 // if any fast tracks, then status is ready 3228 mMixerStatusIgnoringFastTracks = mixerStatus; 3229 if (fastTracks > 0) { 3230 mixerStatus = MIXER_TRACKS_READY; 3231 } 3232 return mixerStatus; 3233} 3234 3235/* 3236The derived values that are cached: 3237 - mixBufferSize from frame count * frame size 3238 - activeSleepTime from activeSleepTimeUs() 3239 - idleSleepTime from idleSleepTimeUs() 3240 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 3241 - maxPeriod from frame count and sample rate (MIXER only) 3242 3243The parameters that affect these derived values are: 3244 - frame count 3245 - frame size 3246 - sample rate 3247 - device type: A2DP or not 3248 - device latency 3249 - format: PCM or not 3250 - active sleep time 3251 - idle sleep time 3252*/ 3253 3254void AudioFlinger::PlaybackThread::cacheParameters_l() 3255{ 3256 mixBufferSize = mNormalFrameCount * mFrameSize; 3257 activeSleepTime = activeSleepTimeUs(); 3258 idleSleepTime = idleSleepTimeUs(); 3259} 3260 3261void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 3262{ 3263 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 3264 this, streamType, mTracks.size()); 3265 Mutex::Autolock _l(mLock); 3266 3267 size_t size = mTracks.size(); 3268 for (size_t i = 0; i < size; i++) { 3269 sp<Track> t = mTracks[i]; 3270 if (t->streamType() == streamType) { 3271 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 3272 t->mCblk->cv.signal(); 3273 } 3274 } 3275} 3276 3277// getTrackName_l() must be called with ThreadBase::mLock held 3278int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 3279{ 3280 return mAudioMixer->getTrackName(channelMask); 3281} 3282 3283// deleteTrackName_l() must be called with ThreadBase::mLock held 3284void AudioFlinger::MixerThread::deleteTrackName_l(int name) 3285{ 3286 ALOGV("remove track (%d) and delete from mixer", name); 3287 mAudioMixer->deleteTrackName(name); 3288} 3289 3290// checkForNewParameters_l() must be called with ThreadBase::mLock held 3291bool AudioFlinger::MixerThread::checkForNewParameters_l() 3292{ 3293 // if !&IDLE, holds the FastMixer state to restore after new parameters processed 3294 FastMixerState::Command previousCommand = FastMixerState::HOT_IDLE; 3295 bool reconfig = false; 3296 3297 while (!mNewParameters.isEmpty()) { 3298 3299 if (mFastMixer != NULL) { 3300 FastMixerStateQueue *sq = mFastMixer->sq(); 3301 FastMixerState *state = sq->begin(); 3302 if (!(state->mCommand & FastMixerState::IDLE)) { 3303 previousCommand = state->mCommand; 3304 state->mCommand = FastMixerState::HOT_IDLE; 3305 sq->end(); 3306 sq->push(FastMixerStateQueue::BLOCK_UNTIL_ACKED); 3307 } else { 3308 sq->end(false /*didModify*/); 3309 } 3310 } 3311 3312 status_t status = NO_ERROR; 3313 String8 keyValuePair = mNewParameters[0]; 3314 AudioParameter param = AudioParameter(keyValuePair); 3315 int value; 3316 3317 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 3318 reconfig = true; 3319 } 3320 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 3321 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 3322 status = BAD_VALUE; 3323 } else { 3324 reconfig = true; 3325 } 3326 } 3327 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 3328 if (value != AUDIO_CHANNEL_OUT_STEREO) { 3329 status = BAD_VALUE; 3330 } else { 3331 reconfig = true; 3332 } 3333 } 3334 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3335 // do not accept frame count changes if tracks are open as the track buffer 3336 // size depends on frame count and correct behavior would not be guaranteed 3337 // if frame count is changed after track creation 3338 if (!mTracks.isEmpty()) { 3339 status = INVALID_OPERATION; 3340 } else { 3341 reconfig = true; 3342 } 3343 } 3344 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 3345#ifdef ADD_BATTERY_DATA 3346 // when changing the audio output device, call addBatteryData to notify 3347 // the change 3348 if ((int)mDevice != value) { 3349 uint32_t params = 0; 3350 // check whether speaker is on 3351 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 3352 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 3353 } 3354 3355 int deviceWithoutSpeaker 3356 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 3357 // check if any other device (except speaker) is on 3358 if (value & deviceWithoutSpeaker ) { 3359 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 3360 } 3361 3362 if (params != 0) { 3363 addBatteryData(params); 3364 } 3365 } 3366#endif 3367 3368 // forward device change to effects that have requested to be 3369 // aware of attached audio device. 3370 mDevice = (uint32_t)value; 3371 for (size_t i = 0; i < mEffectChains.size(); i++) { 3372 mEffectChains[i]->setDevice_l(mDevice); 3373 } 3374 } 3375 3376 if (status == NO_ERROR) { 3377 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3378 keyValuePair.string()); 3379 if (!mStandby && status == INVALID_OPERATION) { 3380 mOutput->stream->common.standby(&mOutput->stream->common); 3381 mStandby = true; 3382 mBytesWritten = 0; 3383 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3384 keyValuePair.string()); 3385 } 3386 if (status == NO_ERROR && reconfig) { 3387 delete mAudioMixer; 3388 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 3389 mAudioMixer = NULL; 3390 readOutputParameters(); 3391 mAudioMixer = new AudioMixer(mNormalFrameCount, mSampleRate); 3392 for (size_t i = 0; i < mTracks.size() ; i++) { 3393 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 3394 if (name < 0) break; 3395 mTracks[i]->mName = name; 3396 // limit track sample rate to 2 x new output sample rate 3397 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 3398 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 3399 } 3400 } 3401 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3402 } 3403 } 3404 3405 mNewParameters.removeAt(0); 3406 3407 mParamStatus = status; 3408 mParamCond.signal(); 3409 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3410 // already timed out waiting for the status and will never signal the condition. 3411 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3412 } 3413 3414 if (!(previousCommand & FastMixerState::IDLE)) { 3415 ALOG_ASSERT(mFastMixer != NULL); 3416 FastMixerStateQueue *sq = mFastMixer->sq(); 3417 FastMixerState *state = sq->begin(); 3418 ALOG_ASSERT(state->mCommand == FastMixerState::HOT_IDLE); 3419 state->mCommand = previousCommand; 3420 sq->end(); 3421 sq->push(FastMixerStateQueue::BLOCK_UNTIL_PUSHED); 3422 } 3423 3424 return reconfig; 3425} 3426 3427status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 3428{ 3429 const size_t SIZE = 256; 3430 char buffer[SIZE]; 3431 String8 result; 3432 3433 PlaybackThread::dumpInternals(fd, args); 3434 3435 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 3436 result.append(buffer); 3437 write(fd, result.string(), result.size()); 3438 3439 // Make a non-atomic copy of fast mixer dump state so it won't change underneath us 3440 FastMixerDumpState copy = mFastMixerDumpState; 3441 copy.dump(fd); 3442 3443 return NO_ERROR; 3444} 3445 3446uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 3447{ 3448 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000) / 2; 3449} 3450 3451uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 3452{ 3453 return (uint32_t)(((mNormalFrameCount * 1000) / mSampleRate) * 1000); 3454} 3455 3456void AudioFlinger::MixerThread::cacheParameters_l() 3457{ 3458 PlaybackThread::cacheParameters_l(); 3459 3460 // FIXME: Relaxed timing because of a certain device that can't meet latency 3461 // Should be reduced to 2x after the vendor fixes the driver issue 3462 // increase threshold again due to low power audio mode. The way this warning 3463 // threshold is calculated and its usefulness should be reconsidered anyway. 3464 maxPeriod = seconds(mNormalFrameCount) / mSampleRate * 15; 3465} 3466 3467// ---------------------------------------------------------------------------- 3468AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 3469 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 3470 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 3471 // mLeftVolFloat, mRightVolFloat 3472 // mLeftVolShort, mRightVolShort 3473{ 3474} 3475 3476AudioFlinger::DirectOutputThread::~DirectOutputThread() 3477{ 3478} 3479 3480AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 3481 Vector< sp<Track> > *tracksToRemove 3482) 3483{ 3484 sp<Track> trackToRemove; 3485 3486 mixer_state mixerStatus = MIXER_IDLE; 3487 3488 // find out which tracks need to be processed 3489 if (mActiveTracks.size() != 0) { 3490 sp<Track> t = mActiveTracks[0].promote(); 3491 // The track died recently 3492 if (t == 0) return MIXER_IDLE; 3493 3494 Track* const track = t.get(); 3495 audio_track_cblk_t* cblk = track->cblk(); 3496 3497 // The first time a track is added we wait 3498 // for all its buffers to be filled before processing it 3499 if (cblk->framesReady() && track->isReady() && 3500 !track->isPaused() && !track->isTerminated()) 3501 { 3502 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 3503 3504 if (track->mFillingUpStatus == Track::FS_FILLED) { 3505 track->mFillingUpStatus = Track::FS_ACTIVE; 3506 mLeftVolFloat = mRightVolFloat = 0; 3507 mLeftVolShort = mRightVolShort = 0; 3508 if (track->mState == TrackBase::RESUMING) { 3509 track->mState = TrackBase::ACTIVE; 3510 rampVolume = true; 3511 } 3512 } else if (cblk->server != 0) { 3513 // If the track is stopped before the first frame was mixed, 3514 // do not apply ramp 3515 rampVolume = true; 3516 } 3517 // compute volume for this track 3518 float left, right; 3519 if (track->isMuted() || mMasterMute || track->isPausing() || 3520 mStreamTypes[track->streamType()].mute) { 3521 left = right = 0; 3522 if (track->isPausing()) { 3523 track->setPaused(); 3524 } 3525 } else { 3526 float typeVolume = mStreamTypes[track->streamType()].volume; 3527 float v = mMasterVolume * typeVolume; 3528 uint32_t vlr = cblk->getVolumeLR(); 3529 float v_clamped = v * (vlr & 0xFFFF); 3530 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3531 left = v_clamped/MAX_GAIN; 3532 v_clamped = v * (vlr >> 16); 3533 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 3534 right = v_clamped/MAX_GAIN; 3535 } 3536 3537 if (left != mLeftVolFloat || right != mRightVolFloat) { 3538 mLeftVolFloat = left; 3539 mRightVolFloat = right; 3540 3541 // If audio HAL implements volume control, 3542 // force software volume to nominal value 3543 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 3544 left = 1.0f; 3545 right = 1.0f; 3546 } 3547 3548 // Convert volumes from float to 8.24 3549 uint32_t vl = (uint32_t)(left * (1 << 24)); 3550 uint32_t vr = (uint32_t)(right * (1 << 24)); 3551 3552 // Delegate volume control to effect in track effect chain if needed 3553 // only one effect chain can be present on DirectOutputThread, so if 3554 // there is one, the track is connected to it 3555 if (!mEffectChains.isEmpty()) { 3556 // Do not ramp volume if volume is controlled by effect 3557 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 3558 rampVolume = false; 3559 } 3560 } 3561 3562 // Convert volumes from 8.24 to 4.12 format 3563 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 3564 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3565 leftVol = (uint16_t)v_clamped; 3566 v_clamped = (vr + (1 << 11)) >> 12; 3567 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 3568 rightVol = (uint16_t)v_clamped; 3569 } else { 3570 leftVol = mLeftVolShort; 3571 rightVol = mRightVolShort; 3572 rampVolume = false; 3573 } 3574 3575 // reset retry count 3576 track->mRetryCount = kMaxTrackRetriesDirect; 3577 mActiveTrack = t; 3578 mixerStatus = MIXER_TRACKS_READY; 3579 } else { 3580 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 3581 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 3582 // We have consumed all the buffers of this track. 3583 // Remove it from the list of active tracks. 3584 // TODO: implement behavior for compressed audio 3585 size_t audioHALFrames = 3586 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 3587 size_t framesWritten = 3588 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 3589 if (track->presentationComplete(framesWritten, audioHALFrames)) { 3590 if (track->isStopped()) { 3591 track->reset(); 3592 } 3593 trackToRemove = track; 3594 } 3595 } else { 3596 // No buffers for this track. Give it a few chances to 3597 // fill a buffer, then remove it from active list. 3598 if (--(track->mRetryCount) <= 0) { 3599 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 3600 trackToRemove = track; 3601 } else { 3602 mixerStatus = MIXER_TRACKS_ENABLED; 3603 } 3604 } 3605 } 3606 } 3607 3608 // FIXME merge this with similar code for removing multiple tracks 3609 // remove all the tracks that need to be... 3610 if (CC_UNLIKELY(trackToRemove != 0)) { 3611 tracksToRemove->add(trackToRemove); 3612 mActiveTracks.remove(trackToRemove); 3613 if (!mEffectChains.isEmpty()) { 3614 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 3615 trackToRemove->sessionId()); 3616 mEffectChains[0]->decActiveTrackCnt(); 3617 } 3618 if (trackToRemove->isTerminated()) { 3619 removeTrack_l(trackToRemove); 3620 } 3621 } 3622 3623 return mixerStatus; 3624} 3625 3626void AudioFlinger::DirectOutputThread::threadLoop_mix() 3627{ 3628 AudioBufferProvider::Buffer buffer; 3629 size_t frameCount = mFrameCount; 3630 int8_t *curBuf = (int8_t *)mMixBuffer; 3631 // output audio to hardware 3632 while (frameCount) { 3633 buffer.frameCount = frameCount; 3634 mActiveTrack->getNextBuffer(&buffer); 3635 if (CC_UNLIKELY(buffer.raw == NULL)) { 3636 memset(curBuf, 0, frameCount * mFrameSize); 3637 break; 3638 } 3639 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 3640 frameCount -= buffer.frameCount; 3641 curBuf += buffer.frameCount * mFrameSize; 3642 mActiveTrack->releaseBuffer(&buffer); 3643 } 3644 sleepTime = 0; 3645 standbyTime = systemTime() + standbyDelay; 3646 mActiveTrack.clear(); 3647 3648 // apply volume 3649 3650 // Do not apply volume on compressed audio 3651 if (!audio_is_linear_pcm(mFormat)) { 3652 return; 3653 } 3654 3655 // convert to signed 16 bit before volume calculation 3656 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3657 size_t count = mFrameCount * mChannelCount; 3658 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 3659 int16_t *dst = mMixBuffer + count-1; 3660 while (count--) { 3661 *dst-- = (int16_t)(*src--^0x80) << 8; 3662 } 3663 } 3664 3665 frameCount = mFrameCount; 3666 int16_t *out = mMixBuffer; 3667 if (rampVolume) { 3668 if (mChannelCount == 1) { 3669 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3670 int32_t vlInc = d / (int32_t)frameCount; 3671 int32_t vl = ((int32_t)mLeftVolShort << 16); 3672 do { 3673 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3674 out++; 3675 vl += vlInc; 3676 } while (--frameCount); 3677 3678 } else { 3679 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3680 int32_t vlInc = d / (int32_t)frameCount; 3681 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3682 int32_t vrInc = d / (int32_t)frameCount; 3683 int32_t vl = ((int32_t)mLeftVolShort << 16); 3684 int32_t vr = ((int32_t)mRightVolShort << 16); 3685 do { 3686 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3687 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3688 out += 2; 3689 vl += vlInc; 3690 vr += vrInc; 3691 } while (--frameCount); 3692 } 3693 } else { 3694 if (mChannelCount == 1) { 3695 do { 3696 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3697 out++; 3698 } while (--frameCount); 3699 } else { 3700 do { 3701 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3702 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3703 out += 2; 3704 } while (--frameCount); 3705 } 3706 } 3707 3708 // convert back to unsigned 8 bit after volume calculation 3709 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3710 size_t count = mFrameCount * mChannelCount; 3711 int16_t *src = mMixBuffer; 3712 uint8_t *dst = (uint8_t *)mMixBuffer; 3713 while (count--) { 3714 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3715 } 3716 } 3717 3718 mLeftVolShort = leftVol; 3719 mRightVolShort = rightVol; 3720} 3721 3722void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3723{ 3724 if (sleepTime == 0) { 3725 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3726 sleepTime = activeSleepTime; 3727 } else { 3728 sleepTime = idleSleepTime; 3729 } 3730 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3731 memset(mMixBuffer, 0, mFrameCount * mFrameSize); 3732 sleepTime = 0; 3733 } 3734} 3735 3736// getTrackName_l() must be called with ThreadBase::mLock held 3737int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3738{ 3739 return 0; 3740} 3741 3742// deleteTrackName_l() must be called with ThreadBase::mLock held 3743void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3744{ 3745} 3746 3747// checkForNewParameters_l() must be called with ThreadBase::mLock held 3748bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3749{ 3750 bool reconfig = false; 3751 3752 while (!mNewParameters.isEmpty()) { 3753 status_t status = NO_ERROR; 3754 String8 keyValuePair = mNewParameters[0]; 3755 AudioParameter param = AudioParameter(keyValuePair); 3756 int value; 3757 3758 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3759 // do not accept frame count changes if tracks are open as the track buffer 3760 // size depends on frame count and correct behavior would not be garantied 3761 // if frame count is changed after track creation 3762 if (!mTracks.isEmpty()) { 3763 status = INVALID_OPERATION; 3764 } else { 3765 reconfig = true; 3766 } 3767 } 3768 if (status == NO_ERROR) { 3769 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3770 keyValuePair.string()); 3771 if (!mStandby && status == INVALID_OPERATION) { 3772 mOutput->stream->common.standby(&mOutput->stream->common); 3773 mStandby = true; 3774 mBytesWritten = 0; 3775 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3776 keyValuePair.string()); 3777 } 3778 if (status == NO_ERROR && reconfig) { 3779 readOutputParameters(); 3780 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3781 } 3782 } 3783 3784 mNewParameters.removeAt(0); 3785 3786 mParamStatus = status; 3787 mParamCond.signal(); 3788 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3789 // already timed out waiting for the status and will never signal the condition. 3790 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3791 } 3792 return reconfig; 3793} 3794 3795uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3796{ 3797 uint32_t time; 3798 if (audio_is_linear_pcm(mFormat)) { 3799 time = PlaybackThread::activeSleepTimeUs(); 3800 } else { 3801 time = 10000; 3802 } 3803 return time; 3804} 3805 3806uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3807{ 3808 uint32_t time; 3809 if (audio_is_linear_pcm(mFormat)) { 3810 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3811 } else { 3812 time = 10000; 3813 } 3814 return time; 3815} 3816 3817uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3818{ 3819 uint32_t time; 3820 if (audio_is_linear_pcm(mFormat)) { 3821 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3822 } else { 3823 time = 10000; 3824 } 3825 return time; 3826} 3827 3828void AudioFlinger::DirectOutputThread::cacheParameters_l() 3829{ 3830 PlaybackThread::cacheParameters_l(); 3831 3832 // use shorter standby delay as on normal output to release 3833 // hardware resources as soon as possible 3834 standbyDelay = microseconds(activeSleepTime*2); 3835} 3836 3837// ---------------------------------------------------------------------------- 3838 3839AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3840 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3841 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3842 mWaitTimeMs(UINT_MAX) 3843{ 3844 addOutputTrack(mainThread); 3845} 3846 3847AudioFlinger::DuplicatingThread::~DuplicatingThread() 3848{ 3849 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3850 mOutputTracks[i]->destroy(); 3851 } 3852} 3853 3854void AudioFlinger::DuplicatingThread::threadLoop_mix() 3855{ 3856 // mix buffers... 3857 if (outputsReady(outputTracks)) { 3858 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3859 } else { 3860 memset(mMixBuffer, 0, mixBufferSize); 3861 } 3862 sleepTime = 0; 3863 writeFrames = mNormalFrameCount; 3864} 3865 3866void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3867{ 3868 if (sleepTime == 0) { 3869 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3870 sleepTime = activeSleepTime; 3871 } else { 3872 sleepTime = idleSleepTime; 3873 } 3874 } else if (mBytesWritten != 0) { 3875 // flush remaining overflow buffers in output tracks 3876 for (size_t i = 0; i < outputTracks.size(); i++) { 3877 if (outputTracks[i]->isActive()) { 3878 sleepTime = 0; 3879 writeFrames = 0; 3880 memset(mMixBuffer, 0, mixBufferSize); 3881 break; 3882 } 3883 } 3884 } 3885} 3886 3887void AudioFlinger::DuplicatingThread::threadLoop_write() 3888{ 3889 standbyTime = systemTime() + standbyDelay; 3890 for (size_t i = 0; i < outputTracks.size(); i++) { 3891 outputTracks[i]->write(mMixBuffer, writeFrames); 3892 } 3893 mBytesWritten += mixBufferSize; 3894} 3895 3896void AudioFlinger::DuplicatingThread::threadLoop_standby() 3897{ 3898 // DuplicatingThread implements standby by stopping all tracks 3899 for (size_t i = 0; i < outputTracks.size(); i++) { 3900 outputTracks[i]->stop(); 3901 } 3902} 3903 3904void AudioFlinger::DuplicatingThread::saveOutputTracks() 3905{ 3906 outputTracks = mOutputTracks; 3907} 3908 3909void AudioFlinger::DuplicatingThread::clearOutputTracks() 3910{ 3911 outputTracks.clear(); 3912} 3913 3914void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3915{ 3916 Mutex::Autolock _l(mLock); 3917 // FIXME explain this formula 3918 int frameCount = (3 * mNormalFrameCount * mSampleRate) / thread->sampleRate(); 3919 OutputTrack *outputTrack = new OutputTrack(thread, 3920 this, 3921 mSampleRate, 3922 mFormat, 3923 mChannelMask, 3924 frameCount); 3925 if (outputTrack->cblk() != NULL) { 3926 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3927 mOutputTracks.add(outputTrack); 3928 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3929 updateWaitTime_l(); 3930 } 3931} 3932 3933void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3934{ 3935 Mutex::Autolock _l(mLock); 3936 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3937 if (mOutputTracks[i]->thread() == thread) { 3938 mOutputTracks[i]->destroy(); 3939 mOutputTracks.removeAt(i); 3940 updateWaitTime_l(); 3941 return; 3942 } 3943 } 3944 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3945} 3946 3947// caller must hold mLock 3948void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3949{ 3950 mWaitTimeMs = UINT_MAX; 3951 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3952 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3953 if (strong != 0) { 3954 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3955 if (waitTimeMs < mWaitTimeMs) { 3956 mWaitTimeMs = waitTimeMs; 3957 } 3958 } 3959 } 3960} 3961 3962 3963bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3964{ 3965 for (size_t i = 0; i < outputTracks.size(); i++) { 3966 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3967 if (thread == 0) { 3968 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3969 return false; 3970 } 3971 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3972 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3973 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3974 return false; 3975 } 3976 } 3977 return true; 3978} 3979 3980uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3981{ 3982 return (mWaitTimeMs * 1000) / 2; 3983} 3984 3985void AudioFlinger::DuplicatingThread::cacheParameters_l() 3986{ 3987 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3988 updateWaitTime_l(); 3989 3990 MixerThread::cacheParameters_l(); 3991} 3992 3993// ---------------------------------------------------------------------------- 3994 3995// TrackBase constructor must be called with AudioFlinger::mLock held 3996AudioFlinger::ThreadBase::TrackBase::TrackBase( 3997 ThreadBase *thread, 3998 const sp<Client>& client, 3999 uint32_t sampleRate, 4000 audio_format_t format, 4001 uint32_t channelMask, 4002 int frameCount, 4003 const sp<IMemory>& sharedBuffer, 4004 int sessionId) 4005 : RefBase(), 4006 mThread(thread), 4007 mClient(client), 4008 mCblk(NULL), 4009 // mBuffer 4010 // mBufferEnd 4011 mFrameCount(0), 4012 mState(IDLE), 4013 mSampleRate(sampleRate), 4014 mFormat(format), 4015 mStepServerFailed(false), 4016 mSessionId(sessionId) 4017 // mChannelCount 4018 // mChannelMask 4019{ 4020 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 4021 4022 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 4023 size_t size = sizeof(audio_track_cblk_t); 4024 uint8_t channelCount = popcount(channelMask); 4025 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 4026 if (sharedBuffer == 0) { 4027 size += bufferSize; 4028 } 4029 4030 if (client != NULL) { 4031 mCblkMemory = client->heap()->allocate(size); 4032 if (mCblkMemory != 0) { 4033 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 4034 if (mCblk != NULL) { // construct the shared structure in-place. 