AudioFlinger.cpp revision 7d5b26230a179cd7bcc01f6578cd80d8c15a92a5
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 <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75// ---------------------------------------------------------------------------- 76 77 78namespace android { 79 80static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 81static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 82 83static const float MAX_GAIN = 4096.0f; 84static const uint32_t MAX_GAIN_INT = 0x1000; 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95static const int kDumpLockRetries = 50; 96static const int kDumpLockSleepUs = 20000; 97 98// don't warn about blocked writes or record buffer overflows more often than this 99static const nsecs_t kWarningThrottleNs = seconds(5); 100 101// RecordThread loop sleep time upon application overrun or audio HAL read error 102static const int kRecordThreadSleepUs = 5000; 103 104// maximum time to wait for setParameters to complete 105static const nsecs_t kSetParametersTimeoutNs = seconds(2); 106 107// minimum sleep time for the mixer thread loop when tracks are active but in underrun 108static const uint32_t kMinThreadSleepTimeUs = 5000; 109// maximum divider applied to the active sleep time in the mixer thread loop 110static const uint32_t kMaxThreadSleepTimeShift = 2; 111 112nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 113 114// ---------------------------------------------------------------------------- 115 116#ifdef ADD_BATTERY_DATA 117// To collect the amplifier usage 118static void addBatteryData(uint32_t params) { 119 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 120 if (service == NULL) { 121 // it already logged 122 return; 123 } 124 125 service->addBatteryData(params); 126} 127#endif 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152// ---------------------------------------------------------------------------- 153 154AudioFlinger::AudioFlinger() 155 : BnAudioFlinger(), 156 mPrimaryHardwareDev(NULL), 157 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 158 mMasterVolume(1.0f), 159 mMasterVolumeSupportLvl(MVS_NONE), 160 mMasterMute(false), 161 mNextUniqueId(1), 162 mMode(AUDIO_MODE_INVALID), 163 mBtNrecIsOff(false) 164{ 165} 166 167void AudioFlinger::onFirstRef() 168{ 169 int rc = 0; 170 171 Mutex::Autolock _l(mLock); 172 173 /* TODO: move all this work into an Init() function */ 174 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 175 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 176 uint32_t int_val; 177 if (1 == sscanf(val_str, "%u", &int_val)) { 178 mStandbyTimeInNsecs = milliseconds(int_val); 179 ALOGI("Using %u mSec as standby time.", int_val); 180 } else { 181 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 182 ALOGI("Using default %u mSec as standby time.", 183 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 184 } 185 } 186 187 mMode = AUDIO_MODE_NORMAL; 188 mMasterVolumeSW = 1.0; 189 mMasterVolume = 1.0; 190 mHardwareStatus = AUDIO_HW_IDLE; 191} 192 193AudioFlinger::~AudioFlinger() 194{ 195 196 while (!mRecordThreads.isEmpty()) { 197 // closeInput() will remove first entry from mRecordThreads 198 closeInput(mRecordThreads.keyAt(0)); 199 } 200 while (!mPlaybackThreads.isEmpty()) { 201 // closeOutput() will remove first entry from mPlaybackThreads 202 closeOutput(mPlaybackThreads.keyAt(0)); 203 } 204 205 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 206 // no mHardwareLock needed, as there are no other references to this 207 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 208 delete mAudioHwDevs.valueAt(i); 209 } 210} 211 212static const char * const audio_interfaces[] = { 213 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 214 AUDIO_HARDWARE_MODULE_ID_A2DP, 215 AUDIO_HARDWARE_MODULE_ID_USB, 216}; 217#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 218 219audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(audio_module_handle_t module, uint32_t devices) 220{ 221 // if module is 0, the request comes from an old policy manager and we should load 222 // well known modules 223 if (module == 0) { 224 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 225 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 226 loadHwModule_l(audio_interfaces[i]); 227 } 228 } else { 229 // check a match for the requested module handle 230 AudioHwDevice *audioHwdevice = mAudioHwDevs.valueFor(module); 231 if (audioHwdevice != NULL) { 232 return audioHwdevice->hwDevice(); 233 } 234 } 235 // then try to find a module supporting the requested device. 