4035 new(mCblk) audio_track_cblk_t(); 4036 // clear all buffers 4037 mCblk->frameCount = frameCount; 4038 mCblk->sampleRate = sampleRate; 4039// uncomment the following lines to quickly test 32-bit wraparound 4040// mCblk->user = 0xffff0000; 4041// mCblk->server = 0xffff0000; 4042// mCblk->userBase = 0xffff0000; 4043// mCblk->serverBase = 0xffff0000; 4044 mChannelCount = channelCount; 4045 mChannelMask = channelMask; 4046 if (sharedBuffer == 0) { 4047 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4048 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4049 // Force underrun condition to avoid false underrun callback until first data is 4050 // written to buffer (other flags are cleared) 4051 mCblk->flags = CBLK_UNDERRUN_ON; 4052 } else { 4053 mBuffer = sharedBuffer->pointer(); 4054 } 4055 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4056 } 4057 } else { 4058 ALOGE("not enough memory for AudioTrack size=%u", size); 4059 client->heap()->dump("AudioTrack"); 4060 return; 4061 } 4062 } else { 4063 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 4064 // construct the shared structure in-place. 4065 new(mCblk) audio_track_cblk_t(); 4066 // clear all buffers 4067 mCblk->frameCount = frameCount; 4068 mCblk->sampleRate = sampleRate; 4069// uncomment the following lines to quickly test 32-bit wraparound 4070// mCblk->user = 0xffff0000; 4071// mCblk->server = 0xffff0000; 4072// mCblk->userBase = 0xffff0000; 4073// mCblk->serverBase = 0xffff0000; 4074 mChannelCount = channelCount; 4075 mChannelMask = channelMask; 4076 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 4077 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 4078 // Force underrun condition to avoid false underrun callback until first data is 4079 // written to buffer (other flags are cleared) 4080 mCblk->flags = CBLK_UNDERRUN_ON; 4081 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 4082 } 4083} 4084 4085AudioFlinger::ThreadBase::TrackBase::~TrackBase() 4086{ 4087 if (mCblk != NULL) { 4088 if (mClient == 0) { 4089 delete mCblk; 4090 } else { 4091 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 4092 } 4093 } 4094 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 4095 if (mClient != 0) { 4096 // Client destructor must run with AudioFlinger mutex locked 4097 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 4098 // If the client's reference count drops to zero, the associated destructor 4099 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 4100 // relying on the automatic clear() at end of scope. 4101 mClient.clear(); 4102 } 4103} 4104 4105// AudioBufferProvider interface 4106// getNextBuffer() = 0; 4107// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 4108void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4109{ 4110 buffer->raw = NULL; 4111 mFrameCount = buffer->frameCount; 4112 // FIXME See note at getNextBuffer() 4113 (void) step(); // ignore return value of step() 4114 buffer->frameCount = 0; 4115} 4116 4117bool AudioFlinger::ThreadBase::TrackBase::step() { 4118 bool result; 4119 audio_track_cblk_t* cblk = this->cblk(); 4120 4121 result = cblk->stepServer(mFrameCount); 4122 if (!result) { 4123 ALOGV("stepServer failed acquiring cblk mutex"); 4124 mStepServerFailed = true; 4125 } 4126 return result; 4127} 4128 4129void AudioFlinger::ThreadBase::TrackBase::reset() { 4130 audio_track_cblk_t* cblk = this->cblk(); 4131 4132 cblk->user = 0; 4133 cblk->server = 0; 4134 cblk->userBase = 0; 4135 cblk->serverBase = 0; 4136 mStepServerFailed = false; 4137 ALOGV("TrackBase::reset"); 4138} 4139 4140int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 4141 return (int)mCblk->sampleRate; 4142} 4143 4144void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 4145 audio_track_cblk_t* cblk = this->cblk(); 4146 size_t frameSize = cblk->frameSize; 4147 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 4148 int8_t *bufferEnd = bufferStart + frames * frameSize; 4149 4150 // Check validity of returned pointer in case the track control block would have been corrupted. 4151 ALOG_ASSERT(!(bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd), 4152 "TrackBase::getBuffer buffer out of range:\n" 4153 " start: %p, end %p , mBuffer %p mBufferEnd %p\n" 4154 " server %u, serverBase %u, user %u, userBase %u, frameSize %d", 4155 bufferStart, bufferEnd, mBuffer, mBufferEnd, 4156 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, frameSize); 4157 4158 return bufferStart; 4159} 4160 4161status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 4162{ 4163 mSyncEvents.add(event); 4164 return NO_ERROR; 4165} 4166 4167// ---------------------------------------------------------------------------- 4168 4169// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 4170AudioFlinger::PlaybackThread::Track::Track( 4171 PlaybackThread *thread, 4172 const sp<Client>& client, 4173 audio_stream_type_t streamType, 4174 uint32_t sampleRate, 4175 audio_format_t format, 4176 uint32_t channelMask, 4177 int frameCount, 4178 const sp<IMemory>& sharedBuffer, 4179 int sessionId, 4180 IAudioFlinger::track_flags_t flags) 4181 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 4182 mMute(false), 4183 mFillingUpStatus(FS_INVALID), 4184 // mRetryCount initialized later when needed 4185 mSharedBuffer(sharedBuffer), 4186 mStreamType(streamType), 4187 mName(-1), // see note below 4188 mMainBuffer(thread->mixBuffer()), 4189 mAuxBuffer(NULL), 4190 mAuxEffectId(0), mHasVolumeController(false), 4191 mPresentationCompleteFrames(0), 4192 mFlags(flags), 4193 mFastIndex(-1), 4194 mUnderrunCount(0), 4195 mCachedVolume(1.0) 4196{ 4197 if (mCblk != NULL) { 4198 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 4199 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 4200 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 4201 if (flags & IAudioFlinger::TRACK_FAST) { 4202 mCblk->flags |= CBLK_FAST; // atomic op not needed yet 4203 ALOG_ASSERT(thread->mFastTrackAvailMask != 0); 4204 int i = __builtin_ctz(thread->mFastTrackAvailMask); 4205 ALOG_ASSERT(0 < i && i < (int)FastMixerState::kMaxFastTracks); 4206 // FIXME This is too eager. We allocate a fast track index before the 4207 // fast track becomes active. Since fast tracks are a scarce resource, 4208 // this means we are potentially denying other more important fast tracks from 4209 // being created. It would be better to allocate the index dynamically. 4210 mFastIndex = i; 4211 // Read the initial underruns because this field is never cleared by the fast mixer 4212 mObservedUnderruns = thread->getFastTrackUnderruns(i); 4213 thread->mFastTrackAvailMask &= ~(1 << i); 4214 } 4215 // to avoid leaking a track name, do not allocate one unless there is an mCblk 4216 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 4217 if (mName < 0) { 4218 ALOGE("no more track names available"); 4219 // FIXME bug - if sufficient fast track indices, but insufficient normal mixer names, 4220 // then we leak a fast track index. Should swap these two sections, or better yet 4221 // only allocate a normal mixer name for normal tracks. 4222 } 4223 } 4224 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4225} 4226 4227AudioFlinger::PlaybackThread::Track::~Track() 4228{ 4229 ALOGV("PlaybackThread::Track destructor"); 4230 sp<ThreadBase> thread = mThread.promote(); 4231 if (thread != 0) { 4232 Mutex::Autolock _l(thread->mLock); 4233 mState = TERMINATED; 4234 } 4235} 4236 4237void AudioFlinger::PlaybackThread::Track::destroy() 4238{ 4239 // NOTE: destroyTrack_l() can remove a strong reference to this Track 4240 // by removing it from mTracks vector, so there is a risk that this Tracks's 4241 // destructor is called. As the destructor needs to lock mLock, 4242 // we must acquire a strong reference on this Track before locking mLock 4243 // here so that the destructor is called only when exiting this function. 4244 // On the other hand, as long as Track::destroy() is only called by 4245 // TrackHandle destructor, the TrackHandle still holds a strong ref on 4246 // this Track with its member mTrack. 4247 sp<Track> keep(this); 4248 { // scope for mLock 4249 sp<ThreadBase> thread = mThread.promote(); 4250 if (thread != 0) { 4251 if (!isOutputTrack()) { 4252 if (mState == ACTIVE || mState == RESUMING) { 4253 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4254 4255#ifdef ADD_BATTERY_DATA 4256 // to track the speaker usage 4257 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4258#endif 4259 } 4260 AudioSystem::releaseOutput(thread->id()); 4261 } 4262 Mutex::Autolock _l(thread->mLock); 4263 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4264 playbackThread->destroyTrack_l(this); 4265 } 4266 } 4267} 4268 4269/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result) 4270{ 4271 result.append(" Name Client Type Fmt Chn mask Session mFrCnt fCount S M F SRate L dB R dB " 4272 " Server User Main buf Aux Buf Flags FastUnder\n"); 4273} 4274 4275void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 4276{ 4277 uint32_t vlr = mCblk->getVolumeLR(); 4278 if (isFastTrack()) { 4279 sprintf(buffer, " F %2d", mFastIndex); 4280 } else { 4281 sprintf(buffer, " %4d", mName - AudioMixer::TRACK0); 4282 } 4283 track_state state = mState; 4284 char stateChar; 4285 switch (state) { 4286 case IDLE: 4287 stateChar = 'I'; 4288 break; 4289 case TERMINATED: 4290 stateChar = 'T'; 4291 break; 4292 case STOPPING_1: 4293 stateChar = 's'; 4294 break; 4295 case STOPPING_2: 4296 stateChar = '5'; 4297 break; 4298 case STOPPED: 4299 stateChar = 'S'; 4300 break; 4301 case RESUMING: 4302 stateChar = 'R'; 4303 break; 4304 case ACTIVE: 4305 stateChar = 'A'; 4306 break; 4307 case PAUSING: 4308 stateChar = 'p'; 4309 break; 4310 case PAUSED: 4311 stateChar = 'P'; 4312 break; 4313 case FLUSHED: 4314 stateChar = 'F'; 4315 break; 4316 default: 4317 stateChar = '?'; 4318 break; 4319 } 4320 char nowInUnderrun; 4321 switch (mObservedUnderruns.mBitFields.mMostRecent) { 4322 case UNDERRUN_FULL: 4323 nowInUnderrun = ' '; 4324 break; 4325 case UNDERRUN_PARTIAL: 4326 nowInUnderrun = '<'; 4327 break; 4328 case UNDERRUN_EMPTY: 4329 nowInUnderrun = '*'; 4330 break; 4331 default: 4332 nowInUnderrun = '?'; 4333 break; 4334 } 4335 snprintf(&buffer[7], size-7, " %6d %4u %3u 0x%08x %7u %6u %6u %1c %1d %1d %5u %5.2g %5.2g " 4336 "0x%08x 0x%08x 0x%08x 0x%08x %#5x %9u%c\n", 4337 (mClient == 0) ? getpid_cached : mClient->pid(), 4338 mStreamType, 4339 mFormat, 4340 mChannelMask, 4341 mSessionId, 4342 mFrameCount, 4343 mCblk->frameCount, 4344 stateChar, 4345 mMute, 4346 mFillingUpStatus, 4347 mCblk->sampleRate, 4348 20.0 * log10((vlr & 0xFFFF) / 4096.0), 4349 20.0 * log10((vlr >> 16) / 4096.0), 4350 mCblk->server, 4351 mCblk->user, 4352 (int)mMainBuffer, 4353 (int)mAuxBuffer, 4354 mCblk->flags, 4355 mUnderrunCount, 4356 nowInUnderrun); 4357} 4358 4359// AudioBufferProvider interface 4360status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 4361 AudioBufferProvider::Buffer* buffer, int64_t pts) 4362{ 4363 audio_track_cblk_t* cblk = this->cblk(); 4364 uint32_t framesReady; 4365 uint32_t framesReq = buffer->frameCount; 4366 4367 // Check if last stepServer failed, try to step now 4368 if (mStepServerFailed) { 4369 // FIXME When called by fast mixer, this takes a mutex with tryLock(). 4370 // Since the fast mixer is higher priority than client callback thread, 4371 // it does not result in priority inversion for client. 4372 // But a non-blocking solution would be preferable to avoid 4373 // fast mixer being unable to tryLock(), and 4374 // to avoid the extra context switches if the client wakes up, 4375 // discovers the mutex is locked, then has to wait for fast mixer to unlock. 4376 if (!step()) goto getNextBuffer_exit; 4377 ALOGV("stepServer recovered"); 4378 mStepServerFailed = false; 4379 } 4380 4381 // FIXME Same as above 4382 framesReady = cblk->framesReady(); 4383 4384 if (CC_LIKELY(framesReady)) { 4385 uint32_t s = cblk->server; 4386 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4387 4388 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 4389 if (framesReq > framesReady) { 4390 framesReq = framesReady; 4391 } 4392 if (framesReq > bufferEnd - s) { 4393 framesReq = bufferEnd - s; 4394 } 4395 4396 buffer->raw = getBuffer(s, framesReq); 4397 if (buffer->raw == NULL) goto getNextBuffer_exit; 4398 4399 buffer->frameCount = framesReq; 4400 return NO_ERROR; 4401 } 4402 4403getNextBuffer_exit: 4404 buffer->raw = NULL; 4405 buffer->frameCount = 0; 4406 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 4407 return NOT_ENOUGH_DATA; 4408} 4409 4410// Note that framesReady() takes a mutex on the control block using tryLock(). 4411// This could result in priority inversion if framesReady() is called by the normal mixer, 4412// as the normal mixer thread runs at lower 4413// priority than the client's callback thread: there is a short window within framesReady() 4414// during which the normal mixer could be preempted, and the client callback would block. 4415// Another problem can occur if framesReady() is called by the fast mixer: 4416// the tryLock() could block for up to 1 ms, and a sequence of these could delay fast mixer. 4417// FIXME Replace AudioTrackShared control block implementation by a non-blocking FIFO queue. 4418size_t AudioFlinger::PlaybackThread::Track::framesReady() const { 4419 return mCblk->framesReady(); 4420} 4421 4422// Don't call for fast tracks; the framesReady() could result in priority inversion 4423bool AudioFlinger::PlaybackThread::Track::isReady() const { 4424 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 4425 4426 if (framesReady() >= mCblk->frameCount || 4427 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 4428 mFillingUpStatus = FS_FILLED; 4429 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4430 return true; 4431 } 4432 return false; 4433} 4434 4435status_t AudioFlinger::PlaybackThread::Track::start(AudioSystem::sync_event_t event, 4436 int triggerSession) 4437{ 4438 status_t status = NO_ERROR; 4439 ALOGV("start(%d), calling pid %d session %d", 4440 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 4441 4442 sp<ThreadBase> thread = mThread.promote(); 4443 if (thread != 0) { 4444 Mutex::Autolock _l(thread->mLock); 4445 track_state state = mState; 4446 // here the track could be either new, or restarted 4447 // in both cases "unstop" the track 4448 if (mState == PAUSED) { 4449 mState = TrackBase::RESUMING; 4450 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 4451 } else { 4452 mState = TrackBase::ACTIVE; 4453 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 4454 } 4455 4456 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 4457 thread->mLock.unlock(); 4458 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 4459 thread->mLock.lock(); 4460 4461#ifdef ADD_BATTERY_DATA 4462 // to track the speaker usage 4463 if (status == NO_ERROR) { 4464 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 4465 } 4466#endif 4467 } 4468 if (status == NO_ERROR) { 4469 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4470 playbackThread->addTrack_l(this); 4471 } else { 4472 mState = state; 4473 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4474 } 4475 } else { 4476 status = BAD_VALUE; 4477 } 4478 return status; 4479} 4480 4481void AudioFlinger::PlaybackThread::Track::stop() 4482{ 4483 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4484 sp<ThreadBase> thread = mThread.promote(); 4485 if (thread != 0) { 4486 Mutex::Autolock _l(thread->mLock); 4487 track_state state = mState; 4488 if (state == RESUMING || state == ACTIVE || state == PAUSING || state == PAUSED) { 4489 // If the track is not active (PAUSED and buffers full), flush buffers 4490 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4491 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4492 reset(); 4493 mState = STOPPED; 4494 } else if (!isFastTrack()) { 4495 mState = STOPPED; 4496 } else { 4497 // prepareTracks_l() will set state to STOPPING_2 after next underrun, 4498 // and then to STOPPED and reset() when presentation is complete 4499 mState = STOPPING_1; 4500 } 4501 ALOGV("not stopping/stopped => stopping/stopped (%d) on thread %p", mName, playbackThread); 4502 } 4503 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 4504 thread->mLock.unlock(); 4505 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4506 thread->mLock.lock(); 4507 4508#ifdef ADD_BATTERY_DATA 4509 // to track the speaker usage 4510 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4511#endif 4512 } 4513 } 4514} 4515 4516void AudioFlinger::PlaybackThread::Track::pause() 4517{ 4518 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 4519 sp<ThreadBase> thread = mThread.promote(); 4520 if (thread != 0) { 4521 Mutex::Autolock _l(thread->mLock); 4522 if (mState == ACTIVE || mState == RESUMING) { 4523 mState = PAUSING; 4524 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 4525 if (!isOutputTrack()) { 4526 thread->mLock.unlock(); 4527 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 4528 thread->mLock.lock(); 4529 4530#ifdef ADD_BATTERY_DATA 4531 // to track the speaker usage 4532 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 4533#endif 4534 } 4535 } 4536 } 4537} 4538 4539void AudioFlinger::PlaybackThread::Track::flush() 4540{ 4541 ALOGV("flush(%d)", mName); 4542 sp<ThreadBase> thread = mThread.promote(); 4543 if (thread != 0) { 4544 Mutex::Autolock _l(thread->mLock); 4545 if (mState != STOPPING_1 && mState != STOPPING_2 && mState != STOPPED && mState != PAUSED && 4546 mState != PAUSING) { 4547 return; 4548 } 4549 // No point remaining in PAUSED state after a flush => go to 4550 // FLUSHED state 4551 mState = FLUSHED; 4552 // do not reset the track if it is still in the process of being stopped or paused. 4553 // this will be done by prepareTracks_l() when the track is stopped. 4554 // prepareTracks_l() will see mState == FLUSHED, then 4555 // remove from active track list, reset(), and trigger presentation complete 4556 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4557 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 4558 reset(); 4559 } 4560 } 4561} 4562 4563void AudioFlinger::PlaybackThread::Track::reset() 4564{ 4565 // Do not reset twice to avoid discarding data written just after a flush and before 4566 // the audioflinger thread detects the track is stopped. 4567 if (!mResetDone) { 4568 TrackBase::reset(); 4569 // Force underrun condition to avoid false underrun callback until first data is 4570 // written to buffer 4571 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 4572 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4573 mFillingUpStatus = FS_FILLING; 4574 mResetDone = true; 4575 if (mState == FLUSHED) { 4576 mState = IDLE; 4577 } 4578 } 4579} 4580 4581void AudioFlinger::PlaybackThread::Track::mute(bool muted) 4582{ 4583 mMute = muted; 4584} 4585 4586status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 4587{ 4588 status_t status = DEAD_OBJECT; 4589 sp<ThreadBase> thread = mThread.promote(); 4590 if (thread != 0) { 4591 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 4592 status = playbackThread->attachAuxEffect(this, EffectId); 4593 } 4594 return status; 4595} 4596 4597void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 4598{ 4599 mAuxEffectId = EffectId; 4600 mAuxBuffer = buffer; 4601} 4602 4603bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 4604 size_t audioHalFrames) 4605{ 4606 // a track is considered presented when the total number of frames written to audio HAL 4607 // corresponds to the number of frames written when presentationComplete() is called for the 4608 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 4609 if (mPresentationCompleteFrames == 0) { 4610 mPresentationCompleteFrames = framesWritten + audioHalFrames; 4611 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 4612 mPresentationCompleteFrames, audioHalFrames); 4613 } 4614 if (framesWritten >= mPresentationCompleteFrames) { 4615 ALOGV("presentationComplete() session %d complete: framesWritten %d", 4616 mSessionId, framesWritten); 4617 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 4618 return true; 4619 } 4620 return false; 4621} 4622 4623void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 4624{ 4625 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 4626 if (mSyncEvents[i]->type() == type) { 4627 mSyncEvents[i]->trigger(); 4628 mSyncEvents.removeAt(i); 4629 i--; 4630 } 4631 } 4632} 4633 4634// implement VolumeBufferProvider interface 4635 4636uint32_t AudioFlinger::PlaybackThread::Track::getVolumeLR() 4637{ 4638 // called by FastMixer, so not allowed to take any locks, block, or do I/O including logs 4639 ALOG_ASSERT(isFastTrack() && (mCblk != NULL)); 4640 uint32_t vlr = mCblk->getVolumeLR(); 4641 uint32_t vl = vlr & 0xFFFF; 4642 uint32_t vr = vlr >> 16; 4643 // track volumes come from shared memory, so can't be trusted and must be clamped 4644 if (vl > MAX_GAIN_INT) { 4645 vl = MAX_GAIN_INT; 4646 } 4647 if (vr > MAX_GAIN_INT) { 4648 vr = MAX_GAIN_INT; 4649 } 4650 // now apply the cached master volume and stream type volume; 4651 // this is trusted but lacks any synchronization or barrier so may be stale 4652 float v = mCachedVolume; 4653 vl *= v; 4654 vr *= v; 4655 // re-combine into U4.16 4656 vlr = (vr << 16) | (vl & 0xFFFF); 4657 // FIXME look at mute, pause, and stop flags 4658 return vlr; 4659} 4660 4661status_t AudioFlinger::PlaybackThread::Track::setSyncEvent(const sp<SyncEvent>& event) 4662{ 4663 if (mState == TERMINATED || mState == PAUSED || 4664 ((framesReady() == 0) && ((mSharedBuffer != 0) || 4665 (mState == STOPPED)))) { 4666 ALOGW("Track::setSyncEvent() in invalid state %d on session %d %s mode, framesReady %d ", 4667 mState, mSessionId, (mSharedBuffer != 0) ? "static" : "stream", framesReady()); 4668 event->cancel(); 4669 return INVALID_OPERATION; 4670 } 4671 TrackBase::setSyncEvent(event); 4672 return NO_ERROR; 4673} 4674 4675// timed audio tracks 4676 4677sp<AudioFlinger::PlaybackThread::TimedTrack> 4678AudioFlinger::PlaybackThread::TimedTrack::create( 4679 PlaybackThread *thread, 4680 const sp<Client>& client, 4681 audio_stream_type_t streamType, 4682 uint32_t sampleRate, 4683 audio_format_t format, 4684 uint32_t channelMask, 4685 int frameCount, 4686 const sp<IMemory>& sharedBuffer, 4687 int sessionId) { 4688 if (!client->reserveTimedTrack()) 4689 return NULL; 4690 4691 return new TimedTrack( 4692 thread, client, streamType, sampleRate, format, channelMask, frameCount, 4693 sharedBuffer, sessionId); 4694} 4695 4696AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 4697 PlaybackThread *thread, 4698 const sp<Client>& client, 4699 audio_stream_type_t streamType, 4700 uint32_t sampleRate, 4701 audio_format_t format, 4702 uint32_t channelMask, 4703 int frameCount, 4704 const sp<IMemory>& sharedBuffer, 4705 int sessionId) 4706 : Track(thread, client, streamType, sampleRate, format, channelMask, 4707 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 4708 mQueueHeadInFlight(false), 4709 mTrimQueueHeadOnRelease(false), 4710 mFramesPendingInQueue(0), 4711 mTimedSilenceBuffer(NULL), 4712 mTimedSilenceBufferSize(0), 4713 mTimedAudioOutputOnTime(false), 4714 mMediaTimeTransformValid(false) 4715{ 4716 LocalClock lc; 4717 mLocalTimeFreq = lc.getLocalFreq(); 4718 4719 mLocalTimeToSampleTransform.a_zero = 0; 4720 mLocalTimeToSampleTransform.b_zero = 0; 4721 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 4722 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 4723 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 4724 &mLocalTimeToSampleTransform.a_to_b_denom); 4725 4726 mMediaTimeToSampleTransform.a_zero = 0; 4727 mMediaTimeToSampleTransform.b_zero = 0; 4728 mMediaTimeToSampleTransform.a_to_b_numer = sampleRate; 4729 mMediaTimeToSampleTransform.a_to_b_denom = 1000000; 4730 LinearTransform::reduce(&mMediaTimeToSampleTransform.a_to_b_numer, 4731 &mMediaTimeToSampleTransform.a_to_b_denom); 4732} 4733 4734AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 4735 mClient->releaseTimedTrack(); 4736 delete [] mTimedSilenceBuffer; 4737} 4738 4739status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 4740 size_t size, sp<IMemory>* buffer) { 4741 4742 Mutex::Autolock _l(mTimedBufferQueueLock); 4743 4744 trimTimedBufferQueue_l(); 4745 4746 // lazily initialize the shared memory heap for timed buffers 4747 if (mTimedMemoryDealer == NULL) { 4748 const int kTimedBufferHeapSize = 512 << 10; 4749 4750 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 4751 "AudioFlingerTimed"); 4752 if (mTimedMemoryDealer == NULL) 4753 return NO_MEMORY; 4754 } 4755 4756 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 4757 if (newBuffer == NULL) { 4758 newBuffer = mTimedMemoryDealer->allocate(size); 4759 if (newBuffer == NULL) 4760 return NO_MEMORY; 4761 } 4762 4763 *buffer = newBuffer; 4764 return NO_ERROR; 4765} 4766 4767// caller must hold mTimedBufferQueueLock 4768void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 4769 int64_t mediaTimeNow; 4770 { 4771 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4772 if (!mMediaTimeTransformValid) 4773 return; 4774 4775 int64_t targetTimeNow; 4776 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 4777 ? mCCHelper.getCommonTime(&targetTimeNow) 4778 : mCCHelper.getLocalTime(&targetTimeNow); 4779 4780 if (OK != res) 4781 return; 4782 4783 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 4784 &mediaTimeNow)) { 4785 return; 4786 } 4787 } 4788 4789 size_t trimEnd; 4790 for (trimEnd = 0; trimEnd < mTimedBufferQueue.size(); trimEnd++) { 4791 int64_t bufEnd; 4792 4793 if ((trimEnd + 1) < mTimedBufferQueue.size()) { 4794 // We have a next buffer. Just use its PTS as the PTS of the frame 4795 // following the last frame in this buffer. If the stream is sparse 4796 // (ie, there are deliberate gaps left in the stream which should be 4797 // filled with silence by the TimedAudioTrack), then this can result 4798 // in one extra buffer being left un-trimmed when it could have 4799 // been. In general, this is not typical, and we would rather 4800 // optimized away the TS calculation below for the more common case 4801 // where PTSes are contiguous. 4802 bufEnd = mTimedBufferQueue[trimEnd + 1].pts(); 4803 } else { 4804 // We have no next buffer. Compute the PTS of the frame following 4805 // the last frame in this buffer by computing the duration of of 4806 // this frame in media time units and adding it to the PTS of the 4807 // buffer. 4808 int64_t frameCount = mTimedBufferQueue[trimEnd].buffer()->size() 4809 / mCblk->frameSize; 4810 4811 if (!mMediaTimeToSampleTransform.doReverseTransform(frameCount, 4812 &bufEnd)) { 4813 ALOGE("Failed to convert frame count of %lld to media time" 4814 " duration" " (scale factor %d/%u) in %s", 4815 frameCount, 4816 mMediaTimeToSampleTransform.a_to_b_numer, 4817 mMediaTimeToSampleTransform.a_to_b_denom, 4818 __PRETTY_FUNCTION__); 4819 break; 4820 } 4821 bufEnd += mTimedBufferQueue[trimEnd].pts(); 4822 } 4823 4824 if (bufEnd > mediaTimeNow) 4825 break; 4826 4827 // Is the buffer we want to use in the middle of a mix operation right 4828 // now? If so, don't actually trim it. Just wait for the releaseBuffer 4829 // from the mixer which should be coming back shortly. 4830 if (!trimEnd && mQueueHeadInFlight) { 4831 mTrimQueueHeadOnRelease = true; 4832 } 4833 } 4834 4835 size_t trimStart = mTrimQueueHeadOnRelease ? 1 : 0; 4836 if (trimStart < trimEnd) { 4837 // Update the bookkeeping for framesReady() 4838 for (size_t i = trimStart; i < trimEnd; ++i) { 4839 updateFramesPendingAfterTrim_l(mTimedBufferQueue[i], "trim"); 4840 } 4841 4842 // Now actually remove the buffers from the queue. 4843 mTimedBufferQueue.removeItemsAt(trimStart, trimEnd); 4844 } 4845} 4846 4847void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueueHead_l( 4848 const char* logTag) { 4849 ALOG_ASSERT(mTimedBufferQueue.size() > 0, 4850 "%s called (reason \"%s\"), but timed buffer queue has no" 4851 " elements to trim.", __FUNCTION__, logTag); 4852 4853 updateFramesPendingAfterTrim_l(mTimedBufferQueue[0], logTag); 4854 mTimedBufferQueue.removeAt(0); 4855} 4856 4857void AudioFlinger::PlaybackThread::TimedTrack::updateFramesPendingAfterTrim_l( 4858 const TimedBuffer& buf, 4859 const char* logTag) { 4860 uint32_t bufBytes = buf.buffer()->size(); 4861 uint32_t consumedAlready = buf.position(); 4862 4863 ALOG_ASSERT(consumedAlready <= bufBytes, 4864 "Bad bookkeeping while updating frames pending. Timed buffer is" 4865 " only %u bytes long, but claims to have consumed %u" 4866 " bytes. (update reason: \"%s\")", 4867 bufBytes, consumedAlready, logTag); 4868 4869 uint32_t bufFrames = (bufBytes - consumedAlready) / mCblk->frameSize; 4870 ALOG_ASSERT(mFramesPendingInQueue >= bufFrames, 4871 "Bad bookkeeping while updating frames pending. Should have at" 4872 " least %u queued frames, but we think we have only %u. (update" 4873 " reason: \"%s\")", 4874 bufFrames, mFramesPendingInQueue, logTag); 4875 4876 mFramesPendingInQueue -= bufFrames; 4877} 4878 4879status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 4880 const sp<IMemory>& buffer, int64_t pts) { 4881 4882 { 4883 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4884 if (!mMediaTimeTransformValid) 4885 return INVALID_OPERATION; 4886 } 4887 4888 Mutex::Autolock _l(mTimedBufferQueueLock); 4889 4890 uint32_t bufFrames = buffer->size() / mCblk->frameSize; 4891 mFramesPendingInQueue += bufFrames; 4892 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 4893 4894 return NO_ERROR; 4895} 4896 4897status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 4898 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 4899 4900 ALOGVV("setMediaTimeTransform az=%lld bz=%lld n=%d d=%u tgt=%d", 4901 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4902 target); 4903 4904 if (!(target == TimedAudioTrack::LOCAL_TIME || 4905 target == TimedAudioTrack::COMMON_TIME)) { 4906 return BAD_VALUE; 4907 } 4908 4909 Mutex::Autolock lock(mMediaTimeTransformLock); 4910 mMediaTimeTransform = xform; 4911 mMediaTimeTransformTarget = target; 4912 mMediaTimeTransformValid = true; 4913 4914 return NO_ERROR; 4915} 4916 4917#define min(a, b) ((a) < (b) ? (a) : (b)) 4918 4919// implementation of getNextBuffer for tracks whose buffers have timestamps 4920status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4921 AudioBufferProvider::Buffer* buffer, int64_t pts) 4922{ 4923 if (pts == AudioBufferProvider::kInvalidPTS) { 4924 buffer->raw = 0; 4925 buffer->frameCount = 0; 4926 mTimedAudioOutputOnTime = false; 4927 return INVALID_OPERATION; 4928 } 4929 4930 Mutex::Autolock _l(mTimedBufferQueueLock); 4931 4932 ALOG_ASSERT(!mQueueHeadInFlight, 4933 "getNextBuffer called without releaseBuffer!"); 4934 4935 while (true) { 4936 4937 // if we have no timed buffers, then fail 4938 if (mTimedBufferQueue.isEmpty()) { 4939 buffer->raw = 0; 4940 buffer->frameCount = 0; 4941 return NOT_ENOUGH_DATA; 4942 } 4943 4944 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4945 4946 // calculate the PTS of the head of the timed buffer queue expressed in 4947 // local time 4948 int64_t headLocalPTS; 4949 { 4950 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4951 4952 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4953 4954 if (mMediaTimeTransform.a_to_b_denom == 0) { 4955 // the transform represents a pause, so yield silence 4956 timedYieldSilence_l(buffer->frameCount, buffer); 4957 return NO_ERROR; 4958 } 4959 4960 int64_t transformedPTS; 4961 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4962 &transformedPTS)) { 4963 // the transform failed. this shouldn't happen, but if it does 4964 // then just drop this buffer 4965 ALOGW("timedGetNextBuffer transform failed"); 4966 buffer->raw = 0; 4967 buffer->frameCount = 0; 4968 trimTimedBufferQueueHead_l("getNextBuffer; no transform"); 4969 return NO_ERROR; 4970 } 4971 4972 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4973 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4974 &headLocalPTS)) { 4975 buffer->raw = 0; 4976 buffer->frameCount = 0; 4977 return INVALID_OPERATION; 4978 } 4979 } else { 4980 headLocalPTS = transformedPTS; 4981 } 4982 } 4983 4984 // adjust the head buffer's PTS to reflect the portion of the head buffer 4985 // that has already been consumed 4986 int64_t effectivePTS = headLocalPTS + 4987 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4988 4989 // Calculate the delta in samples between the head of the input buffer 4990 // queue and the start of the next output buffer that will be written. 4991 // If the transformation fails because of over or underflow, it means 4992 // that the sample's position in the output stream is so far out of 4993 // whack that it should just be dropped. 4994 int64_t sampleDelta; 4995 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4996 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4997 trimTimedBufferQueueHead_l("getNextBuffer, buf pts too far from" 4998 " mix"); 4999 continue; 5000 } 5001 if (!mLocalTimeToSampleTransform.doForwardTransform( 5002 (effectivePTS - pts) << 32, &sampleDelta)) { 5003 ALOGV("*** too late during sample rate transform: dropped buffer"); 5004 trimTimedBufferQueueHead_l("getNextBuffer, bad local to sample"); 5005 continue; 5006 } 5007 5008 ALOGVV("*** getNextBuffer head.pts=%lld head.pos=%d pts=%lld" 5009 " sampleDelta=[%d.%08x]", 5010 head.pts(), head.position(), pts, 5011 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) 5012 + (sampleDelta >> 32)), 5013 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 5014 5015 // if the delta between the ideal placement for the next input sample and 5016 // the current output position is within this threshold, then we will 5017 // concatenate the next input samples to the previous output 5018 const int64_t kSampleContinuityThreshold = 5019 (static_cast<int64_t>(sampleRate()) << 32) / 250; 5020 5021 // if this is the first buffer of audio that we're emitting from this track 5022 // then it should be almost exactly on time. 5023 const int64_t kSampleStartupThreshold = 1LL << 32; 5024 5025 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 5026 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 5027 // the next input is close enough to being on time, so concatenate it 5028 // with the last output 5029 timedYieldSamples_l(buffer); 5030 5031 ALOGVV("*** on time: head.pos=%d frameCount=%u", 5032 head.position(), buffer->frameCount); 5033 return NO_ERROR; 5034 } 5035 5036 // Looks like our output is not on time. Reset our on timed status. 5037 // Next time we mix samples from our input queue, then should be within 5038 // the StartupThreshold. 5039 mTimedAudioOutputOnTime = false; 5040 if (sampleDelta > 0) { 5041 // the gap between the current output position and the proper start of 5042 // the next input sample is too big, so fill it with silence 5043 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 5044 5045 timedYieldSilence_l(framesUntilNextInput, buffer); 5046 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 5047 return NO_ERROR; 5048 } else { 5049 // the next input sample is late 5050 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 5051 size_t onTimeSamplePosition = 5052 head.position() + lateFrames * mCblk->frameSize; 5053 5054 if (onTimeSamplePosition > head.buffer()->size()) { 5055 // all the remaining samples in the head are too late, so 5056 // drop it and move on 5057 ALOGV("*** too late: dropped buffer"); 5058 trimTimedBufferQueueHead_l("getNextBuffer, dropped late buffer"); 5059 continue; 5060 } else { 5061 // skip over the late samples 5062 head.setPosition(onTimeSamplePosition); 5063 5064 // yield the available samples 5065 timedYieldSamples_l(buffer); 5066 5067 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 5068 return NO_ERROR; 5069 } 5070 } 5071 } 5072} 5073 5074// Yield samples from the timed buffer queue head up to the given output 5075// buffer's capacity. 5076// 5077// Caller must hold mTimedBufferQueueLock 5078void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples_l( 5079 AudioBufferProvider::Buffer* buffer) { 5080 5081 const TimedBuffer& head = mTimedBufferQueue[0]; 5082 5083 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 5084 head.position()); 5085 5086 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 5087 mCblk->frameSize); 5088 size_t framesRequested = buffer->frameCount; 5089 buffer->frameCount = min(framesLeftInHead, framesRequested); 5090 5091 mQueueHeadInFlight = true; 5092 mTimedAudioOutputOnTime = true; 5093} 5094 5095// Yield samples of silence up to the given output buffer's capacity 5096// 5097// Caller must hold mTimedBufferQueueLock 5098void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence_l( 5099 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 5100 5101 // lazily allocate a buffer filled with silence 5102 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 5103 delete [] mTimedSilenceBuffer; 5104 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 5105 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 5106 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 5107 } 5108 5109 buffer->raw = mTimedSilenceBuffer; 5110 size_t framesRequested = buffer->frameCount; 5111 buffer->frameCount = min(numFrames, framesRequested); 5112 5113 mTimedAudioOutputOnTime = false; 5114} 5115 5116// AudioBufferProvider interface 5117void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 5118 AudioBufferProvider::Buffer* buffer) { 5119 5120 Mutex::Autolock _l(mTimedBufferQueueLock); 5121 5122 // If the buffer which was just released is part of the buffer at the head 5123 // of the queue, be sure to update the amt of the buffer which has been 5124 // consumed. If the buffer being returned is not part of the head of the 5125 // queue, its either because the buffer is part of the silence buffer, or 5126 // because the head of the timed queue was trimmed after the mixer called 5127 // getNextBuffer but before the mixer called releaseBuffer. 5128 if (buffer->raw == mTimedSilenceBuffer) { 5129 ALOG_ASSERT(!mQueueHeadInFlight, 5130 "Queue head in flight during release of silence buffer!"); 5131 goto done; 5132 } 5133 5134 ALOG_ASSERT(mQueueHeadInFlight, 5135 "TimedTrack::releaseBuffer of non-silence buffer, but no queue" 5136 " head in flight."); 5137 5138 if (mTimedBufferQueue.size()) { 5139 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 5140 5141 void* start = head.buffer()->pointer(); 5142 void* end = reinterpret_cast<void*>( 5143 reinterpret_cast<uint8_t*>(head.buffer()->pointer()) 5144 + head.buffer()->size()); 5145 5146 ALOG_ASSERT((buffer->raw >= start) && (buffer->raw < end), 5147 "released buffer not within the head of the timed buffer" 5148 " queue; qHead = [%p, %p], released buffer = %p", 5149 start, end, buffer->raw); 5150 5151 head.setPosition(head.position() + 5152 (buffer->frameCount * mCblk->frameSize)); 5153 mQueueHeadInFlight = false; 5154 5155 ALOG_ASSERT(mFramesPendingInQueue >= buffer->frameCount, 5156 "Bad bookkeeping during releaseBuffer! Should have at" 5157 " least %u queued frames, but we think we have only %u", 5158 buffer->frameCount, mFramesPendingInQueue); 5159 5160 mFramesPendingInQueue -= buffer->frameCount; 5161 5162 if ((static_cast<size_t>(head.position()) >= head.buffer()->size()) 5163 || mTrimQueueHeadOnRelease) { 5164 trimTimedBufferQueueHead_l("releaseBuffer"); 5165 mTrimQueueHeadOnRelease = false; 5166 } 5167 } else { 5168 LOG_FATAL("TimedTrack::releaseBuffer of non-silence buffer with no" 5169 " buffers in the timed buffer queue"); 5170 } 5171 5172done: 5173 buffer->raw = 0; 5174 buffer->frameCount = 0; 5175} 5176 5177size_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 5178 Mutex::Autolock _l(mTimedBufferQueueLock); 5179 return mFramesPendingInQueue; 5180} 5181 5182AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 5183 : mPTS(0), mPosition(0) {} 5184 5185AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 5186 const sp<IMemory>& buffer, int64_t pts) 5187 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 5188 5189// ---------------------------------------------------------------------------- 5190 5191// RecordTrack constructor must be called with AudioFlinger::mLock held 5192AudioFlinger::RecordThread::RecordTrack::RecordTrack( 5193 RecordThread *thread, 5194 const sp<Client>& client, 5195 uint32_t sampleRate, 5196 audio_format_t format, 5197 uint32_t channelMask, 5198 int frameCount, 5199 int sessionId) 5200 : TrackBase(thread, client, sampleRate, format, 5201 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 5202 mOverflow(false) 5203{ 5204 if (mCblk != NULL) { 5205 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 5206 if (format == AUDIO_FORMAT_PCM_16_BIT) { 5207 mCblk->frameSize = mChannelCount * sizeof(int16_t); 5208 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 5209 mCblk->frameSize = mChannelCount * sizeof(int8_t); 5210 } else { 5211 mCblk->frameSize = sizeof(int8_t); 5212 } 5213 } 5214} 5215 5216AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 5217{ 5218 sp<ThreadBase> thread = mThread.promote(); 5219 if (thread != 0) { 5220 AudioSystem::releaseInput(thread->id()); 5221 } 5222} 5223 5224// AudioBufferProvider interface 5225status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5226{ 5227 audio_track_cblk_t* cblk = this->cblk(); 5228 uint32_t framesAvail; 5229 uint32_t framesReq = buffer->frameCount; 5230 5231 // Check if last stepServer failed, try to step now 5232 if (mStepServerFailed) { 5233 if (!step()) goto getNextBuffer_exit; 5234 ALOGV("stepServer recovered"); 5235 mStepServerFailed = false; 5236 } 5237 5238 framesAvail = cblk->framesAvailable_l(); 5239 5240 if (CC_LIKELY(framesAvail)) { 5241 uint32_t s = cblk->server; 5242 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 5243 5244 if (framesReq > framesAvail) { 5245 framesReq = framesAvail; 5246 } 5247 if (framesReq > bufferEnd - s) { 5248 framesReq = bufferEnd - s; 5249 } 5250 5251 buffer->raw = getBuffer(s, framesReq); 5252 if (buffer->raw == NULL) goto getNextBuffer_exit; 5253 5254 buffer->frameCount = framesReq; 5255 return NO_ERROR; 5256 } 5257 5258getNextBuffer_exit: 5259 buffer->raw = NULL; 5260 buffer->frameCount = 0; 5261 return NOT_ENOUGH_DATA; 5262} 5263 5264status_t AudioFlinger::RecordThread::RecordTrack::start(AudioSystem::sync_event_t event, 5265 int triggerSession) 5266{ 5267 sp<ThreadBase> thread = mThread.promote(); 5268 if (thread != 0) { 5269 RecordThread *recordThread = (RecordThread *)thread.get(); 5270 return recordThread->start(this, event, triggerSession); 5271 } else { 5272 return BAD_VALUE; 5273 } 5274} 5275 5276void AudioFlinger::RecordThread::RecordTrack::stop() 5277{ 5278 sp<ThreadBase> thread = mThread.promote(); 5279 if (thread != 0) { 5280 RecordThread *recordThread = (RecordThread *)thread.get(); 5281 recordThread->stop(this); 5282 TrackBase::reset(); 5283 // Force overrun condition to avoid false overrun callback until first data is 5284 // read from buffer 5285 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 5286 } 5287} 5288 5289void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 5290{ 5291 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 5292 (mClient == 0) ? getpid_cached : mClient->pid(), 5293 mFormat, 5294 mChannelMask, 5295 mSessionId, 5296 mFrameCount, 5297 mState, 5298 mCblk->sampleRate, 5299 mCblk->server, 5300 mCblk->user); 5301} 5302 5303 5304// ---------------------------------------------------------------------------- 5305 5306AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 5307 PlaybackThread *playbackThread, 5308 DuplicatingThread *sourceThread, 5309 uint32_t sampleRate, 5310 audio_format_t format, 5311 uint32_t channelMask, 5312 int frameCount) 5313 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 5314 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 5315 mActive(false), mSourceThread(sourceThread) 5316{ 5317 5318 if (mCblk != NULL) { 5319 mCblk->flags |= CBLK_DIRECTION_OUT; 5320 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 5321 mOutBuffer.frameCount = 0; 5322 playbackThread->mTracks.add(this); 5323 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 5324 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 5325 mCblk, mBuffer, mCblk->buffers, 5326 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 5327 } else { 5328 ALOGW("Error creating output track on thread %p", playbackThread); 5329 } 5330} 5331 5332AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 5333{ 5334 clearBufferQueue(); 5335} 5336 5337status_t AudioFlinger::PlaybackThread::OutputTrack::start(AudioSystem::sync_event_t event, 5338 int triggerSession) 5339{ 5340 status_t status = Track::start(event, triggerSession); 5341 if (status != NO_ERROR) { 5342 return status; 5343 } 5344 5345 mActive = true; 5346 mRetryCount = 127; 5347 return status; 5348} 5349 5350void AudioFlinger::PlaybackThread::OutputTrack::stop() 5351{ 5352 Track::stop(); 5353 clearBufferQueue(); 5354 mOutBuffer.frameCount = 0; 5355 mActive = false; 5356} 5357 5358bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 5359{ 5360 Buffer *pInBuffer; 5361 Buffer inBuffer; 5362 uint32_t channelCount = mChannelCount; 5363 bool outputBufferFull = false; 5364 inBuffer.frameCount = frames; 5365 inBuffer.i16 = data; 5366 5367 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 5368 5369 if (!mActive && frames != 0) { 5370 start(); 5371 sp<ThreadBase> thread = mThread.promote(); 5372 if (thread != 0) { 5373 MixerThread *mixerThread = (MixerThread *)thread.get(); 5374 if (mCblk->frameCount > frames){ 5375 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5376 uint32_t startFrames = (mCblk->frameCount - frames); 5377 pInBuffer = new Buffer; 5378 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 5379 pInBuffer->frameCount = startFrames; 5380 pInBuffer->i16 = pInBuffer->mBuffer; 5381 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 5382 mBufferQueue.add(pInBuffer); 5383 } else { 5384 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 5385 } 5386 } 5387 } 5388 } 5389 5390 while (waitTimeLeftMs) { 5391 // First write pending buffers, then new data 5392 if (mBufferQueue.size()) { 5393 pInBuffer = mBufferQueue.itemAt(0); 5394 } else { 5395 pInBuffer = &inBuffer; 5396 } 5397 5398 if (pInBuffer->frameCount == 0) { 5399 break; 5400 } 5401 5402 if (mOutBuffer.frameCount == 0) { 5403 mOutBuffer.frameCount = pInBuffer->frameCount; 5404 nsecs_t startTime = systemTime(); 5405 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 5406 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 5407 outputBufferFull = true; 5408 break; 5409 } 5410 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 5411 if (waitTimeLeftMs >= waitTimeMs) { 5412 waitTimeLeftMs -= waitTimeMs; 5413 } else { 5414 waitTimeLeftMs = 0; 5415 } 5416 } 5417 5418 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 5419 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 5420 mCblk->stepUser(outFrames); 5421 pInBuffer->frameCount -= outFrames; 5422 pInBuffer->i16 += outFrames * channelCount; 5423 mOutBuffer.frameCount -= outFrames; 5424 mOutBuffer.i16 += outFrames * channelCount; 5425 5426 if (pInBuffer->frameCount == 0) { 5427 if (mBufferQueue.size()) { 5428 mBufferQueue.removeAt(0); 5429 delete [] pInBuffer->mBuffer; 5430 delete pInBuffer; 5431 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5432 } else { 5433 break; 5434 } 5435 } 5436 } 5437 5438 // If we could not write all frames, allocate a buffer and queue it for next time. 5439 if (inBuffer.frameCount) { 5440 sp<ThreadBase> thread = mThread.promote(); 5441 if (thread != 0 && !thread->standby()) { 5442 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 5443 pInBuffer = new Buffer; 5444 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 5445 pInBuffer->frameCount = inBuffer.frameCount; 5446 pInBuffer->i16 = pInBuffer->mBuffer; 5447 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 5448 mBufferQueue.add(pInBuffer); 5449 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 5450 } else { 5451 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 5452 } 5453 } 5454 } 5455 5456 // Calling write() with a 0 length buffer, means that no more data will be written: 5457 // If no more buffers are pending, fill output track buffer to make sure it is started 5458 // by output mixer. 5459 if (frames == 0 && mBufferQueue.size() == 0) { 5460 if (mCblk->user < mCblk->frameCount) { 5461 frames = mCblk->frameCount - mCblk->user; 5462 pInBuffer = new Buffer; 5463 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 5464 pInBuffer->frameCount = frames; 5465 pInBuffer->i16 = pInBuffer->mBuffer; 5466 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 5467 mBufferQueue.add(pInBuffer); 5468 } else if (mActive) { 5469 stop(); 5470 } 5471 } 5472 5473 return outputBufferFull; 5474} 5475 5476status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 5477{ 5478 int active; 5479 status_t result; 5480 audio_track_cblk_t* cblk = mCblk; 5481 uint32_t framesReq = buffer->frameCount; 5482 5483// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 5484 buffer->frameCount = 0; 5485 5486 uint32_t framesAvail = cblk->framesAvailable(); 5487 5488 5489 if (framesAvail == 0) { 5490 Mutex::Autolock _l(cblk->lock); 5491 goto start_loop_here; 5492 while (framesAvail == 0) { 5493 active = mActive; 5494 if (CC_UNLIKELY(!active)) { 5495 ALOGV("Not active and NO_MORE_BUFFERS"); 5496 return NO_MORE_BUFFERS; 5497 } 5498 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 5499 if (result != NO_ERROR) { 5500 return NO_MORE_BUFFERS; 5501 } 5502 // read the server count again 5503 start_loop_here: 5504 framesAvail = cblk->framesAvailable_l(); 5505 } 5506 } 5507 5508// if (framesAvail < framesReq) { 5509// return NO_MORE_BUFFERS; 5510// } 5511 5512 if (framesReq > framesAvail) { 5513 framesReq = framesAvail; 5514 } 5515 5516 uint32_t u = cblk->user; 5517 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 5518 5519 if (framesReq > bufferEnd - u) { 5520 framesReq = bufferEnd - u; 5521 } 5522 5523 buffer->frameCount = framesReq; 5524 buffer->raw = (void *)cblk->buffer(u); 5525 return NO_ERROR; 5526} 5527 5528 5529void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 5530{ 5531 size_t size = mBufferQueue.size(); 5532 5533 for (size_t i = 0; i < size; i++) { 5534 Buffer *pBuffer = mBufferQueue.itemAt(i); 5535 delete [] pBuffer->mBuffer; 5536 delete pBuffer; 5537 } 5538 mBufferQueue.clear(); 5539} 5540 5541// ---------------------------------------------------------------------------- 5542 5543AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 5544 : RefBase(), 5545 mAudioFlinger(audioFlinger), 5546 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 5547 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 5548 mPid(pid), 5549 mTimedTrackCount(0) 5550{ 5551 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 5552} 5553 5554// Client destructor must be called with AudioFlinger::mLock held 5555AudioFlinger::Client::~Client() 5556{ 5557 mAudioFlinger->removeClient_l(mPid); 5558} 5559 5560sp<MemoryDealer> AudioFlinger::Client::heap() const 5561{ 5562 return mMemoryDealer; 5563} 5564 5565// Reserve one of the limited slots for a timed audio track associated 5566// with this client 5567bool AudioFlinger::Client::reserveTimedTrack() 5568{ 5569 const int kMaxTimedTracksPerClient = 4; 5570 5571 Mutex::Autolock _l(mTimedTrackLock); 5572 5573 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 5574 ALOGW("can not create timed track - pid %d has exceeded the limit", 5575 mPid); 5576 return false; 5577 } 5578 5579 mTimedTrackCount++; 5580 return true; 5581} 5582 5583// Release a slot for a timed audio track 5584void AudioFlinger::Client::releaseTimedTrack() 5585{ 5586 Mutex::Autolock _l(mTimedTrackLock); 5587 mTimedTrackCount--; 5588} 5589 5590// ---------------------------------------------------------------------------- 5591 5592AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 5593 const sp<IAudioFlingerClient>& client, 5594 pid_t pid) 5595 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 5596{ 5597} 5598 5599AudioFlinger::NotificationClient::~NotificationClient() 5600{ 5601} 5602 5603void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 5604{ 5605 sp<NotificationClient> keep(this); 5606 mAudioFlinger->removeNotificationClient(mPid); 5607} 5608 5609// ---------------------------------------------------------------------------- 5610 5611AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 5612 : BnAudioTrack(), 5613 mTrack(track) 5614{ 5615} 5616 5617AudioFlinger::TrackHandle::~TrackHandle() { 5618 // just stop the track on deletion, associated resources 5619 // will be freed from the main thread once all pending buffers have 5620 // been played. Unless it's not in the active track list, in which 5621 // case we free everything now... 5622 mTrack->destroy(); 5623} 5624 5625sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 5626 return mTrack->getCblk(); 5627} 5628 5629status_t AudioFlinger::TrackHandle::start() { 5630 return mTrack->start(); 5631} 5632 5633void AudioFlinger::TrackHandle::stop() { 5634 mTrack->stop(); 5635} 5636 5637void AudioFlinger::TrackHandle::flush() { 5638 mTrack->flush(); 5639} 5640 5641void AudioFlinger::TrackHandle::mute(bool e) { 5642 mTrack->mute(e); 5643} 5644 5645void AudioFlinger::TrackHandle::pause() { 5646 mTrack->pause(); 5647} 5648 5649status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 5650{ 5651 return mTrack->attachAuxEffect(EffectId); 5652} 5653 5654status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 5655 sp<IMemory>* buffer) { 5656 if (!mTrack->isTimedTrack()) 5657 return INVALID_OPERATION; 5658 5659 PlaybackThread::TimedTrack* tt = 5660 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5661 return tt->allocateTimedBuffer(size, buffer); 5662} 5663 5664status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 5665 int64_t pts) { 5666 if (!mTrack->isTimedTrack()) 5667 return INVALID_OPERATION; 5668 5669 PlaybackThread::TimedTrack* tt = 5670 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5671 return tt->queueTimedBuffer(buffer, pts); 5672} 5673 5674status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 5675 const LinearTransform& xform, int target) { 5676 5677 if (!mTrack->isTimedTrack()) 5678 return INVALID_OPERATION; 5679 5680 PlaybackThread::TimedTrack* tt = 5681 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 5682 return tt->setMediaTimeTransform( 5683 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 5684} 5685 5686status_t AudioFlinger::TrackHandle::onTransact( 5687 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5688{ 5689 return BnAudioTrack::onTransact(code, data, reply, flags); 5690} 5691 5692// ---------------------------------------------------------------------------- 5693 5694sp<IAudioRecord> AudioFlinger::openRecord( 5695 pid_t pid, 5696 audio_io_handle_t input, 5697 uint32_t sampleRate, 5698 audio_format_t format, 5699 uint32_t channelMask, 5700 int frameCount, 5701 IAudioFlinger::track_flags_t flags, 5702 int *sessionId, 5703 status_t *status) 5704{ 5705 sp<RecordThread::RecordTrack> recordTrack; 5706 sp<RecordHandle> recordHandle; 5707 sp<Client> client; 5708 status_t lStatus; 5709 RecordThread *thread; 5710 size_t inFrameCount; 5711 int lSessionId; 5712 5713 // check calling permissions 5714 if (!recordingAllowed()) { 5715 lStatus = PERMISSION_DENIED; 5716 goto Exit; 5717 } 5718 5719 // add client to list 5720 { // scope for mLock 5721 Mutex::Autolock _l(mLock); 5722 thread = checkRecordThread_l(input); 5723 if (thread == NULL) { 5724 lStatus = BAD_VALUE; 5725 goto Exit; 5726 } 5727 5728 client = registerPid_l(pid); 5729 5730 // If no audio session id is provided, create one here 5731 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 5732 lSessionId = *sessionId; 5733 } else { 5734 lSessionId = nextUniqueId(); 5735 if (sessionId != NULL) { 5736 *sessionId = lSessionId; 5737 } 5738 } 5739 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 5740 recordTrack = thread->createRecordTrack_l(client, 5741 sampleRate, 5742 format, 5743 channelMask, 5744 frameCount, 5745 lSessionId, 5746 &lStatus); 5747 } 5748 if (lStatus != NO_ERROR) { 5749 // remove local strong reference to Client before deleting the RecordTrack so that the Client 5750 // destructor is called by the TrackBase destructor with mLock held 5751 client.clear(); 5752 recordTrack.clear(); 5753 goto Exit; 5754 } 5755 5756 // return to handle to client 5757 recordHandle = new RecordHandle(recordTrack); 5758 lStatus = NO_ERROR; 5759 5760Exit: 5761 if (status) { 5762 *status = lStatus; 5763 } 5764 return recordHandle; 5765} 5766 5767// ---------------------------------------------------------------------------- 5768 5769AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 5770 : BnAudioRecord(), 5771 mRecordTrack(recordTrack) 5772{ 5773} 5774 5775AudioFlinger::RecordHandle::~RecordHandle() { 5776 stop(); 5777} 5778 5779sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 5780 return mRecordTrack->getCblk(); 5781} 5782 5783status_t AudioFlinger::RecordHandle::start(int event, int triggerSession) { 5784 ALOGV("RecordHandle::start()"); 5785 return mRecordTrack->start((AudioSystem::sync_event_t)event, triggerSession); 5786} 5787 5788void AudioFlinger::RecordHandle::stop() { 5789 ALOGV("RecordHandle::stop()"); 5790 mRecordTrack->stop(); 5791} 5792 5793status_t AudioFlinger::RecordHandle::onTransact( 5794 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 5795{ 5796 return BnAudioRecord::onTransact(code, data, reply, flags); 5797} 5798 5799// ---------------------------------------------------------------------------- 5800 5801AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 5802 AudioStreamIn *input, 5803 uint32_t sampleRate, 5804 uint32_t channels, 5805 audio_io_handle_t id, 5806 uint32_t device) : 5807 ThreadBase(audioFlinger, id, device, RECORD), 5808 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 5809 // mRsmpInIndex and mInputBytes set by readInputParameters() 5810 mReqChannelCount(popcount(channels)), 5811 mReqSampleRate(sampleRate) 5812 // mBytesRead is only meaningful while active, and so is cleared in start() 5813 // (but might be better to also clear here for dump?) 5814{ 5815 snprintf(mName, kNameLength, "AudioIn_%X", id); 5816 5817 readInputParameters(); 5818} 5819 5820 5821AudioFlinger::RecordThread::~RecordThread() 5822{ 5823 delete[] mRsmpInBuffer; 5824 delete mResampler; 5825 delete[] mRsmpOutBuffer; 5826} 5827 5828void AudioFlinger::RecordThread::onFirstRef() 5829{ 5830 run(mName, PRIORITY_URGENT_AUDIO); 5831} 5832 5833status_t AudioFlinger::RecordThread::readyToRun() 5834{ 5835 status_t status = initCheck(); 5836 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 5837 return status; 5838} 5839 5840bool AudioFlinger::RecordThread::threadLoop() 5841{ 5842 AudioBufferProvider::Buffer buffer; 5843 sp<RecordTrack> activeTrack; 5844 Vector< sp<EffectChain> > effectChains; 5845 5846 nsecs_t lastWarning = 0; 5847 5848 acquireWakeLock(); 5849 5850 // start recording 5851 while (!exitPending()) { 5852 5853 processConfigEvents(); 5854 5855 { // scope for mLock 5856 Mutex::Autolock _l(mLock); 5857 checkForNewParameters_l(); 5858 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 5859 if (!mStandby) { 5860 mInput->stream->common.standby(&mInput->stream->common); 5861 mStandby = true; 5862 } 5863 5864 if (exitPending()) break; 5865 5866 releaseWakeLock_l(); 5867 ALOGV("RecordThread: loop stopping"); 5868 // go to sleep 5869 mWaitWorkCV.wait(mLock); 5870 ALOGV("RecordThread: loop starting"); 5871 acquireWakeLock_l(); 5872 continue; 5873 } 5874 if (mActiveTrack != 0) { 5875 if (mActiveTrack->mState == TrackBase::PAUSING) { 5876 if (!mStandby) { 5877 mInput->stream->common.standby(&mInput->stream->common); 5878 mStandby = true; 5879 } 5880 mActiveTrack.clear(); 5881 mStartStopCond.broadcast(); 5882 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 5883 if (mReqChannelCount != mActiveTrack->channelCount()) { 5884 mActiveTrack.clear(); 5885 mStartStopCond.broadcast(); 5886 } else if (mBytesRead != 0) { 5887 // record start succeeds only if first read from audio input 5888 // succeeds 5889 if (mBytesRead > 0) { 5890 mActiveTrack->mState = TrackBase::ACTIVE; 5891 } else { 5892 mActiveTrack.clear(); 5893 } 5894 mStartStopCond.broadcast(); 5895 } 5896 mStandby = false; 5897 } 5898 } 5899 lockEffectChains_l(effectChains); 5900 } 5901 5902 if (mActiveTrack != 0) { 5903 if (mActiveTrack->mState != TrackBase::ACTIVE && 5904 mActiveTrack->mState != TrackBase::RESUMING) { 5905 unlockEffectChains(effectChains); 5906 usleep(kRecordThreadSleepUs); 5907 continue; 5908 } 5909 for (size_t i = 0; i < effectChains.size(); i ++) { 5910 effectChains[i]->process_l(); 5911 } 5912 5913 buffer.frameCount = mFrameCount; 5914 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 5915 size_t framesOut = buffer.frameCount; 5916 if (mResampler == NULL) { 5917 // no resampling 5918 while (framesOut) { 5919 size_t framesIn = mFrameCount - mRsmpInIndex; 5920 if (framesIn) { 5921 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 5922 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 5923 if (framesIn > framesOut) 5924 framesIn = framesOut; 5925 mRsmpInIndex += framesIn; 5926 framesOut -= framesIn; 5927 if ((int)mChannelCount == mReqChannelCount || 5928 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 5929 memcpy(dst, src, framesIn * mFrameSize); 5930 } else { 5931 int16_t *src16 = (int16_t *)src; 5932 int16_t *dst16 = (int16_t *)dst; 5933 if (mChannelCount == 1) { 5934 while (framesIn--) { 5935 *dst16++ = *src16; 5936 *dst16++ = *src16++; 5937 } 5938 } else { 5939 while (framesIn--) { 5940 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5941 src16 += 2; 5942 } 5943 } 5944 } 5945 } 5946 if (framesOut && mFrameCount == mRsmpInIndex) { 5947 if (framesOut == mFrameCount && 5948 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5949 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5950 framesOut = 0; 5951 } else { 5952 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5953 mRsmpInIndex = 0; 5954 } 5955 if (mBytesRead < 0) { 5956 ALOGE("Error reading audio input"); 5957 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5958 // Force input into standby so that it tries to 5959 // recover at next read attempt 5960 mInput->stream->common.standby(&mInput->stream->common); 5961 usleep(kRecordThreadSleepUs); 5962 } 5963 mRsmpInIndex = mFrameCount; 5964 framesOut = 0; 5965 buffer.frameCount = 0; 5966 } 5967 } 5968 } 5969 } else { 5970 // resampling 5971 5972 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5973 // alter output frame count as if we were expecting stereo samples 5974 if (mChannelCount == 1 && mReqChannelCount == 1) { 5975 framesOut >>= 1; 5976 } 5977 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5978 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5979 // are 32 bit aligned which should be always true. 5980 if (mChannelCount == 2 && mReqChannelCount == 1) { 5981 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5982 // the resampler always outputs stereo samples: do post stereo to mono conversion 5983 int16_t *src = (int16_t *)mRsmpOutBuffer; 5984 int16_t *dst = buffer.i16; 5985 while (framesOut--) { 5986 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5987 src += 2; 5988 } 5989 } else { 5990 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5991 } 5992 5993 } 5994 if (mFramestoDrop == 0) { 5995 mActiveTrack->releaseBuffer(&buffer); 5996 } else { 5997 if (mFramestoDrop > 0) { 5998 mFramestoDrop -= buffer.frameCount; 5999 if (mFramestoDrop <= 0) { 6000 clearSyncStartEvent(); 6001 } 6002 } else { 6003 mFramestoDrop += buffer.frameCount; 6004 if (mFramestoDrop >= 0 || mSyncStartEvent == 0 || 6005 mSyncStartEvent->isCancelled()) { 6006 ALOGW("Synced record %s, session %d, trigger session %d", 6007 (mFramestoDrop >= 0) ? "timed out" : "cancelled", 6008 mActiveTrack->sessionId(), 6009 (mSyncStartEvent != 0) ? mSyncStartEvent->triggerSession() : 0); 6010 clearSyncStartEvent(); 6011 } 6012 } 6013 } 6014 mActiveTrack->overflow(); 6015 } 6016 // client isn't retrieving buffers fast enough 6017 else { 6018 if (!mActiveTrack->setOverflow()) { 6019 nsecs_t now = systemTime(); 6020 if ((now - lastWarning) > kWarningThrottleNs) { 6021 ALOGW("RecordThread: buffer overflow"); 6022 lastWarning = now; 6023 } 6024 } 6025 // Release the processor for a while before asking for a new buffer. 6026 // This will give the application more chance to read from the buffer and 6027 // clear the overflow. 6028 usleep(kRecordThreadSleepUs); 6029 } 6030 } 6031 // enable changes in effect chain 6032 unlockEffectChains(effectChains); 6033 effectChains.clear(); 6034 } 6035 6036 if (!mStandby) { 6037 mInput->stream->common.standby(&mInput->stream->common); 6038 } 6039 mActiveTrack.clear(); 6040 6041 mStartStopCond.broadcast(); 6042 6043 releaseWakeLock(); 6044 6045 ALOGV("RecordThread %p exiting", this); 6046 return false; 6047} 6048 6049 6050sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 6051 const sp<AudioFlinger::Client>& client, 6052 uint32_t sampleRate, 6053 audio_format_t format, 6054 int channelMask, 6055 int frameCount, 6056 int sessionId, 6057 status_t *status) 6058{ 6059 sp<RecordTrack> track; 6060 status_t lStatus; 6061 6062 lStatus = initCheck(); 6063 if (lStatus != NO_ERROR) { 6064 ALOGE("Audio driver not initialized."); 6065 goto Exit; 6066 } 6067 6068 { // scope for mLock 6069 Mutex::Autolock _l(mLock); 6070 6071 track = new RecordTrack(this, client, sampleRate, 6072 format, channelMask, frameCount, sessionId); 6073 6074 if (track->getCblk() == 0) { 6075 lStatus = NO_MEMORY; 6076 goto Exit; 6077 } 6078 6079 mTrack = track.get(); 6080 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6081 bool suspend = audio_is_bluetooth_sco_device( 6082 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 6083 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 6084 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 6085 } 6086 lStatus = NO_ERROR; 6087 6088Exit: 6089 if (status) { 6090 *status = lStatus; 6091 } 6092 return track; 6093} 6094 6095status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 6096 AudioSystem::sync_event_t event, 6097 int triggerSession) 6098{ 6099 ALOGV("RecordThread::start event %d, triggerSession %d", event, triggerSession); 6100 sp<ThreadBase> strongMe = this; 6101 status_t status = NO_ERROR; 6102 6103 if (event == AudioSystem::SYNC_EVENT_NONE) { 6104 clearSyncStartEvent(); 6105 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 6106 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 6107 triggerSession, 6108 recordTrack->sessionId(), 6109 syncStartEventCallback, 6110 this); 6111 // Sync event can be cancelled by the trigger session if the track is not in a 6112 // compatible state in which case we start record immediately 6113 if (mSyncStartEvent->isCancelled()) { 6114 clearSyncStartEvent(); 6115 } else { 6116 // do not wait for the event for more than AudioSystem::kSyncRecordStartTimeOutMs 6117 mFramestoDrop = - ((AudioSystem::kSyncRecordStartTimeOutMs * mReqSampleRate) / 1000); 6118 } 6119 } 6120 6121 { 6122 AutoMutex lock(mLock); 6123 if (mActiveTrack != 0) { 6124 if (recordTrack != mActiveTrack.get()) { 6125 status = -EBUSY; 6126 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 6127 mActiveTrack->mState = TrackBase::ACTIVE; 6128 } 6129 return status; 6130 } 6131 6132 recordTrack->mState = TrackBase::IDLE; 6133 mActiveTrack = recordTrack; 6134 mLock.unlock(); 6135 status_t status = AudioSystem::startInput(mId); 6136 mLock.lock(); 6137 if (status != NO_ERROR) { 6138 mActiveTrack.clear(); 6139 clearSyncStartEvent(); 6140 return status; 6141 } 6142 mRsmpInIndex = mFrameCount; 6143 mBytesRead = 0; 6144 if (mResampler != NULL) { 6145 mResampler->reset(); 6146 } 6147 mActiveTrack->mState = TrackBase::RESUMING; 6148 // signal thread to start 6149 ALOGV("Signal record thread"); 6150 mWaitWorkCV.signal(); 6151 // do not wait for mStartStopCond if exiting 6152 if (exitPending()) { 6153 mActiveTrack.clear(); 6154 status = INVALID_OPERATION; 6155 goto startError; 6156 } 6157 mStartStopCond.wait(mLock); 6158 if (mActiveTrack == 0) { 6159 ALOGV("Record failed to start"); 6160 status = BAD_VALUE; 6161 goto startError; 6162 } 6163 ALOGV("Record started OK"); 6164 return status; 6165 } 6166startError: 6167 AudioSystem::stopInput(mId); 6168 clearSyncStartEvent(); 6169 return status; 6170} 6171 6172void AudioFlinger::RecordThread::clearSyncStartEvent() 6173{ 6174 if (mSyncStartEvent != 0) { 6175 mSyncStartEvent->cancel(); 6176 } 6177 mSyncStartEvent.clear(); 6178 mFramestoDrop = 0; 6179} 6180 6181void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 6182{ 6183 sp<SyncEvent> strongEvent = event.promote(); 6184 6185 if (strongEvent != 0) { 6186 RecordThread *me = (RecordThread *)strongEvent->cookie(); 6187 me->handleSyncStartEvent(strongEvent); 6188 } 6189} 6190 6191void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 6192{ 6193 if (event == mSyncStartEvent) { 6194 // TODO: use actual buffer filling status instead of 2 buffers when info is available 6195 // from audio HAL 6196 mFramestoDrop = mFrameCount * 2; 6197 } 6198} 6199 6200void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 6201 ALOGV("RecordThread::stop"); 6202 sp<ThreadBase> strongMe = this; 6203 { 6204 AutoMutex lock(mLock); 6205 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 6206 mActiveTrack->mState = TrackBase::PAUSING; 6207 // do not wait for mStartStopCond if exiting 6208 if (exitPending()) { 6209 return; 6210 } 6211 mStartStopCond.wait(mLock); 6212 // if we have been restarted, recordTrack == mActiveTrack.get() here 6213 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 6214 mLock.unlock(); 6215 AudioSystem::stopInput(mId); 6216 mLock.lock(); 6217 ALOGV("Record stopped OK"); 6218 } 6219 } 6220 } 6221} 6222 6223bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 6224{ 6225 return false; 6226} 6227 6228status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 6229{ 6230 if (!isValidSyncEvent(event)) { 6231 return BAD_VALUE; 6232 } 6233 6234 Mutex::Autolock _l(mLock); 6235 6236 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 6237 mTrack->setSyncEvent(event); 6238 return NO_ERROR; 6239 } 6240 return NAME_NOT_FOUND; 6241} 6242 6243status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 6244{ 6245 const size_t SIZE = 256; 6246 char buffer[SIZE]; 6247 String8 result; 6248 6249 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 6250 result.append(buffer); 6251 6252 if (mActiveTrack != 0) { 6253 result.append("Active Track:\n"); 6254 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 6255 mActiveTrack->dump(buffer, SIZE); 6256 result.append(buffer); 6257 6258 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 6259 result.append(buffer); 6260 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 6261 result.append(buffer); 6262 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 6263 result.append(buffer); 6264 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 6265 result.append(buffer); 6266 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 6267 result.append(buffer); 6268 6269 6270 } else { 6271 result.append("No record client\n"); 6272 } 6273 write(fd, result.string(), result.size()); 6274 6275 dumpBase(fd, args); 6276 dumpEffectChains(fd, args); 6277 6278 return NO_ERROR; 6279} 6280 6281// AudioBufferProvider interface 6282status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 6283{ 6284 size_t framesReq = buffer->frameCount; 6285 size_t framesReady = mFrameCount - mRsmpInIndex; 6286 int channelCount; 6287 6288 if (framesReady == 0) { 6289 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 6290 if (mBytesRead < 0) { 6291 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 6292 if (mActiveTrack->mState == TrackBase::ACTIVE) { 6293 // Force input into standby so that it tries to 6294 // recover at next read attempt 6295 mInput->stream->common.standby(&mInput->stream->common); 6296 usleep(kRecordThreadSleepUs); 6297 } 6298 buffer->raw = NULL; 6299 buffer->frameCount = 0; 6300 return NOT_ENOUGH_DATA; 6301 } 6302 mRsmpInIndex = 0; 6303 framesReady = mFrameCount; 6304 } 6305 6306 if (framesReq > framesReady) { 6307 framesReq = framesReady; 6308 } 6309 6310 if (mChannelCount == 1 && mReqChannelCount == 2) { 6311 channelCount = 1; 6312 } else { 6313 channelCount = 2; 6314 } 6315 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 6316 buffer->frameCount = framesReq; 6317 return NO_ERROR; 6318} 6319 6320// AudioBufferProvider interface 6321void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 6322{ 6323 mRsmpInIndex += buffer->frameCount; 6324 buffer->frameCount = 0; 6325} 6326 6327bool AudioFlinger::RecordThread::checkForNewParameters_l() 6328{ 6329 bool reconfig = false; 6330 6331 while (!mNewParameters.isEmpty()) { 6332 status_t status = NO_ERROR; 6333 String8 keyValuePair = mNewParameters[0]; 6334 AudioParameter param = AudioParameter(keyValuePair); 6335 int value; 6336 audio_format_t reqFormat = mFormat; 6337 int reqSamplingRate = mReqSampleRate; 6338 int reqChannelCount = mReqChannelCount; 6339 6340 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 6341 reqSamplingRate = value; 6342 reconfig = true; 6343 } 6344 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 6345 reqFormat = (audio_format_t) value; 6346 reconfig = true; 6347 } 6348 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 6349 reqChannelCount = popcount(value); 6350 reconfig = true; 6351 } 6352 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 6353 // do not accept frame count changes if tracks are open as the track buffer 6354 // size depends on frame count and correct behavior would not be guaranteed 6355 // if frame count is changed after track creation 6356 if (mActiveTrack != 0) { 6357 status = INVALID_OPERATION; 6358 } else { 6359 reconfig = true; 6360 } 6361 } 6362 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 6363 // forward device change to effects that have requested to be 6364 // aware of attached audio device. 6365 for (size_t i = 0; i < mEffectChains.size(); i++) { 6366 mEffectChains[i]->setDevice_l(value); 6367 } 6368 // store input device and output device but do not forward output device to audio HAL. 6369 // Note that status is ignored by the caller for output device 6370 // (see AudioFlinger::setParameters() 6371 if (value & AUDIO_DEVICE_OUT_ALL) { 6372 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 6373 status = BAD_VALUE; 6374 } else { 6375 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 6376 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 6377 if (mTrack != NULL) { 6378 bool suspend = audio_is_bluetooth_sco_device( 6379 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 6380 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 6381 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 6382 } 6383 } 6384 mDevice |= (uint32_t)value; 6385 } 6386 if (status == NO_ERROR) { 6387 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 6388 if (status == INVALID_OPERATION) { 6389 mInput->stream->common.standby(&mInput->stream->common); 6390 status = mInput->stream->common.set_parameters(&mInput->stream->common, 6391 keyValuePair.string()); 6392 } 6393 if (reconfig) { 6394 if (status == BAD_VALUE && 6395 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 6396 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 6397 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 6398 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 6399 (reqChannelCount <= FCC_2)) { 6400 status = NO_ERROR; 6401 } 6402 if (status == NO_ERROR) { 6403 readInputParameters(); 6404 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 6405 } 6406 } 6407 } 6408 6409 mNewParameters.removeAt(0); 6410 6411 mParamStatus = status; 6412 mParamCond.signal(); 6413 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 6414 // already timed out waiting for the status and will never signal the condition. 6415 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 6416 } 6417 return reconfig; 6418} 6419 6420String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 6421{ 6422 char *s; 6423 String8 out_s8 = String8(); 6424 6425 Mutex::Autolock _l(mLock); 6426 if (initCheck() != NO_ERROR) { 6427 return out_s8; 6428 } 6429 6430 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 6431 out_s8 = String8(s); 6432 free(s); 6433 return out_s8; 6434} 6435 6436void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 6437 AudioSystem::OutputDescriptor desc; 6438 void *param2 = NULL; 6439 6440 switch (event) { 6441 case AudioSystem::INPUT_OPENED: 6442 case AudioSystem::INPUT_CONFIG_CHANGED: 6443 desc.channels = mChannelMask; 6444 desc.samplingRate = mSampleRate; 6445 desc.format = mFormat; 6446 desc.frameCount = mFrameCount; 6447 desc.latency = 0; 6448 param2 = &desc; 6449 break; 6450 6451 case AudioSystem::INPUT_CLOSED: 6452 default: 6453 break; 6454 } 6455 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 6456} 6457 6458void AudioFlinger::RecordThread::readInputParameters() 6459{ 6460 delete mRsmpInBuffer; 6461 // mRsmpInBuffer is always assigned a new[] below 6462 delete mRsmpOutBuffer; 6463 mRsmpOutBuffer = NULL; 6464 delete mResampler; 6465 mResampler = NULL; 6466 6467 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 6468 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 6469 mChannelCount = (uint16_t)popcount(mChannelMask); 6470 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 6471 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 6472 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 6473 mFrameCount = mInputBytes / mFrameSize; 6474 mNormalFrameCount = mFrameCount; // not used by record, but used by input effects 6475 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 6476 6477 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 6478 { 6479 int channelCount; 6480 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 6481 // stereo to mono post process as the resampler always outputs stereo. 6482 if (mChannelCount == 1 && mReqChannelCount == 2) { 6483 channelCount = 1; 6484 } else { 6485 channelCount = 2; 6486 } 6487 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 6488 mResampler->setSampleRate(mSampleRate); 6489 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 6490 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 6491 6492 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 6493 if (mChannelCount == 1 && mReqChannelCount == 1) { 6494 mFrameCount >>= 1; 6495 } 6496 6497 } 6498 mRsmpInIndex = mFrameCount; 6499} 6500 6501unsigned int AudioFlinger::RecordThread::getInputFramesLost() 6502{ 6503 Mutex::Autolock _l(mLock); 6504 if (initCheck() != NO_ERROR) { 6505 return 0; 6506 } 6507 6508 return mInput->stream->get_input_frames_lost(mInput->stream); 6509} 6510 6511uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 6512{ 6513 Mutex::Autolock _l(mLock); 6514 uint32_t result = 0; 6515 if (getEffectChain_l(sessionId) != 0) { 6516 result = EFFECT_SESSION; 6517 } 6518 6519 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 6520 result |= TRACK_SESSION; 6521 } 6522 6523 return result; 6524} 6525 6526AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 6527{ 6528 Mutex::Autolock _l(mLock); 6529 return mTrack; 6530} 6531 6532AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 6533{ 6534 Mutex::Autolock _l(mLock); 6535 return mInput; 6536} 6537 6538AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 6539{ 6540 Mutex::Autolock _l(mLock); 6541 AudioStreamIn *input = mInput; 6542 mInput = NULL; 6543 return input; 6544} 6545 6546// this method must always be called either with ThreadBase mLock held or inside the thread loop 6547audio_stream_t* AudioFlinger::RecordThread::stream() const 6548{ 6549 if (mInput == NULL) { 6550 return NULL; 6551 } 6552 return &mInput->stream->common; 6553} 6554 6555 6556// ---------------------------------------------------------------------------- 6557 6558audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 6559{ 6560 if (!settingsAllowed()) { 6561 return 0; 6562 } 6563 Mutex::Autolock _l(mLock); 6564 return loadHwModule_l(name); 6565} 6566 6567// loadHwModule_l() must be called with AudioFlinger::mLock held 6568audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 6569{ 6570 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6571 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 6572 ALOGW("loadHwModule() module %s already loaded", name); 6573 return mAudioHwDevs.keyAt(i); 6574 } 6575 } 6576 6577 audio_hw_device_t *dev; 6578 6579 int rc = load_audio_interface(name, &dev); 6580 if (rc) { 6581 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 6582 return 0; 6583 } 6584 6585 mHardwareStatus = AUDIO_HW_INIT; 6586 rc = dev->init_check(dev); 6587 mHardwareStatus = AUDIO_HW_IDLE; 6588 if (rc) { 6589 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 6590 return 0; 6591 } 6592 6593 if ((mMasterVolumeSupportLvl != MVS_NONE) && 6594 (NULL != dev->set_master_volume)) { 6595 AutoMutex lock(mHardwareLock); 6596 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6597 dev->set_master_volume(dev, mMasterVolume); 6598 mHardwareStatus = AUDIO_HW_IDLE; 6599 } 6600 6601 audio_module_handle_t handle = nextUniqueId(); 6602 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 6603 6604 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 6605 name, dev->common.module->name, dev->common.module->id, handle); 6606 6607 return handle; 6608 6609} 6610 6611audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 6612 audio_devices_t *pDevices, 6613 uint32_t *pSamplingRate, 6614 audio_format_t *pFormat, 6615 audio_channel_mask_t *pChannelMask, 6616 uint32_t *pLatencyMs, 6617 audio_output_flags_t flags) 6618{ 6619 status_t status; 6620 PlaybackThread *thread = NULL; 6621 struct audio_config config = { 6622 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6623 channel_mask: pChannelMask ? *pChannelMask : 0, 6624 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6625 }; 6626 audio_stream_out_t *outStream = NULL; 6627 audio_hw_device_t *outHwDev; 6628 6629 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 6630 module, 6631 (pDevices != NULL) ? (int)*pDevices : 0, 6632 config.sample_rate, 6633 config.format, 6634 config.channel_mask, 6635 flags); 6636 6637 if (pDevices == NULL || *pDevices == 0) { 6638 return 0; 6639 } 6640 6641 Mutex::Autolock _l(mLock); 6642 6643 outHwDev = findSuitableHwDev_l(module, *pDevices); 6644 if (outHwDev == NULL) 6645 return 0; 6646 6647 audio_io_handle_t id = nextUniqueId(); 6648 6649 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 6650 6651 status = outHwDev->open_output_stream(outHwDev, 6652 id, 6653 *pDevices, 6654 (audio_output_flags_t)flags, 6655 &config, 6656 &outStream); 6657 6658 mHardwareStatus = AUDIO_HW_IDLE; 6659 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 6660 outStream, 6661 config.sample_rate, 6662 config.format, 6663 config.channel_mask, 6664 status); 6665 6666 if (status == NO_ERROR && outStream != NULL) { 6667 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 6668 6669 if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) || 6670 (config.format != AUDIO_FORMAT_PCM_16_BIT) || 6671 (config.channel_mask != AUDIO_CHANNEL_OUT_STEREO)) { 6672 thread = new DirectOutputThread(this, output, id, *pDevices); 6673 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 6674 } else { 6675 thread = new MixerThread(this, output, id, *pDevices); 6676 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 6677 } 6678 mPlaybackThreads.add(id, thread); 6679 6680 if (pSamplingRate != NULL) *pSamplingRate = config.sample_rate; 6681 if (pFormat != NULL) *pFormat = config.format; 6682 if (pChannelMask != NULL) *pChannelMask = config.channel_mask; 6683 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 6684 6685 // notify client processes of the new output creation 6686 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6687 6688 // the first primary output opened designates the primary hw device 6689 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 6690 ALOGI("Using module %d has the primary audio interface", module); 6691 mPrimaryHardwareDev = outHwDev; 6692 6693 AutoMutex lock(mHardwareLock); 6694 mHardwareStatus = AUDIO_HW_SET_MODE; 6695 outHwDev->set_mode(outHwDev, mMode); 6696 6697 // Determine the level of master volume support the primary audio HAL has, 6698 // and set the initial master volume at the same time. 6699 float initialVolume = 1.0; 6700 mMasterVolumeSupportLvl = MVS_NONE; 6701 6702 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 6703 if ((NULL != outHwDev->get_master_volume) && 6704 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 6705 mMasterVolumeSupportLvl = MVS_FULL; 6706 } else { 6707 mMasterVolumeSupportLvl = MVS_SETONLY; 6708 initialVolume = 1.0; 6709 } 6710 6711 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 6712 if ((NULL == outHwDev->set_master_volume) || 6713 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 6714 mMasterVolumeSupportLvl = MVS_NONE; 6715 } 6716 // now that we have a primary device, initialize master volume on other devices 6717 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 6718 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 6719 6720 if ((dev != mPrimaryHardwareDev) && 6721 (NULL != dev->set_master_volume)) { 6722 dev->set_master_volume(dev, initialVolume); 6723 } 6724 } 6725 mHardwareStatus = AUDIO_HW_IDLE; 6726 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 6727 ? initialVolume 6728 : 1.0; 6729 mMasterVolume = initialVolume; 6730 } 6731 return id; 6732 } 6733 6734 return 0; 6735} 6736 6737audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 6738 audio_io_handle_t output2) 6739{ 6740 Mutex::Autolock _l(mLock); 6741 MixerThread *thread1 = checkMixerThread_l(output1); 6742 MixerThread *thread2 = checkMixerThread_l(output2); 6743 6744 if (thread1 == NULL || thread2 == NULL) { 6745 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 6746 return 0; 6747 } 6748 6749 audio_io_handle_t id = nextUniqueId(); 6750 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 6751 thread->addOutputTrack(thread2); 6752 mPlaybackThreads.add(id, thread); 6753 // notify client processes of the new output creation 6754 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 6755 return id; 6756} 6757 6758status_t AudioFlinger::closeOutput(audio_io_handle_t output) 6759{ 6760 // keep strong reference on the playback thread so that 6761 // it is not destroyed while exit() is executed 6762 sp<PlaybackThread> thread; 6763 { 6764 Mutex::Autolock _l(mLock); 6765 thread = checkPlaybackThread_l(output); 6766 if (thread == NULL) { 6767 return BAD_VALUE; 6768 } 6769 6770 ALOGV("closeOutput() %d", output); 6771 6772 if (thread->type() == ThreadBase::MIXER) { 6773 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6774 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 6775 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 6776 dupThread->removeOutputTrack((MixerThread *)thread.get()); 6777 } 6778 } 6779 } 6780 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 6781 mPlaybackThreads.removeItem(output); 6782 } 6783 thread->exit(); 6784 // The thread entity (active unit of execution) is no longer running here, 6785 // but the ThreadBase container still exists. 6786 6787 if (thread->type() != ThreadBase::DUPLICATING) { 6788 AudioStreamOut *out = thread->clearOutput(); 6789 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 6790 // from now on thread->mOutput is NULL 6791 out->hwDev->close_output_stream(out->hwDev, out->stream); 6792 delete out; 6793 } 6794 return NO_ERROR; 6795} 6796 6797status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 6798{ 6799 Mutex::Autolock _l(mLock); 6800 PlaybackThread *thread = checkPlaybackThread_l(output); 6801 6802 if (thread == NULL) { 6803 return BAD_VALUE; 6804 } 6805 6806 ALOGV("suspendOutput() %d", output); 6807 thread->suspend(); 6808 6809 return NO_ERROR; 6810} 6811 6812status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 6813{ 6814 Mutex::Autolock _l(mLock); 6815 PlaybackThread *thread = checkPlaybackThread_l(output); 6816 6817 if (thread == NULL) { 6818 return BAD_VALUE; 6819 } 6820 6821 ALOGV("restoreOutput() %d", output); 6822 6823 thread->restore(); 6824 6825 return NO_ERROR; 6826} 6827 6828audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 6829 audio_devices_t *pDevices, 6830 uint32_t *pSamplingRate, 6831 audio_format_t *pFormat, 6832 uint32_t *pChannelMask) 6833{ 6834 status_t status; 6835 RecordThread *thread = NULL; 6836 struct audio_config config = { 6837 sample_rate: pSamplingRate ? *pSamplingRate : 0, 6838 channel_mask: pChannelMask ? *pChannelMask : 0, 6839 format: pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT, 6840 }; 6841 uint32_t reqSamplingRate = config.sample_rate; 6842 audio_format_t reqFormat = config.format; 6843 audio_channel_mask_t reqChannels = config.channel_mask; 6844 audio_stream_in_t *inStream = NULL; 6845 audio_hw_device_t *inHwDev; 6846 6847 if (pDevices == NULL || *pDevices == 0) { 6848 return 0; 6849 } 6850 6851 Mutex::Autolock _l(mLock); 6852 6853 inHwDev = findSuitableHwDev_l(module, *pDevices); 6854 if (inHwDev == NULL) 6855 return 0; 6856 6857 audio_io_handle_t id = nextUniqueId(); 6858 6859 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, 6860 &inStream); 6861 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 6862 inStream, 6863 config.sample_rate, 6864 config.format, 6865 config.channel_mask, 6866 status); 6867 6868 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 6869 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 6870 // or stereo to mono conversions on 16 bit PCM inputs. 6871 if (status == BAD_VALUE && 6872 reqFormat == config.format && config.format == AUDIO_FORMAT_PCM_16_BIT && 6873 (config.sample_rate <= 2 * reqSamplingRate) && 6874 (popcount(config.channel_mask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 6875 ALOGV("openInput() reopening with proposed sampling rate and channels"); 6876 inStream = NULL; 6877 status = inHwDev->open_input_stream(inHwDev, id, *pDevices, &config, &inStream); 6878 } 6879 6880 if (status == NO_ERROR && inStream != NULL) { 6881 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 6882 6883 // Start record thread 6884 // RecorThread require both input and output device indication to forward to audio 6885 // pre processing modules 6886 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 6887 thread = new RecordThread(this, 6888 input, 6889 reqSamplingRate, 6890 reqChannels, 6891 id, 6892 device); 6893 mRecordThreads.add(id, thread); 6894 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 6895 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 6896 if (pFormat != NULL) *pFormat = config.format; 6897 if (pChannelMask != NULL) *pChannelMask = reqChannels; 6898 6899 input->stream->common.standby(&input->stream->common); 6900 6901 // notify client processes of the new input creation 6902 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 6903 return id; 6904 } 6905 6906 return 0; 6907} 6908 6909status_t AudioFlinger::closeInput(audio_io_handle_t input) 6910{ 6911 // keep strong reference on the record thread so that 6912 // it is not destroyed while exit() is executed 6913 sp<RecordThread> thread; 6914 { 6915 Mutex::Autolock _l(mLock); 6916 thread = checkRecordThread_l(input); 6917 if (thread == NULL) { 6918 return BAD_VALUE; 6919 } 6920 6921 ALOGV("closeInput() %d", input); 6922 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 6923 mRecordThreads.removeItem(input); 6924 } 6925 thread->exit(); 6926 // The thread entity (active unit of execution) is no longer running here, 6927 // but the ThreadBase container still exists. 6928 6929 AudioStreamIn *in = thread->clearInput(); 6930 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 6931 // from now on thread->mInput is NULL 6932 in->hwDev->close_input_stream(in->hwDev, in->stream); 6933 delete in; 6934 6935 return NO_ERROR; 6936} 6937 6938status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 6939{ 6940 Mutex::Autolock _l(mLock); 6941 MixerThread *dstThread = checkMixerThread_l(output); 6942 if (dstThread == NULL) { 6943 ALOGW("setStreamOutput() bad output id %d", output); 6944 return BAD_VALUE; 6945 } 6946 6947 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 6948 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 6949 6950 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6951 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6952 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6953 MixerThread *srcThread = (MixerThread *)thread; 6954 srcThread->invalidateTracks(stream); 6955 } 6956 } 6957 6958 return NO_ERROR; 6959} 6960 6961 6962int AudioFlinger::newAudioSessionId() 6963{ 6964 return nextUniqueId(); 6965} 6966 6967void AudioFlinger::acquireAudioSessionId(int audioSession) 6968{ 6969 Mutex::Autolock _l(mLock); 6970 pid_t caller = IPCThreadState::self()->getCallingPid(); 6971 ALOGV("acquiring %d from %d", audioSession, caller); 6972 size_t num = mAudioSessionRefs.size(); 6973 for (size_t i = 0; i< num; i++) { 6974 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6975 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6976 ref->mCnt++; 6977 ALOGV(" incremented refcount to %d", ref->mCnt); 6978 return; 6979 } 6980 } 6981 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6982 ALOGV(" added new entry for %d", audioSession); 6983} 6984 6985void AudioFlinger::releaseAudioSessionId(int audioSession) 6986{ 6987 Mutex::Autolock _l(mLock); 6988 pid_t caller = IPCThreadState::self()->getCallingPid(); 6989 ALOGV("releasing %d from %d", audioSession, caller); 6990 size_t num = mAudioSessionRefs.size(); 6991 for (size_t i = 0; i< num; i++) { 6992 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6993 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6994 ref->mCnt--; 6995 ALOGV(" decremented refcount to %d", ref->mCnt); 6996 if (ref->mCnt == 0) { 6997 mAudioSessionRefs.removeAt(i); 6998 delete ref; 6999 purgeStaleEffects_l(); 7000 } 7001 return; 7002 } 7003 } 7004 ALOGW("session id %d not found for pid %d", audioSession, caller); 7005} 7006 7007void AudioFlinger::purgeStaleEffects_l() { 7008 7009 ALOGV("purging stale effects"); 7010 7011 Vector< sp<EffectChain> > chains; 7012 7013 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7014 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 7015 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7016 sp<EffectChain> ec = t->mEffectChains[j]; 7017 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 7018 chains.push(ec); 7019 } 7020 } 7021 } 7022 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7023 sp<RecordThread> t = mRecordThreads.valueAt(i); 7024 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 7025 sp<EffectChain> ec = t->mEffectChains[j]; 7026 chains.push(ec); 7027 } 7028 } 7029 7030 for (size_t i = 0; i < chains.size(); i++) { 7031 sp<EffectChain> ec = chains[i]; 7032 int sessionid = ec->sessionId(); 7033 sp<ThreadBase> t = ec->mThread.promote(); 7034 if (t == 0) { 7035 continue; 7036 } 7037 size_t numsessionrefs = mAudioSessionRefs.size(); 7038 bool found = false; 7039 for (size_t k = 0; k < numsessionrefs; k++) { 7040 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 7041 if (ref->mSessionid == sessionid) { 7042 ALOGV(" session %d still exists for %d with %d refs", 7043 sessionid, ref->mPid, ref->mCnt); 7044 found = true; 7045 break; 7046 } 7047 } 7048 if (!found) { 7049 // remove all effects from the chain 7050 while (ec->mEffects.size()) { 7051 sp<EffectModule> effect = ec->mEffects[0]; 7052 effect->unPin(); 7053 Mutex::Autolock _l (t->mLock); 7054 t->removeEffect_l(effect); 7055 for (size_t j = 0; j < effect->mHandles.size(); j++) { 7056 sp<EffectHandle> handle = effect->mHandles[j].promote(); 7057 if (handle != 0) { 7058 handle->mEffect.clear(); 7059 if (handle->mHasControl && handle->mEnabled) { 7060 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 7061 } 7062 } 7063 } 7064 AudioSystem::unregisterEffect(effect->id()); 7065 } 7066 } 7067 } 7068 return; 7069} 7070 7071// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 7072AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 7073{ 7074 return mPlaybackThreads.valueFor(output).get(); 7075} 7076 7077// checkMixerThread_l() must be called with AudioFlinger::mLock held 7078AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 7079{ 7080 PlaybackThread *thread = checkPlaybackThread_l(output); 7081 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 7082} 7083 7084// checkRecordThread_l() must be called with AudioFlinger::mLock held 7085AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 7086{ 7087 return mRecordThreads.valueFor(input).get(); 7088} 7089 7090uint32_t AudioFlinger::nextUniqueId() 7091{ 7092 return android_atomic_inc(&mNextUniqueId); 7093} 7094 7095AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 7096{ 7097 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7098 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 7099 AudioStreamOut *output = thread->getOutput(); 7100 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 7101 return thread; 7102 } 7103 } 7104 return NULL; 7105} 7106 7107uint32_t AudioFlinger::primaryOutputDevice_l() const 7108{ 7109 PlaybackThread *thread = primaryPlaybackThread_l(); 7110 7111 if (thread == NULL) { 7112 return 0; 7113 } 7114 7115 return thread->device(); 7116} 7117 7118sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 7119 int triggerSession, 7120 int listenerSession, 7121 sync_event_callback_t callBack, 7122 void *cookie) 7123{ 7124 Mutex::Autolock _l(mLock); 7125 7126 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 7127 status_t playStatus = NAME_NOT_FOUND; 7128 status_t recStatus = NAME_NOT_FOUND; 7129 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7130 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 7131 if (playStatus == NO_ERROR) { 7132 return event; 7133 } 7134 } 7135 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7136 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 7137 if (recStatus == NO_ERROR) { 7138 return event; 7139 } 7140 } 7141 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 7142 mPendingSyncEvents.add(event); 7143 } else { 7144 ALOGV("createSyncEvent() invalid event %d", event->type()); 7145 event.clear(); 7146 } 7147 return event; 7148} 7149 7150// ---------------------------------------------------------------------------- 7151// Effect management 7152// ---------------------------------------------------------------------------- 7153 7154 7155status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 7156{ 7157 Mutex::Autolock _l(mLock); 7158 return EffectQueryNumberEffects(numEffects); 7159} 7160 7161status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 7162{ 7163 Mutex::Autolock _l(mLock); 7164 return EffectQueryEffect(index, descriptor); 7165} 7166 7167status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 7168 effect_descriptor_t *descriptor) const 7169{ 7170 Mutex::Autolock _l(mLock); 7171 return EffectGetDescriptor(pUuid, descriptor); 7172} 7173 7174 7175sp<IEffect> AudioFlinger::createEffect(pid_t pid, 7176 effect_descriptor_t *pDesc, 7177 const sp<IEffectClient>& effectClient, 7178 int32_t priority, 7179 audio_io_handle_t io, 7180 int sessionId, 7181 status_t *status, 7182 int *id, 7183 int *enabled) 7184{ 7185 status_t lStatus = NO_ERROR; 7186 sp<EffectHandle> handle; 7187 effect_descriptor_t desc; 7188 7189 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 7190 pid, effectClient.get(), priority, sessionId, io); 7191 7192 if (pDesc == NULL) { 7193 lStatus = BAD_VALUE; 7194 goto Exit; 7195 } 7196 7197 // check audio settings permission for global effects 7198 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 7199 lStatus = PERMISSION_DENIED; 7200 goto Exit; 7201 } 7202 7203 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 7204 // that can only be created by audio policy manager (running in same process) 7205 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 7206 lStatus = PERMISSION_DENIED; 7207 goto Exit; 7208 } 7209 7210 if (io == 0) { 7211 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 7212 // output must be specified by AudioPolicyManager when using session 7213 // AUDIO_SESSION_OUTPUT_STAGE 7214 lStatus = BAD_VALUE; 7215 goto Exit; 7216 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 7217 // if the output returned by getOutputForEffect() is removed before we lock the 7218 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 7219 // and we will exit safely 7220 io = AudioSystem::getOutputForEffect(&desc); 7221 } 7222 } 7223 7224 { 7225 Mutex::Autolock _l(mLock); 7226 7227 7228 if (!EffectIsNullUuid(&pDesc->uuid)) { 7229 // if uuid is specified, request effect descriptor 7230 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 7231 if (lStatus < 0) { 7232 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 7233 goto Exit; 7234 } 7235 } else { 7236 // if uuid is not specified, look for an available implementation 7237 // of the required type in effect factory 7238 if (EffectIsNullUuid(&pDesc->type)) { 7239 ALOGW("createEffect() no effect type"); 7240 lStatus = BAD_VALUE; 7241 goto Exit; 7242 } 7243 uint32_t numEffects = 0; 7244 effect_descriptor_t d; 7245 d.flags = 0; // prevent compiler warning 7246 bool found = false; 7247 7248 lStatus = EffectQueryNumberEffects(&numEffects); 7249 if (lStatus < 0) { 7250 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 7251 goto Exit; 7252 } 7253 for (uint32_t i = 0; i < numEffects; i++) { 7254 lStatus = EffectQueryEffect(i, &desc); 7255 if (lStatus < 0) { 7256 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 7257 continue; 7258 } 7259 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 7260 // If matching type found save effect descriptor. If the session is 7261 // 0 and the effect is not auxiliary, continue enumeration in case 7262 // an auxiliary version of this effect type is available 7263 found = true; 7264 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 7265 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 7266 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7267 break; 7268 } 7269 } 7270 } 7271 if (!found) { 7272 lStatus = BAD_VALUE; 7273 ALOGW("createEffect() effect not found"); 7274 goto Exit; 7275 } 7276 // For same effect type, chose auxiliary version over insert version if 7277 // connect to output mix (Compliance to OpenSL ES) 7278 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 7279 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 7280 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 7281 } 7282 } 7283 7284 // Do not allow auxiliary effects on a session different from 0 (output mix) 7285 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 7286 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7287 lStatus = INVALID_OPERATION; 7288 goto Exit; 7289 } 7290 7291 // check recording permission for visualizer 7292 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 7293 !recordingAllowed()) { 7294 lStatus = PERMISSION_DENIED; 7295 goto Exit; 7296 } 7297 7298 // return effect descriptor 7299 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 7300 7301 // If output is not specified try to find a matching audio session ID in one of the 7302 // output threads. 7303 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 7304 // because of code checking output when entering the function. 7305 // Note: io is never 0 when creating an effect on an input 7306 if (io == 0) { 7307 // look for the thread where the specified audio session is present 7308 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 7309 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7310 io = mPlaybackThreads.keyAt(i); 7311 break; 7312 } 7313 } 7314 if (io == 0) { 7315 for (size_t i = 0; i < mRecordThreads.size(); i++) { 7316 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 7317 io = mRecordThreads.keyAt(i); 7318 break; 7319 } 7320 } 7321 } 7322 // If no output thread contains the requested session ID, default to 7323 // first output. The effect chain will be moved to the correct output 7324 // thread when a track with the same session ID is created 7325 if (io == 0 && mPlaybackThreads.size()) { 7326 io = mPlaybackThreads.keyAt(0); 7327 } 7328 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 7329 } 7330 ThreadBase *thread = checkRecordThread_l(io); 7331 if (thread == NULL) { 7332 thread = checkPlaybackThread_l(io); 7333 if (thread == NULL) { 7334 ALOGE("createEffect() unknown output thread"); 7335 lStatus = BAD_VALUE; 7336 goto Exit; 7337 } 7338 } 7339 7340 sp<Client> client = registerPid_l(pid); 7341 7342 // create effect on selected output thread 7343 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 7344 &desc, enabled, &lStatus); 7345 if (handle != 0 && id != NULL) { 7346 *id = handle->id(); 7347 } 7348 } 7349 7350Exit: 7351 if (status != NULL) { 7352 *status = lStatus; 7353 } 7354 return handle; 7355} 7356 7357status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 7358 audio_io_handle_t dstOutput) 7359{ 7360 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 7361 sessionId, srcOutput, dstOutput); 7362 Mutex::Autolock _l(mLock); 7363 if (srcOutput == dstOutput) { 7364 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 7365 return NO_ERROR; 7366 } 7367 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 7368 if (srcThread == NULL) { 7369 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 7370 return BAD_VALUE; 7371 } 7372 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 7373 if (dstThread == NULL) { 7374 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 7375 return BAD_VALUE; 7376 } 7377 7378 Mutex::Autolock _dl(dstThread->mLock); 7379 Mutex::Autolock _sl(srcThread->mLock); 7380 moveEffectChain_l(sessionId, srcThread, dstThread, false); 7381 7382 return NO_ERROR; 7383} 7384 7385// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 7386status_t AudioFlinger::moveEffectChain_l(int sessionId, 7387 AudioFlinger::PlaybackThread *srcThread, 7388 AudioFlinger::PlaybackThread *dstThread, 7389 bool reRegister) 7390{ 7391 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 7392 sessionId, srcThread, dstThread); 7393 7394 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 7395 if (chain == 0) { 7396 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 7397 sessionId, srcThread); 7398 return INVALID_OPERATION; 7399 } 7400 7401 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 7402 // so that a new chain is created with correct parameters when first effect is added. This is 7403 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 7404 // removed. 7405 srcThread->removeEffectChain_l(chain); 7406 7407 // transfer all effects one by one so that new effect chain is created on new thread with 7408 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 7409 audio_io_handle_t dstOutput = dstThread->id(); 7410 sp<EffectChain> dstChain; 7411 uint32_t strategy = 0; // prevent compiler warning 7412 sp<EffectModule> effect = chain->getEffectFromId_l(0); 7413 while (effect != 0) { 7414 srcThread->removeEffect_l(effect); 7415 dstThread->addEffect_l(effect); 7416 // removeEffect_l() has stopped the effect if it was active so it must be restarted 7417 if (effect->state() == EffectModule::ACTIVE || 7418 effect->state() == EffectModule::STOPPING) { 7419 effect->start(); 7420 } 7421 // if the move request is not received from audio policy manager, the effect must be 7422 // re-registered with the new strategy and output 7423 if (dstChain == 0) { 7424 dstChain = effect->chain().promote(); 7425 if (dstChain == 0) { 7426 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 7427 srcThread->addEffect_l(effect); 7428 return NO_INIT; 7429 } 7430 strategy = dstChain->strategy(); 7431 } 7432 if (reRegister) { 7433 AudioSystem::unregisterEffect(effect->id()); 7434 AudioSystem::registerEffect(&effect->desc(), 7435 dstOutput, 7436 strategy, 7437 sessionId, 7438 effect->id()); 7439 } 7440 effect = chain->getEffectFromId_l(0); 7441 } 7442 7443 return NO_ERROR; 7444} 7445 7446 7447// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 7448sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 7449 const sp<AudioFlinger::Client>& client, 7450 const sp<IEffectClient>& effectClient, 7451 int32_t priority, 7452 int sessionId, 7453 effect_descriptor_t *desc, 7454 int *enabled, 7455 status_t *status 7456 ) 7457{ 7458 sp<EffectModule> effect; 7459 sp<EffectHandle> handle; 7460 status_t lStatus; 7461 sp<EffectChain> chain; 7462 bool chainCreated = false; 7463 bool effectCreated = false; 7464 bool effectRegistered = false; 7465 7466 lStatus = initCheck(); 7467 if (lStatus != NO_ERROR) { 7468 ALOGW("createEffect_l() Audio driver not initialized."); 7469 goto Exit; 7470 } 7471 7472 // Do not allow effects with session ID 0 on direct output or duplicating threads 7473 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 7474 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 7475 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 7476 desc->name, sessionId); 7477 lStatus = BAD_VALUE; 7478 goto Exit; 7479 } 7480 // Only Pre processor effects are allowed on input threads and only on input threads 7481 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 7482 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 7483 desc->name, desc->flags, mType); 7484 lStatus = BAD_VALUE; 7485 goto Exit; 7486 } 7487 7488 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 7489 7490 { // scope for mLock 7491 Mutex::Autolock _l(mLock); 7492 7493 // check for existing effect chain with the requested audio session 7494 chain = getEffectChain_l(sessionId); 7495 if (chain == 0) { 7496 // create a new chain for this session 7497 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 7498 chain = new EffectChain(this, sessionId); 7499 addEffectChain_l(chain); 7500 chain->setStrategy(getStrategyForSession_l(sessionId)); 7501 chainCreated = true; 7502 } else { 7503 effect = chain->getEffectFromDesc_l(desc); 7504 } 7505 7506 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 7507 7508 if (effect == 0) { 7509 int id = mAudioFlinger->nextUniqueId(); 7510 // Check CPU and memory usage 7511 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 7512 if (lStatus != NO_ERROR) { 7513 goto Exit; 7514 } 7515 effectRegistered = true; 7516 // create a new effect module if none present in the chain 7517 effect = new EffectModule(this, chain, desc, id, sessionId); 7518 lStatus = effect->status(); 7519 if (lStatus != NO_ERROR) { 7520 goto Exit; 7521 } 7522 lStatus = chain->addEffect_l(effect); 7523 if (lStatus != NO_ERROR) { 7524 goto Exit; 7525 } 7526 effectCreated = true; 7527 7528 effect->setDevice(mDevice); 7529 effect->setMode(mAudioFlinger->getMode()); 7530 } 7531 // create effect handle and connect it to effect module 7532 handle = new EffectHandle(effect, client, effectClient, priority); 7533 lStatus = effect->addHandle(handle); 7534 if (enabled != NULL) { 7535 *enabled = (int)effect->isEnabled(); 7536 } 7537 } 7538 7539Exit: 7540 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 7541 Mutex::Autolock _l(mLock); 7542 if (effectCreated) { 7543 chain->removeEffect_l(effect); 7544 } 7545 if (effectRegistered) { 7546 AudioSystem::unregisterEffect(effect->id()); 7547 } 7548 if (chainCreated) { 7549 removeEffectChain_l(chain); 7550 } 7551 handle.clear(); 7552 } 7553 7554 if (status != NULL) { 7555 *status = lStatus; 7556 } 7557 return handle; 7558} 7559 7560sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 7561{ 7562 sp<EffectChain> chain = getEffectChain_l(sessionId); 7563 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 7564} 7565 7566// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 7567// PlaybackThread::mLock held 7568status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 7569{ 7570 // check for existing effect chain with the requested audio session 7571 int sessionId = effect->sessionId(); 7572 sp<EffectChain> chain = getEffectChain_l(sessionId); 7573 bool chainCreated = false; 7574 7575 if (chain == 0) { 7576 // create a new chain for this session 7577 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 7578 chain = new EffectChain(this, sessionId); 7579 addEffectChain_l(chain); 7580 chain->setStrategy(getStrategyForSession_l(sessionId)); 7581 chainCreated = true; 7582 } 7583 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 7584 7585 if (chain->getEffectFromId_l(effect->id()) != 0) { 7586 ALOGW("addEffect_l() %p effect %s already present in chain %p", 7587 this, effect->desc().name, chain.get()); 7588 return BAD_VALUE; 7589 } 7590 7591 status_t status = chain->addEffect_l(effect); 7592 if (status != NO_ERROR) { 7593 if (chainCreated) { 7594 removeEffectChain_l(chain); 7595 } 7596 return status; 7597 } 7598 7599 effect->setDevice(mDevice); 7600 effect->setMode(mAudioFlinger->getMode()); 7601 return NO_ERROR; 7602} 7603 7604void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 7605 7606 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 7607 effect_descriptor_t desc = effect->desc(); 7608 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7609 detachAuxEffect_l(effect->id()); 7610 } 7611 7612 sp<EffectChain> chain = effect->chain().promote(); 7613 if (chain != 0) { 7614 // remove effect chain if removing last effect 7615 if (chain->removeEffect_l(effect) == 0) { 7616 removeEffectChain_l(chain); 7617 } 7618 } else { 7619 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 7620 } 7621} 7622 7623void AudioFlinger::ThreadBase::lockEffectChains_l( 7624 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7625{ 7626 effectChains = mEffectChains; 7627 for (size_t i = 0; i < mEffectChains.size(); i++) { 7628 mEffectChains[i]->lock(); 7629 } 7630} 7631 7632void AudioFlinger::ThreadBase::unlockEffectChains( 7633 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 7634{ 7635 for (size_t i = 0; i < effectChains.size(); i++) { 7636 effectChains[i]->unlock(); 7637 } 7638} 7639 7640sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 7641{ 7642 Mutex::Autolock _l(mLock); 7643 return getEffectChain_l(sessionId); 7644} 7645 7646sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 7647{ 7648 size_t size = mEffectChains.size(); 7649 for (size_t i = 0; i < size; i++) { 7650 if (mEffectChains[i]->sessionId() == sessionId) { 7651 return mEffectChains[i]; 7652 } 7653 } 7654 return 0; 7655} 7656 7657void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 7658{ 7659 Mutex::Autolock _l(mLock); 7660 size_t size = mEffectChains.size(); 7661 for (size_t i = 0; i < size; i++) { 7662 mEffectChains[i]->setMode_l(mode); 7663 } 7664} 7665 7666void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 7667 const wp<EffectHandle>& handle, 7668 bool unpinIfLast) { 7669 7670 Mutex::Autolock _l(mLock); 7671 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 7672 // delete the effect module if removing last handle on it 7673 if (effect->removeHandle(handle) == 0) { 7674 if (!effect->isPinned() || unpinIfLast) { 7675 removeEffect_l(effect); 7676 AudioSystem::unregisterEffect(effect->id()); 7677 } 7678 } 7679} 7680 7681status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 7682{ 7683 int session = chain->sessionId(); 7684 int16_t *buffer = mMixBuffer; 7685 bool ownsBuffer = false; 7686 7687 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 7688 if (session > 0) { 7689 // Only one effect chain can be present in direct output thread and it uses 7690 // the mix buffer as input 7691 if (mType != DIRECT) { 7692 size_t numSamples = mNormalFrameCount * mChannelCount; 7693 buffer = new int16_t[numSamples]; 7694 memset(buffer, 0, numSamples * sizeof(int16_t)); 7695 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 7696 ownsBuffer = true; 7697 } 7698 7699 // Attach all tracks with same session ID to this chain. 7700 for (size_t i = 0; i < mTracks.size(); ++i) { 7701 sp<Track> track = mTracks[i]; 7702 if (session == track->sessionId()) { 7703 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 7704 track->setMainBuffer(buffer); 7705 chain->incTrackCnt(); 7706 } 7707 } 7708 7709 // indicate all active tracks in the chain 7710 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7711 sp<Track> track = mActiveTracks[i].promote(); 7712 if (track == 0) continue; 7713 if (session == track->sessionId()) { 7714 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 7715 chain->incActiveTrackCnt(); 7716 } 7717 } 7718 } 7719 7720 chain->setInBuffer(buffer, ownsBuffer); 7721 chain->setOutBuffer(mMixBuffer); 7722 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 7723 // chains list in order to be processed last as it contains output stage effects 7724 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 7725 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 7726 // after track specific effects and before output stage 7727 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 7728 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 7729 // Effect chain for other sessions are inserted at beginning of effect 7730 // chains list to be processed before output mix effects. Relative order between other 7731 // sessions is not important 7732 size_t size = mEffectChains.size(); 7733 size_t i = 0; 7734 for (i = 0; i < size; i++) { 7735 if (mEffectChains[i]->sessionId() < session) break; 7736 } 7737 mEffectChains.insertAt(chain, i); 7738 checkSuspendOnAddEffectChain_l(chain); 7739 7740 return NO_ERROR; 7741} 7742 7743size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 7744{ 7745 int session = chain->sessionId(); 7746 7747 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 7748 7749 for (size_t i = 0; i < mEffectChains.size(); i++) { 7750 if (chain == mEffectChains[i]) { 7751 mEffectChains.removeAt(i); 7752 // detach all active tracks from the chain 7753 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 7754 sp<Track> track = mActiveTracks[i].promote(); 7755 if (track == 0) continue; 7756 if (session == track->sessionId()) { 7757 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 7758 chain.get(), session); 7759 chain->decActiveTrackCnt(); 7760 } 7761 } 7762 7763 // detach all tracks with same session ID from this chain 7764 for (size_t i = 0; i < mTracks.size(); ++i) { 7765 sp<Track> track = mTracks[i]; 7766 if (session == track->sessionId()) { 7767 track->setMainBuffer(mMixBuffer); 7768 chain->decTrackCnt(); 7769 } 7770 } 7771 break; 7772 } 7773 } 7774 return mEffectChains.size(); 7775} 7776 7777status_t AudioFlinger::PlaybackThread::attachAuxEffect( 7778 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7779{ 7780 Mutex::Autolock _l(mLock); 7781 return attachAuxEffect_l(track, EffectId); 7782} 7783 7784status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 7785 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 7786{ 7787 status_t status = NO_ERROR; 7788 7789 if (EffectId == 0) { 7790 track->setAuxBuffer(0, NULL); 7791 } else { 7792 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 7793 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 7794 if (effect != 0) { 7795 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7796 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 7797 } else { 7798 status = INVALID_OPERATION; 7799 } 7800 } else { 7801 status = BAD_VALUE; 7802 } 7803 } 7804 return status; 7805} 7806 7807void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 7808{ 7809 for (size_t i = 0; i < mTracks.size(); ++i) { 7810 sp<Track> track = mTracks[i]; 7811 if (track->auxEffectId() == effectId) { 7812 attachAuxEffect_l(track, 0); 7813 } 7814 } 7815} 7816 7817status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 7818{ 7819 // only one chain per input thread 7820 if (mEffectChains.size() != 0) { 7821 return INVALID_OPERATION; 7822 } 7823 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 7824 7825 chain->setInBuffer(NULL); 7826 chain->setOutBuffer(NULL); 7827 7828 checkSuspendOnAddEffectChain_l(chain); 7829 7830 mEffectChains.add(chain); 7831 7832 return NO_ERROR; 7833} 7834 7835size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 7836{ 7837 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 7838 ALOGW_IF(mEffectChains.size() != 1, 7839 "removeEffectChain_l() %p invalid chain size %d on thread %p", 7840 chain.get(), mEffectChains.size(), this); 7841 if (mEffectChains.size() == 1) { 7842 mEffectChains.removeAt(0); 7843 } 7844 return 0; 7845} 7846 7847// ---------------------------------------------------------------------------- 7848// EffectModule implementation 7849// ---------------------------------------------------------------------------- 7850 7851#undef LOG_TAG 7852#define LOG_TAG "AudioFlinger::EffectModule" 7853 7854AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 7855 const wp<AudioFlinger::EffectChain>& chain, 7856 effect_descriptor_t *desc, 7857 int id, 7858 int sessionId) 7859 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 7860 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 7861{ 7862 ALOGV("Constructor %p", this); 7863 int lStatus; 7864 if (thread == NULL) { 7865 return; 7866 } 7867 7868 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 7869 7870 // create effect engine from effect factory 7871 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 7872 7873 if (mStatus != NO_ERROR) { 7874 return; 7875 } 7876 lStatus = init(); 7877 if (lStatus < 0) { 7878 mStatus = lStatus; 7879 goto Error; 7880 } 7881 7882 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 7883 mPinned = true; 7884 } 7885 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 7886 return; 7887Error: 7888 EffectRelease(mEffectInterface); 7889 mEffectInterface = NULL; 7890 ALOGV("Constructor Error %d", mStatus); 7891} 7892 7893AudioFlinger::EffectModule::~EffectModule() 7894{ 7895 ALOGV("Destructor %p", this); 7896 if (mEffectInterface != NULL) { 7897 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7898 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 7899 sp<ThreadBase> thread = mThread.promote(); 7900 if (thread != 0) { 7901 audio_stream_t *stream = thread->stream(); 7902 if (stream != NULL) { 7903 stream->remove_audio_effect(stream, mEffectInterface); 7904 } 7905 } 7906 } 7907 // release effect engine 7908 EffectRelease(mEffectInterface); 7909 } 7910} 7911 7912status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 7913{ 7914 status_t status; 7915 7916 Mutex::Autolock _l(mLock); 7917 int priority = handle->priority(); 7918 size_t size = mHandles.size(); 7919 sp<EffectHandle> h; 7920 size_t i; 7921 for (i = 0; i < size; i++) { 7922 h = mHandles[i].promote(); 7923 if (h == 0) continue; 7924 if (h->priority() <= priority) break; 7925 } 7926 // if inserted in first place, move effect control from previous owner to this handle 7927 if (i == 0) { 7928 bool enabled = false; 7929 if (h != 0) { 7930 enabled = h->enabled(); 7931 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 7932 } 7933 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 7934 status = NO_ERROR; 7935 } else { 7936 status = ALREADY_EXISTS; 7937 } 7938 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 7939 mHandles.insertAt(handle, i); 7940 return status; 7941} 7942 7943size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 7944{ 7945 Mutex::Autolock _l(mLock); 7946 size_t size = mHandles.size(); 7947 size_t i; 7948 for (i = 0; i < size; i++) { 7949 if (mHandles[i] == handle) break; 7950 } 7951 if (i == size) { 7952 return size; 7953 } 7954 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7955 7956 bool enabled = false; 7957 EffectHandle *hdl = handle.unsafe_get(); 7958 if (hdl != NULL) { 7959 ALOGV("removeHandle() unsafe_get OK"); 7960 enabled = hdl->enabled(); 7961 } 7962 mHandles.removeAt(i); 7963 size = mHandles.size(); 7964 // if removed from first place, move effect control from this handle to next in line 7965 if (i == 0 && size != 0) { 7966 sp<EffectHandle> h = mHandles[0].promote(); 7967 if (h != 0) { 7968 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7969 } 7970 } 7971 7972 // Prevent calls to process() and other functions on effect interface from now on. 7973 // The effect engine will be released by the destructor when the last strong reference on 7974 // this object is released which can happen after next process is called. 7975 if (size == 0 && !mPinned) { 7976 mState = DESTROYED; 7977 } 7978 7979 return size; 7980} 7981 7982sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7983{ 7984 Mutex::Autolock _l(mLock); 7985 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7986} 7987 7988void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7989{ 7990 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7991 // keep a strong reference on this EffectModule to avoid calling the 7992 // destructor before we exit 7993 sp<EffectModule> keep(this); 7994 { 7995 sp<ThreadBase> thread = mThread.promote(); 7996 if (thread != 0) { 7997 thread->disconnectEffect(keep, handle, unpinIfLast); 7998 } 7999 } 8000} 8001 8002void AudioFlinger::EffectModule::updateState() { 8003 Mutex::Autolock _l(mLock); 8004 8005 switch (mState) { 8006 case RESTART: 8007 reset_l(); 8008 // FALL THROUGH 8009 8010 case STARTING: 8011 // clear auxiliary effect input buffer for next accumulation 8012 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8013 memset(mConfig.inputCfg.buffer.raw, 8014 0, 8015 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8016 } 8017 start_l(); 8018 mState = ACTIVE; 8019 break; 8020 case STOPPING: 8021 stop_l(); 8022 mDisableWaitCnt = mMaxDisableWaitCnt; 8023 mState = STOPPED; 8024 break; 8025 case STOPPED: 8026 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 8027 // turn off sequence. 8028 if (--mDisableWaitCnt == 0) { 8029 reset_l(); 8030 mState = IDLE; 8031 } 8032 break; 8033 default: //IDLE , ACTIVE, DESTROYED 8034 break; 8035 } 8036} 8037 8038void AudioFlinger::EffectModule::process() 8039{ 8040 Mutex::Autolock _l(mLock); 8041 8042 if (mState == DESTROYED || mEffectInterface == NULL || 8043 mConfig.inputCfg.buffer.raw == NULL || 8044 mConfig.outputCfg.buffer.raw == NULL) { 8045 return; 8046 } 8047 8048 if (isProcessEnabled()) { 8049 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 8050 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8051 ditherAndClamp(mConfig.inputCfg.buffer.s32, 8052 mConfig.inputCfg.buffer.s32, 8053 mConfig.inputCfg.buffer.frameCount/2); 8054 } 8055 8056 // do the actual processing in the effect engine 8057 int ret = (*mEffectInterface)->process(mEffectInterface, 8058 &mConfig.inputCfg.buffer, 8059 &mConfig.outputCfg.buffer); 8060 8061 // force transition to IDLE state when engine is ready 8062 if (mState == STOPPED && ret == -ENODATA) { 8063 mDisableWaitCnt = 1; 8064 } 8065 8066 // clear auxiliary effect input buffer for next accumulation 8067 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8068 memset(mConfig.inputCfg.buffer.raw, 0, 8069 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 8070 } 8071 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 8072 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8073 // If an insert effect is idle and input buffer is different from output buffer, 8074 // accumulate input onto output 8075 sp<EffectChain> chain = mChain.promote(); 8076 if (chain != 0 && chain->activeTrackCnt() != 0) { 8077 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 8078 int16_t *in = mConfig.inputCfg.buffer.s16; 8079 int16_t *out = mConfig.outputCfg.buffer.s16; 8080 for (size_t i = 0; i < frameCnt; i++) { 8081 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 8082 } 8083 } 8084 } 8085} 8086 8087void AudioFlinger::EffectModule::reset_l() 8088{ 8089 if (mEffectInterface == NULL) { 8090 return; 8091 } 8092 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 8093} 8094 8095status_t AudioFlinger::EffectModule::configure() 8096{ 8097 uint32_t channels; 8098 if (mEffectInterface == NULL) { 8099 return NO_INIT; 8100 } 8101 8102 sp<ThreadBase> thread = mThread.promote(); 8103 if (thread == 0) { 8104 return DEAD_OBJECT; 8105 } 8106 8107 // TODO: handle configuration of effects replacing track process 8108 if (thread->channelCount() == 1) { 8109 channels = AUDIO_CHANNEL_OUT_MONO; 8110 } else { 8111 channels = AUDIO_CHANNEL_OUT_STEREO; 8112 } 8113 8114 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8115 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 8116 } else { 8117 mConfig.inputCfg.channels = channels; 8118 } 8119 mConfig.outputCfg.channels = channels; 8120 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8121 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 8122 mConfig.inputCfg.samplingRate = thread->sampleRate(); 8123 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 8124 mConfig.inputCfg.bufferProvider.cookie = NULL; 8125 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 8126 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 8127 mConfig.outputCfg.bufferProvider.cookie = NULL; 8128 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 8129 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 8130 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 8131 // Insert effect: 8132 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 8133 // always overwrites output buffer: input buffer == output buffer 8134 // - in other sessions: 8135 // last effect in the chain accumulates in output buffer: input buffer != output buffer 8136 // other effect: overwrites output buffer: input buffer == output buffer 8137 // Auxiliary effect: 8138 // accumulates in output buffer: input buffer != output buffer 8139 // Therefore: accumulate <=> input buffer != output buffer 8140 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 8141 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 8142 } else { 8143 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 8144 } 8145 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 8146 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 8147 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 8148 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 8149 8150 ALOGV("configure() %p thread %p buffer %p framecount %d", 8151 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 8152 8153 status_t cmdStatus; 8154 uint32_t size = sizeof(int); 8155 status_t status = (*mEffectInterface)->command(mEffectInterface, 8156 EFFECT_CMD_SET_CONFIG, 8157 sizeof(effect_config_t), 8158 &mConfig, 8159 &size, 8160 &cmdStatus); 8161 if (status == 0) { 8162 status = cmdStatus; 8163 } 8164 8165 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 8166 (1000 * mConfig.outputCfg.buffer.frameCount); 8167 8168 return status; 8169} 8170 8171status_t AudioFlinger::EffectModule::init() 8172{ 8173 Mutex::Autolock _l(mLock); 8174 if (mEffectInterface == NULL) { 8175 return NO_INIT; 8176 } 8177 status_t cmdStatus; 8178 uint32_t size = sizeof(status_t); 8179 status_t status = (*mEffectInterface)->command(mEffectInterface, 8180 EFFECT_CMD_INIT, 8181 0, 8182 NULL, 8183 &size, 8184 &cmdStatus); 8185 if (status == 0) { 8186 status = cmdStatus; 8187 } 8188 return status; 8189} 8190 8191status_t AudioFlinger::EffectModule::start() 8192{ 8193 Mutex::Autolock _l(mLock); 8194 return start_l(); 8195} 8196 8197status_t AudioFlinger::EffectModule::start_l() 8198{ 8199 if (mEffectInterface == NULL) { 8200 return NO_INIT; 8201 } 8202 status_t cmdStatus; 8203 uint32_t size = sizeof(status_t); 8204 status_t status = (*mEffectInterface)->command(mEffectInterface, 8205 EFFECT_CMD_ENABLE, 8206 0, 8207 NULL, 8208 &size, 8209 &cmdStatus); 8210 if (status == 0) { 8211 status = cmdStatus; 8212 } 8213 if (status == 0 && 8214 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8215 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8216 sp<ThreadBase> thread = mThread.promote(); 8217 if (thread != 0) { 8218 audio_stream_t *stream = thread->stream(); 8219 if (stream != NULL) { 8220 stream->add_audio_effect(stream, mEffectInterface); 8221 } 8222 } 8223 } 8224 return status; 8225} 8226 8227status_t AudioFlinger::EffectModule::stop() 8228{ 8229 Mutex::Autolock _l(mLock); 8230 return stop_l(); 8231} 8232 8233status_t AudioFlinger::EffectModule::stop_l() 8234{ 8235 if (mEffectInterface == NULL) { 8236 return NO_INIT; 8237 } 8238 status_t cmdStatus; 8239 uint32_t size = sizeof(status_t); 8240 status_t status = (*mEffectInterface)->command(mEffectInterface, 8241 EFFECT_CMD_DISABLE, 8242 0, 8243 NULL, 8244 &size, 8245 &cmdStatus); 8246 if (status == 0) { 8247 status = cmdStatus; 8248 } 8249 if (status == 0 && 8250 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 8251 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 8252 sp<ThreadBase> thread = mThread.promote(); 8253 if (thread != 0) { 8254 audio_stream_t *stream = thread->stream(); 8255 if (stream != NULL) { 8256 stream->remove_audio_effect(stream, mEffectInterface); 8257 } 8258 } 8259 } 8260 return status; 8261} 8262 8263status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 8264 uint32_t cmdSize, 8265 void *pCmdData, 8266 uint32_t *replySize, 8267 void *pReplyData) 8268{ 8269 Mutex::Autolock _l(mLock); 8270// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 8271 8272 if (mState == DESTROYED || mEffectInterface == NULL) { 8273 return NO_INIT; 8274 } 8275 status_t status = (*mEffectInterface)->command(mEffectInterface, 8276 cmdCode, 8277 cmdSize, 8278 pCmdData, 8279 replySize, 8280 pReplyData); 8281 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 8282 uint32_t size = (replySize == NULL) ? 0 : *replySize; 8283 for (size_t i = 1; i < mHandles.size(); i++) { 8284 sp<EffectHandle> h = mHandles[i].promote(); 8285 if (h != 0) { 8286 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 8287 } 8288 } 8289 } 8290 return status; 8291} 8292 8293status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 8294{ 8295 8296 Mutex::Autolock _l(mLock); 8297 ALOGV("setEnabled %p enabled %d", this, enabled); 8298 8299 if (enabled != isEnabled()) { 8300 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 8301 if (enabled && status != NO_ERROR) { 8302 return status; 8303 } 8304 8305 switch (mState) { 8306 // going from disabled to enabled 8307 case IDLE: 8308 mState = STARTING; 8309 break; 8310 case STOPPED: 8311 mState = RESTART; 8312 break; 8313 case STOPPING: 8314 mState = ACTIVE; 8315 break; 8316 8317 // going from enabled to disabled 8318 case RESTART: 8319 mState = STOPPED; 8320 break; 8321 case STARTING: 8322 mState = IDLE; 8323 break; 8324 case ACTIVE: 8325 mState = STOPPING; 8326 break; 8327 case DESTROYED: 8328 return NO_ERROR; // simply ignore as we are being destroyed 8329 } 8330 for (size_t i = 1; i < mHandles.size(); i++) { 8331 sp<EffectHandle> h = mHandles[i].promote(); 8332 if (h != 0) { 8333 h->setEnabled(enabled); 8334 } 8335 } 8336 } 8337 return NO_ERROR; 8338} 8339 8340bool AudioFlinger::EffectModule::isEnabled() const 8341{ 8342 switch (mState) { 8343 case RESTART: 8344 case STARTING: 8345 case ACTIVE: 8346 return true; 8347 case IDLE: 8348 case STOPPING: 8349 case STOPPED: 8350 case DESTROYED: 8351 default: 8352 return false; 8353 } 8354} 8355 8356bool AudioFlinger::EffectModule::isProcessEnabled() const 8357{ 8358 switch (mState) { 8359 case RESTART: 8360 case ACTIVE: 8361 case STOPPING: 8362 case STOPPED: 8363 return true; 8364 case IDLE: 8365 case STARTING: 8366 case DESTROYED: 8367 default: 8368 return false; 8369 } 8370} 8371 8372status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 8373{ 8374 Mutex::Autolock _l(mLock); 8375 status_t status = NO_ERROR; 8376 8377 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 8378 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 8379 if (isProcessEnabled() && 8380 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 8381 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 8382 status_t cmdStatus; 8383 uint32_t volume[2]; 8384 uint32_t *pVolume = NULL; 8385 uint32_t size = sizeof(volume); 8386 volume[0] = *left; 8387 volume[1] = *right; 8388 if (controller) { 8389 pVolume = volume; 8390 } 8391 status = (*mEffectInterface)->command(mEffectInterface, 8392 EFFECT_CMD_SET_VOLUME, 8393 size, 8394 volume, 8395 &size, 8396 pVolume); 8397 if (controller && status == NO_ERROR && size == sizeof(volume)) { 8398 *left = volume[0]; 8399 *right = volume[1]; 8400 } 8401 } 8402 return status; 8403} 8404 8405status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 8406{ 8407 Mutex::Autolock _l(mLock); 8408 status_t status = NO_ERROR; 8409 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 8410 // audio pre processing modules on RecordThread can receive both output and 8411 // input device indication in the same call 8412 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 8413 if (dev) { 8414 status_t cmdStatus; 8415 uint32_t size = sizeof(status_t); 8416 8417 status = (*mEffectInterface)->command(mEffectInterface, 8418 EFFECT_CMD_SET_DEVICE, 8419 sizeof(uint32_t), 8420 &dev, 8421 &size, 8422 &cmdStatus); 8423 if (status == NO_ERROR) { 8424 status = cmdStatus; 8425 } 8426 } 8427 dev = device & AUDIO_DEVICE_IN_ALL; 8428 if (dev) { 8429 status_t cmdStatus; 8430 uint32_t size = sizeof(status_t); 8431 8432 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 8433 EFFECT_CMD_SET_INPUT_DEVICE, 8434 sizeof(uint32_t), 8435 &dev, 8436 &size, 8437 &cmdStatus); 8438 if (status2 == NO_ERROR) { 8439 status2 = cmdStatus; 8440 } 8441 if (status == NO_ERROR) { 8442 status = status2; 8443 } 8444 } 8445 } 8446 return status; 8447} 8448 8449status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 8450{ 8451 Mutex::Autolock _l(mLock); 8452 status_t status = NO_ERROR; 8453 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 8454 status_t cmdStatus; 8455 uint32_t size = sizeof(status_t); 8456 status = (*mEffectInterface)->command(mEffectInterface, 8457 EFFECT_CMD_SET_AUDIO_MODE, 8458 sizeof(audio_mode_t), 8459 &mode, 8460 &size, 8461 &cmdStatus); 8462 if (status == NO_ERROR) { 8463 status = cmdStatus; 8464 } 8465 } 8466 return status; 8467} 8468 8469void AudioFlinger::EffectModule::setSuspended(bool suspended) 8470{ 8471 Mutex::Autolock _l(mLock); 8472 mSuspended = suspended; 8473} 8474 8475bool AudioFlinger::EffectModule::suspended() const 8476{ 8477 Mutex::Autolock _l(mLock); 8478 return mSuspended; 8479} 8480 8481status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 8482{ 8483 const size_t SIZE = 256; 8484 char buffer[SIZE]; 8485 String8 result; 8486 8487 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 8488 result.append(buffer); 8489 8490 bool locked = tryLock(mLock); 8491 // failed to lock - AudioFlinger is probably deadlocked 8492 if (!locked) { 8493 result.append("\t\tCould not lock Fx mutex:\n"); 8494 } 8495 8496 result.append("\t\tSession Status State Engine:\n"); 8497 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 8498 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 8499 result.append(buffer); 8500 8501 result.append("\t\tDescriptor:\n"); 8502 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8503 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 8504 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 8505 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 8506 result.append(buffer); 8507 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 8508 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 8509 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 8510 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 8511 result.append(buffer); 8512 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 8513 mDescriptor.apiVersion, 8514 mDescriptor.flags); 8515 result.append(buffer); 8516 snprintf(buffer, SIZE, "\t\t- name: %s\n", 8517 mDescriptor.name); 8518 result.append(buffer); 8519 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 8520 mDescriptor.implementor); 8521 result.append(buffer); 8522 8523 result.append("\t\t- Input configuration:\n"); 8524 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8525 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8526 (uint32_t)mConfig.inputCfg.buffer.raw, 8527 mConfig.inputCfg.buffer.frameCount, 8528 mConfig.inputCfg.samplingRate, 8529 mConfig.inputCfg.channels, 8530 mConfig.inputCfg.format); 8531 result.append(buffer); 8532 8533 result.append("\t\t- Output configuration:\n"); 8534 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 8535 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 8536 (uint32_t)mConfig.outputCfg.buffer.raw, 8537 mConfig.outputCfg.buffer.frameCount, 8538 mConfig.outputCfg.samplingRate, 8539 mConfig.outputCfg.channels, 8540 mConfig.outputCfg.format); 8541 result.append(buffer); 8542 8543 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 8544 result.append(buffer); 8545 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 8546 for (size_t i = 0; i < mHandles.size(); ++i) { 8547 sp<EffectHandle> handle = mHandles[i].promote(); 8548 if (handle != 0) { 8549 handle->dump(buffer, SIZE); 8550 result.append(buffer); 8551 } 8552 } 8553 8554 result.append("\n"); 8555 8556 write(fd, result.string(), result.length()); 8557 8558 if (locked) { 8559 mLock.unlock(); 8560 } 8561 8562 return NO_ERROR; 8563} 8564 8565// ---------------------------------------------------------------------------- 8566// EffectHandle implementation 8567// ---------------------------------------------------------------------------- 8568 8569#undef LOG_TAG 8570#define LOG_TAG "AudioFlinger::EffectHandle" 8571 8572AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 8573 const sp<AudioFlinger::Client>& client, 8574 const sp<IEffectClient>& effectClient, 8575 int32_t priority) 8576 : BnEffect(), 8577 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 8578 mPriority(priority), mHasControl(false), mEnabled(false) 8579{ 8580 ALOGV("constructor %p", this); 8581 8582 if (client == 0) { 8583 return; 8584 } 8585 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 8586 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 8587 if (mCblkMemory != 0) { 8588 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 8589 8590 if (mCblk != NULL) { 8591 new(mCblk) effect_param_cblk_t(); 8592 mBuffer = (uint8_t *)mCblk + bufOffset; 8593 } 8594 } else { 8595 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 8596 return; 8597 } 8598} 8599 8600AudioFlinger::EffectHandle::~EffectHandle() 8601{ 8602 ALOGV("Destructor %p", this); 8603 disconnect(false); 8604 ALOGV("Destructor DONE %p", this); 8605} 8606 8607status_t AudioFlinger::EffectHandle::enable() 8608{ 8609 ALOGV("enable %p", this); 8610 if (!mHasControl) return INVALID_OPERATION; 8611 if (mEffect == 0) return DEAD_OBJECT; 8612 8613 if (mEnabled) { 8614 return NO_ERROR; 8615 } 8616 8617 mEnabled = true; 8618 8619 sp<ThreadBase> thread = mEffect->thread().promote(); 8620 if (thread != 0) { 8621 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 8622 } 8623 8624 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 8625 if (mEffect->suspended()) { 8626 return NO_ERROR; 8627 } 8628 8629 status_t status = mEffect->setEnabled(true); 8630 if (status != NO_ERROR) { 8631 if (thread != 0) { 8632 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8633 } 8634 mEnabled = false; 8635 } 8636 return status; 8637} 8638 8639status_t AudioFlinger::EffectHandle::disable() 8640{ 8641 ALOGV("disable %p", this); 8642 if (!mHasControl) return INVALID_OPERATION; 8643 if (mEffect == 0) return DEAD_OBJECT; 8644 8645 if (!mEnabled) { 8646 return NO_ERROR; 8647 } 8648 mEnabled = false; 8649 8650 if (mEffect->suspended()) { 8651 return NO_ERROR; 8652 } 8653 8654 status_t status = mEffect->setEnabled(false); 8655 8656 sp<ThreadBase> thread = mEffect->thread().promote(); 8657 if (thread != 0) { 8658 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8659 } 8660 8661 return status; 8662} 8663 8664void AudioFlinger::EffectHandle::disconnect() 8665{ 8666 disconnect(true); 8667} 8668 8669void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 8670{ 8671 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 8672 if (mEffect == 0) { 8673 return; 8674 } 8675 mEffect->disconnect(this, unpinIfLast); 8676 8677 if (mHasControl && mEnabled) { 8678 sp<ThreadBase> thread = mEffect->thread().promote(); 8679 if (thread != 0) { 8680 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 8681 } 8682 } 8683 8684 // release sp on module => module destructor can be called now 8685 mEffect.clear(); 8686 if (mClient != 0) { 8687 if (mCblk != NULL) { 8688 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 8689 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 8690 } 8691 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 8692 // Client destructor must run with AudioFlinger mutex locked 8693 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 8694 mClient.clear(); 8695 } 8696} 8697 8698status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 8699 uint32_t cmdSize, 8700 void *pCmdData, 8701 uint32_t *replySize, 8702 void *pReplyData) 8703{ 8704// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 8705// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 8706 8707 // only get parameter command is permitted for applications not controlling the effect 8708 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 8709 return INVALID_OPERATION; 8710 } 8711 if (mEffect == 0) return DEAD_OBJECT; 8712 if (mClient == 0) return INVALID_OPERATION; 8713 8714 // handle commands that are not forwarded transparently to effect engine 8715 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 8716 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 8717 // no risk to block the whole media server process or mixer threads is we are stuck here 8718 Mutex::Autolock _l(mCblk->lock); 8719 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 8720 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 8721 mCblk->serverIndex = 0; 8722 mCblk->clientIndex = 0; 8723 return BAD_VALUE; 8724 } 8725 status_t status = NO_ERROR; 8726 while (mCblk->serverIndex < mCblk->clientIndex) { 8727 int reply; 8728 uint32_t rsize = sizeof(int); 8729 int *p = (int *)(mBuffer + mCblk->serverIndex); 8730 int size = *p++; 8731 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 8732 ALOGW("command(): invalid parameter block size"); 8733 break; 8734 } 8735 effect_param_t *param = (effect_param_t *)p; 8736 if (param->psize == 0 || param->vsize == 0) { 8737 ALOGW("command(): null parameter or value size"); 8738 mCblk->serverIndex += size; 8739 continue; 8740 } 8741 uint32_t psize = sizeof(effect_param_t) + 8742 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 8743 param->vsize; 8744 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 8745 psize, 8746 p, 8747 &rsize, 8748 &reply); 8749 // stop at first error encountered 8750 if (ret != NO_ERROR) { 8751 status = ret; 8752 *(int *)pReplyData = reply; 8753 break; 8754 } else if (reply != NO_ERROR) { 8755 *(int *)pReplyData = reply; 8756 break; 8757 } 8758 mCblk->serverIndex += size; 8759 } 8760 mCblk->serverIndex = 0; 8761 mCblk->clientIndex = 0; 8762 return status; 8763 } else if (cmdCode == EFFECT_CMD_ENABLE) { 8764 *(int *)pReplyData = NO_ERROR; 8765 return enable(); 8766 } else if (cmdCode == EFFECT_CMD_DISABLE) { 8767 *(int *)pReplyData = NO_ERROR; 8768 return disable(); 8769 } 8770 8771 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8772} 8773 8774void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 8775{ 8776 ALOGV("setControl %p control %d", this, hasControl); 8777 8778 mHasControl = hasControl; 8779 mEnabled = enabled; 8780 8781 if (signal && mEffectClient != 0) { 8782 mEffectClient->controlStatusChanged(hasControl); 8783 } 8784} 8785 8786void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 8787 uint32_t cmdSize, 8788 void *pCmdData, 8789 uint32_t replySize, 8790 void *pReplyData) 8791{ 8792 if (mEffectClient != 0) { 8793 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 8794 } 8795} 8796 8797 8798 8799void AudioFlinger::EffectHandle::setEnabled(bool enabled) 8800{ 8801 if (mEffectClient != 0) { 8802 mEffectClient->enableStatusChanged(enabled); 8803 } 8804} 8805 8806status_t AudioFlinger::EffectHandle::onTransact( 8807 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8808{ 8809 return BnEffect::onTransact(code, data, reply, flags); 8810} 8811 8812 8813void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 8814{ 8815 bool locked = mCblk != NULL && tryLock(mCblk->lock); 8816 8817 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 8818 (mClient == 0) ? getpid_cached : mClient->pid(), 8819 mPriority, 8820 mHasControl, 8821 !locked, 8822 mCblk ? mCblk->clientIndex : 0, 8823 mCblk ? mCblk->serverIndex : 0 8824 ); 8825 8826 if (locked) { 8827 mCblk->lock.unlock(); 8828 } 8829} 8830 8831#undef LOG_TAG 8832#define LOG_TAG "AudioFlinger::EffectChain" 8833 8834AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 8835 int sessionId) 8836 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 8837 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 8838 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 8839{ 8840 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 8841 if (thread == NULL) { 8842 return; 8843 } 8844 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 8845 thread->frameCount(); 8846} 8847 8848AudioFlinger::EffectChain::~EffectChain() 8849{ 8850 if (mOwnInBuffer) { 8851 delete mInBuffer; 8852 } 8853 8854} 8855 8856// getEffectFromDesc_l() must be called with ThreadBase::mLock held 8857sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 8858{ 8859 size_t size = mEffects.size(); 8860 8861 for (size_t i = 0; i < size; i++) { 8862 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 8863 return mEffects[i]; 8864 } 8865 } 8866 return 0; 8867} 8868 8869// getEffectFromId_l() must be called with ThreadBase::mLock held 8870sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 8871{ 8872 size_t size = mEffects.size(); 8873 8874 for (size_t i = 0; i < size; i++) { 8875 // by convention, return first effect if id provided is 0 (0 is never a valid id) 8876 if (id == 0 || mEffects[i]->id() == id) { 8877 return mEffects[i]; 8878 } 8879 } 8880 return 0; 8881} 8882 8883// getEffectFromType_l() must be called with ThreadBase::mLock held 8884sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 8885 const effect_uuid_t *type) 8886{ 8887 size_t size = mEffects.size(); 8888 8889 for (size_t i = 0; i < size; i++) { 8890 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 8891 return mEffects[i]; 8892 } 8893 } 8894 return 0; 8895} 8896 8897// Must be called with EffectChain::mLock locked 8898void AudioFlinger::EffectChain::process_l() 8899{ 8900 sp<ThreadBase> thread = mThread.promote(); 8901 if (thread == 0) { 8902 ALOGW("process_l(): cannot promote mixer thread"); 8903 return; 8904 } 8905 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 8906 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 8907 // always process effects unless no more tracks are on the session and the effect tail 8908 // has been rendered 8909 bool doProcess = true; 8910 if (!isGlobalSession) { 8911 bool tracksOnSession = (trackCnt() != 0); 8912 8913 if (!tracksOnSession && mTailBufferCount == 0) { 8914 doProcess = false; 8915 } 8916 8917 if (activeTrackCnt() == 0) { 8918 // if no track is active and the effect tail has not been rendered, 8919 // the input buffer must be cleared here as the mixer process will not do it 8920 if (tracksOnSession || mTailBufferCount > 0) { 8921 size_t numSamples = thread->frameCount() * thread->channelCount(); 8922 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 8923 if (mTailBufferCount > 0) { 8924 mTailBufferCount--; 8925 } 8926 } 8927 } 8928 } 8929 8930 size_t size = mEffects.size(); 8931 if (doProcess) { 8932 for (size_t i = 0; i < size; i++) { 8933 mEffects[i]->process(); 8934 } 8935 } 8936 for (size_t i = 0; i < size; i++) { 8937 mEffects[i]->updateState(); 8938 } 8939} 8940 8941// addEffect_l() must be called with PlaybackThread::mLock held 8942status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 8943{ 8944 effect_descriptor_t desc = effect->desc(); 8945 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 8946 8947 Mutex::Autolock _l(mLock); 8948 effect->setChain(this); 8949 sp<ThreadBase> thread = mThread.promote(); 8950 if (thread == 0) { 8951 return NO_INIT; 8952 } 8953 effect->setThread(thread); 8954 8955 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8956 // Auxiliary effects are inserted at the beginning of mEffects vector as 8957 // they are processed first and accumulated in chain input buffer 8958 mEffects.insertAt(effect, 0); 8959 8960 // the input buffer for auxiliary effect contains mono samples in 8961 // 32 bit format. This is to avoid saturation in AudoMixer 8962 // accumulation stage. Saturation is done in EffectModule::process() before 8963 // calling the process in effect engine 8964 size_t numSamples = thread->frameCount(); 8965 int32_t *buffer = new int32_t[numSamples]; 8966 memset(buffer, 0, numSamples * sizeof(int32_t)); 8967 effect->setInBuffer((int16_t *)buffer); 8968 // auxiliary effects output samples to chain input buffer for further processing 8969 // by insert effects 8970 effect->setOutBuffer(mInBuffer); 8971 } else { 8972 // Insert effects are inserted at the end of mEffects vector as they are processed 8973 // after track and auxiliary effects. 8974 // Insert effect order as a function of indicated preference: 8975 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8976 // another effect is present 8977 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8978 // last effect claiming first position 8979 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8980 // first effect claiming last position 8981 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8982 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8983 // already present 8984 8985 size_t size = mEffects.size(); 8986 size_t idx_insert = size; 8987 ssize_t idx_insert_first = -1; 8988 ssize_t idx_insert_last = -1; 8989 8990 for (size_t i = 0; i < size; i++) { 8991 effect_descriptor_t d = mEffects[i]->desc(); 8992 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8993 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8994 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8995 // check invalid effect chaining combinations 8996 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8997 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8998 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8999 return INVALID_OPERATION; 9000 } 9001 // remember position of first insert effect and by default 9002 // select this as insert position for new effect 9003 if (idx_insert == size) { 9004 idx_insert = i; 9005 } 9006 // remember position of last insert effect claiming 9007 // first position 9008 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 9009 idx_insert_first = i; 9010 } 9011 // remember position of first insert effect claiming 9012 // last position 9013 if (iPref == EFFECT_FLAG_INSERT_LAST && 9014 idx_insert_last == -1) { 9015 idx_insert_last = i; 9016 } 9017 } 9018 } 9019 9020 // modify idx_insert from first position if needed 9021 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 9022 if (idx_insert_last != -1) { 9023 idx_insert = idx_insert_last; 9024 } else { 9025 idx_insert = size; 9026 } 9027 } else { 9028 if (idx_insert_first != -1) { 9029 idx_insert = idx_insert_first + 1; 9030 } 9031 } 9032 9033 // always read samples from chain input buffer 9034 effect->setInBuffer(mInBuffer); 9035 9036 // if last effect in the chain, output samples to chain 9037 // output buffer, otherwise to chain input buffer 9038 if (idx_insert == size) { 9039 if (idx_insert != 0) { 9040 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 9041 mEffects[idx_insert-1]->configure(); 9042 } 9043 effect->setOutBuffer(mOutBuffer); 9044 } else { 9045 effect->setOutBuffer(mInBuffer); 9046 } 9047 mEffects.insertAt(effect, idx_insert); 9048 9049 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 9050 } 9051 effect->configure(); 9052 return NO_ERROR; 9053} 9054 9055// removeEffect_l() must be called with PlaybackThread::mLock held 9056size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 9057{ 9058 Mutex::Autolock _l(mLock); 9059 size_t size = mEffects.size(); 9060 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 9061 9062 for (size_t i = 0; i < size; i++) { 9063 if (effect == mEffects[i]) { 9064 // calling stop here will remove pre-processing effect from the audio HAL. 9065 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 9066 // the middle of a read from audio HAL 9067 if (mEffects[i]->state() == EffectModule::ACTIVE || 9068 mEffects[i]->state() == EffectModule::STOPPING) { 9069 mEffects[i]->stop(); 9070 } 9071 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 9072 delete[] effect->inBuffer(); 9073 } else { 9074 if (i == size - 1 && i != 0) { 9075 mEffects[i - 1]->setOutBuffer(mOutBuffer); 9076 mEffects[i - 1]->configure(); 9077 } 9078 } 9079 mEffects.removeAt(i); 9080 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 9081 break; 9082 } 9083 } 9084 9085 return mEffects.size(); 9086} 9087 9088// setDevice_l() must be called with PlaybackThread::mLock held 9089void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 9090{ 9091 size_t size = mEffects.size(); 9092 for (size_t i = 0; i < size; i++) { 9093 mEffects[i]->setDevice(device); 9094 } 9095} 9096 9097// setMode_l() must be called with PlaybackThread::mLock held 9098void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 9099{ 9100 size_t size = mEffects.size(); 9101 for (size_t i = 0; i < size; i++) { 9102 mEffects[i]->setMode(mode); 9103 } 9104} 9105 9106// setVolume_l() must be called with PlaybackThread::mLock held 9107bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 9108{ 9109 uint32_t newLeft = *left; 9110 uint32_t newRight = *right; 9111 bool hasControl = false; 9112 int ctrlIdx = -1; 9113 size_t size = mEffects.size(); 9114 9115 // first update volume controller 9116 for (size_t i = size; i > 0; i--) { 9117 if (mEffects[i - 1]->isProcessEnabled() && 9118 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 9119 ctrlIdx = i - 1; 9120 hasControl = true; 9121 break; 9122 } 9123 } 9124 9125 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 9126 if (hasControl) { 9127 *left = mNewLeftVolume; 9128 *right = mNewRightVolume; 9129 } 9130 return hasControl; 9131 } 9132 9133 mVolumeCtrlIdx = ctrlIdx; 9134 mLeftVolume = newLeft; 9135 mRightVolume = newRight; 9136 9137 // second get volume update from volume controller 9138 if (ctrlIdx >= 0) { 9139 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 9140 mNewLeftVolume = newLeft; 9141 mNewRightVolume = newRight; 9142 } 9143 // then indicate volume to all other effects in chain. 9144 // Pass altered volume to effects before volume controller 9145 // and requested volume to effects after controller 9146 uint32_t lVol = newLeft; 9147 uint32_t rVol = newRight; 9148 9149 for (size_t i = 0; i < size; i++) { 9150 if ((int)i == ctrlIdx) continue; 9151 // this also works for ctrlIdx == -1 when there is no volume controller 9152 if ((int)i > ctrlIdx) { 9153 lVol = *left; 9154 rVol = *right; 9155 } 9156 mEffects[i]->setVolume(&lVol, &rVol, false); 9157 } 9158 *left = newLeft; 9159 *right = newRight; 9160 9161 return hasControl; 9162} 9163 9164status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 9165{ 9166 const size_t SIZE = 256; 9167 char buffer[SIZE]; 9168 String8 result; 9169 9170 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 9171 result.append(buffer); 9172 9173 bool locked = tryLock(mLock); 9174 // failed to lock - AudioFlinger is probably deadlocked 9175 if (!locked) { 9176 result.append("\tCould not lock mutex:\n"); 9177 } 9178 9179 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 9180 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 9181 mEffects.size(), 9182 (uint32_t)mInBuffer, 9183 (uint32_t)mOutBuffer, 9184 mActiveTrackCnt); 9185 result.append(buffer); 9186 write(fd, result.string(), result.size()); 9187 9188 for (size_t i = 0; i < mEffects.size(); ++i) { 9189 sp<EffectModule> effect = mEffects[i]; 9190 if (effect != 0) { 9191 effect->dump(fd, args); 9192 } 9193 } 9194 9195 if (locked) { 9196 mLock.unlock(); 9197 } 9198 9199 return NO_ERROR; 9200} 9201 9202// must be called with ThreadBase::mLock held 9203void AudioFlinger::EffectChain::setEffectSuspended_l( 9204 const effect_uuid_t *type, bool suspend) 9205{ 9206 sp<SuspendedEffectDesc> desc; 9207 // use effect type UUID timelow as key as there is no real risk of identical 9208 // timeLow fields among effect type UUIDs. 9209 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 9210 if (suspend) { 9211 if (index >= 0) { 9212 desc = mSuspendedEffects.valueAt(index); 9213 } else { 9214 desc = new SuspendedEffectDesc(); 9215 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 9216 mSuspendedEffects.add(type->timeLow, desc); 9217 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 9218 } 9219 if (desc->mRefCount++ == 0) { 9220 sp<EffectModule> effect = getEffectIfEnabled(type); 9221 if (effect != 0) { 9222 desc->mEffect = effect; 9223 effect->setSuspended(true); 9224 effect->setEnabled(false); 9225 } 9226 } 9227 } else { 9228 if (index < 0) { 9229 return; 9230 } 9231 desc = mSuspendedEffects.valueAt(index); 9232 if (desc->mRefCount <= 0) { 9233 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 9234 desc->mRefCount = 1; 9235 } 9236 if (--desc->mRefCount == 0) { 9237 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9238 if (desc->mEffect != 0) { 9239 sp<EffectModule> effect = desc->mEffect.promote(); 9240 if (effect != 0) { 9241 effect->setSuspended(false); 9242 sp<EffectHandle> handle = effect->controlHandle(); 9243 if (handle != 0) { 9244 effect->setEnabled(handle->enabled()); 9245 } 9246 } 9247 desc->mEffect.clear(); 9248 } 9249 mSuspendedEffects.removeItemsAt(index); 9250 } 9251 } 9252} 9253 9254// must be called with ThreadBase::mLock held 9255void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 9256{ 9257 sp<SuspendedEffectDesc> desc; 9258 9259 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9260 if (suspend) { 9261 if (index >= 0) { 9262 desc = mSuspendedEffects.valueAt(index); 9263 } else { 9264 desc = new SuspendedEffectDesc(); 9265 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 9266 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 9267 } 9268 if (desc->mRefCount++ == 0) { 9269 Vector< sp<EffectModule> > effects; 9270 getSuspendEligibleEffects(effects); 9271 for (size_t i = 0; i < effects.size(); i++) { 9272 setEffectSuspended_l(&effects[i]->desc().type, true); 9273 } 9274 } 9275 } else { 9276 if (index < 0) { 9277 return; 9278 } 9279 desc = mSuspendedEffects.valueAt(index); 9280 if (desc->mRefCount <= 0) { 9281 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 9282 desc->mRefCount = 1; 9283 } 9284 if (--desc->mRefCount == 0) { 9285 Vector<const effect_uuid_t *> types; 9286 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 9287 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 9288 continue; 9289 } 9290 types.add(&mSuspendedEffects.valueAt(i)->mType); 9291 } 9292 for (size_t i = 0; i < types.size(); i++) { 9293 setEffectSuspended_l(types[i], false); 9294 } 9295 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 9296 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 9297 } 9298 } 9299} 9300 9301 9302// The volume effect is used for automated tests only 9303#ifndef OPENSL_ES_H_ 9304static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 9305 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 9306const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 9307#endif //OPENSL_ES_H_ 9308 9309bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 9310{ 9311 // auxiliary effects and visualizer are never suspended on output mix 9312 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 9313 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 9314 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 9315 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 9316 return false; 9317 } 9318 return true; 9319} 9320 9321void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 9322{ 9323 effects.clear(); 9324 for (size_t i = 0; i < mEffects.size(); i++) { 9325 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 9326 effects.add(mEffects[i]); 9327 } 9328 } 9329} 9330 9331sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 9332 const effect_uuid_t *type) 9333{ 9334 sp<EffectModule> effect = getEffectFromType_l(type); 9335 return effect != 0 && effect->isEnabled() ? effect : 0; 9336} 9337 9338void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 9339 bool enabled) 9340{ 9341 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9342 if (enabled) { 9343 if (index < 0) { 9344 // if the effect is not suspend check if all effects are suspended 9345 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 9346 if (index < 0) { 9347 return; 9348 } 9349 if (!isEffectEligibleForSuspend(effect->desc())) { 9350 return; 9351 } 9352 setEffectSuspended_l(&effect->desc().type, enabled); 9353 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 9354 if (index < 0) { 9355 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 9356 return; 9357 } 9358 } 9359 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 9360 effect->desc().type.timeLow); 9361 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9362 // if effect is requested to suspended but was not yet enabled, supend it now. 9363 if (desc->mEffect == 0) { 9364 desc->mEffect = effect; 9365 effect->setEnabled(false); 9366 effect->setSuspended(true); 9367 } 9368 } else { 9369 if (index < 0) { 9370 return; 9371 } 9372 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 9373 effect->desc().type.timeLow); 9374 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 9375 desc->mEffect.clear(); 9376 effect->setSuspended(false); 9377 } 9378} 9379 9380#undef LOG_TAG 9381#define LOG_TAG "AudioFlinger" 9382 9383// ---------------------------------------------------------------------------- 9384 9385status_t AudioFlinger::onTransact( 9386 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 9387{ 9388 return BnAudioFlinger::onTransact(code, data, reply, flags); 9389} 9390 9391}; // namespace android 9392