236 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 237 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 238 if ((dev->get_supported_devices(dev) & devices) == devices) 239 return dev; 240 } 241 242 return NULL; 243} 244 245status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 246{ 247 const size_t SIZE = 256; 248 char buffer[SIZE]; 249 String8 result; 250 251 result.append("Clients:\n"); 252 for (size_t i = 0; i < mClients.size(); ++i) { 253 sp<Client> client = mClients.valueAt(i).promote(); 254 if (client != 0) { 255 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 256 result.append(buffer); 257 } 258 } 259 260 result.append("Global session refs:\n"); 261 result.append(" session pid count\n"); 262 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 263 AudioSessionRef *r = mAudioSessionRefs[i]; 264 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 265 result.append(buffer); 266 } 267 write(fd, result.string(), result.size()); 268 return NO_ERROR; 269} 270 271 272status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 273{ 274 const size_t SIZE = 256; 275 char buffer[SIZE]; 276 String8 result; 277 hardware_call_state hardwareStatus = mHardwareStatus; 278 279 snprintf(buffer, SIZE, "Hardware status: %d\n" 280 "Standby Time mSec: %u\n", 281 hardwareStatus, 282 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 283 result.append(buffer); 284 write(fd, result.string(), result.size()); 285 return NO_ERROR; 286} 287 288status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 289{ 290 const size_t SIZE = 256; 291 char buffer[SIZE]; 292 String8 result; 293 snprintf(buffer, SIZE, "Permission Denial: " 294 "can't dump AudioFlinger from pid=%d, uid=%d\n", 295 IPCThreadState::self()->getCallingPid(), 296 IPCThreadState::self()->getCallingUid()); 297 result.append(buffer); 298 write(fd, result.string(), result.size()); 299 return NO_ERROR; 300} 301 302static bool tryLock(Mutex& mutex) 303{ 304 bool locked = false; 305 for (int i = 0; i < kDumpLockRetries; ++i) { 306 if (mutex.tryLock() == NO_ERROR) { 307 locked = true; 308 break; 309 } 310 usleep(kDumpLockSleepUs); 311 } 312 return locked; 313} 314 315status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 316{ 317 if (!dumpAllowed()) { 318 dumpPermissionDenial(fd, args); 319 } else { 320 // get state of hardware lock 321 bool hardwareLocked = tryLock(mHardwareLock); 322 if (!hardwareLocked) { 323 String8 result(kHardwareLockedString); 324 write(fd, result.string(), result.size()); 325 } else { 326 mHardwareLock.unlock(); 327 } 328 329 bool locked = tryLock(mLock); 330 331 // failed to lock - AudioFlinger is probably deadlocked 332 if (!locked) { 333 String8 result(kDeadlockedString); 334 write(fd, result.string(), result.size()); 335 } 336 337 dumpClients(fd, args); 338 dumpInternals(fd, args); 339 340 // dump playback threads 341 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 342 mPlaybackThreads.valueAt(i)->dump(fd, args); 343 } 344 345 // dump record threads 346 for (size_t i = 0; i < mRecordThreads.size(); i++) { 347 mRecordThreads.valueAt(i)->dump(fd, args); 348 } 349 350 // dump all hardware devs 351 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 352 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 353 dev->dump(dev, fd); 354 } 355 if (locked) mLock.unlock(); 356 } 357 return NO_ERROR; 358} 359 360sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 361{ 362 // If pid is already in the mClients wp<> map, then use that entry 363 // (for which promote() is always != 0), otherwise create a new entry and Client. 364 sp<Client> client = mClients.valueFor(pid).promote(); 365 if (client == 0) { 366 client = new Client(this, pid); 367 mClients.add(pid, client); 368 } 369 370 return client; 371} 372 373// IAudioFlinger interface 374 375 376sp<IAudioTrack> AudioFlinger::createTrack( 377 pid_t pid, 378 audio_stream_type_t streamType, 379 uint32_t sampleRate, 380 audio_format_t format, 381 uint32_t channelMask, 382 int frameCount, 383 IAudioFlinger::track_flags_t flags, 384 const sp<IMemory>& sharedBuffer, 385 audio_io_handle_t output, 386 int *sessionId, 387 status_t *status) 388{ 389 sp<PlaybackThread::Track> track; 390 sp<TrackHandle> trackHandle; 391 sp<Client> client; 392 status_t lStatus; 393 int lSessionId; 394 395 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 396 // but if someone uses binder directly they could bypass that and cause us to crash 397 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 398 ALOGE("createTrack() invalid stream type %d", streamType); 399 lStatus = BAD_VALUE; 400 goto Exit; 401 } 402 403 { 404 Mutex::Autolock _l(mLock); 405 PlaybackThread *thread = checkPlaybackThread_l(output); 406 PlaybackThread *effectThread = NULL; 407 if (thread == NULL) { 408 ALOGE("unknown output thread"); 409 lStatus = BAD_VALUE; 410 goto Exit; 411 } 412 413 client = registerPid_l(pid); 414 415 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 416 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 417 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 418 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 419 if (mPlaybackThreads.keyAt(i) != output) { 420 // prevent same audio session on different output threads 421 uint32_t sessions = t->hasAudioSession(*sessionId); 422 if (sessions & PlaybackThread::TRACK_SESSION) { 423 ALOGE("createTrack() session ID %d already in use", *sessionId); 424 lStatus = BAD_VALUE; 425 goto Exit; 426 } 427 // check if an effect with same session ID is waiting for a track to be created 428 if (sessions & PlaybackThread::EFFECT_SESSION) { 429 effectThread = t.get(); 430 } 431 } 432 } 433 lSessionId = *sessionId; 434 } else { 435 // if no audio session id is provided, create one here 436 lSessionId = nextUniqueId(); 437 if (sessionId != NULL) { 438 *sessionId = lSessionId; 439 } 440 } 441 ALOGV("createTrack() lSessionId: %d", lSessionId); 442 443 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 444 track = thread->createTrack_l(client, streamType, sampleRate, format, 445 channelMask, frameCount, sharedBuffer, lSessionId, flags, &lStatus); 446 447 // move effect chain to this output thread if an effect on same session was waiting 448 // for a track to be created 449 if (lStatus == NO_ERROR && effectThread != NULL) { 450 Mutex::Autolock _dl(thread->mLock); 451 Mutex::Autolock _sl(effectThread->mLock); 452 moveEffectChain_l(lSessionId, effectThread, thread, true); 453 } 454 455 // Look for sync events awaiting for a session to be used. 456 for (int i = 0; i < (int)mPendingSyncEvents.size(); i++) { 457 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 458 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 459 track->setSyncEvent(mPendingSyncEvents[i]); 460 mPendingSyncEvents.removeAt(i); 461 i--; 462 } 463 } 464 } 465 } 466 if (lStatus == NO_ERROR) { 467 trackHandle = new TrackHandle(track); 468 } else { 469 // remove local strong reference to Client before deleting the Track so that the Client 470 // destructor is called by the TrackBase destructor with mLock held 471 client.clear(); 472 track.clear(); 473 } 474 475Exit: 476 if (status != NULL) { 477 *status = lStatus; 478 } 479 return trackHandle; 480} 481 482uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 483{ 484 Mutex::Autolock _l(mLock); 485 PlaybackThread *thread = checkPlaybackThread_l(output); 486 if (thread == NULL) { 487 ALOGW("sampleRate() unknown thread %d", output); 488 return 0; 489 } 490 return thread->sampleRate(); 491} 492 493int AudioFlinger::channelCount(audio_io_handle_t output) const 494{ 495 Mutex::Autolock _l(mLock); 496 PlaybackThread *thread = checkPlaybackThread_l(output); 497 if (thread == NULL) { 498 ALOGW("channelCount() unknown thread %d", output); 499 return 0; 500 } 501 return thread->channelCount(); 502} 503 504audio_format_t AudioFlinger::format(audio_io_handle_t output) const 505{ 506 Mutex::Autolock _l(mLock); 507 PlaybackThread *thread = checkPlaybackThread_l(output); 508 if (thread == NULL) { 509 ALOGW("format() unknown thread %d", output); 510 return AUDIO_FORMAT_INVALID; 511 } 512 return thread->format(); 513} 514 515size_t AudioFlinger::frameCount(audio_io_handle_t output) const 516{ 517 Mutex::Autolock _l(mLock); 518 PlaybackThread *thread = checkPlaybackThread_l(output); 519 if (thread == NULL) { 520 ALOGW("frameCount() unknown thread %d", output); 521 return 0; 522 } 523 return thread->frameCount(); 524} 525 526uint32_t AudioFlinger::latency(audio_io_handle_t output) const 527{ 528 Mutex::Autolock _l(mLock); 529 PlaybackThread *thread = checkPlaybackThread_l(output); 530 if (thread == NULL) { 531 ALOGW("latency() unknown thread %d", output); 532 return 0; 533 } 534 return thread->latency(); 535} 536 537status_t AudioFlinger::setMasterVolume(float value) 538{ 539 status_t ret = initCheck(); 540 if (ret != NO_ERROR) { 541 return ret; 542 } 543 544 // check calling permissions 545 if (!settingsAllowed()) { 546 return PERMISSION_DENIED; 547 } 548 549 float swmv = value; 550 551 Mutex::Autolock _l(mLock); 552 553 // when hw supports master volume, don't scale in sw mixer 554 if (MVS_NONE != mMasterVolumeSupportLvl) { 555 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 556 AutoMutex lock(mHardwareLock); 557 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 558 559 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 560 if (NULL != dev->set_master_volume) { 561 dev->set_master_volume(dev, value); 562 } 563 mHardwareStatus = AUDIO_HW_IDLE; 564 } 565 566 swmv = 1.0; 567 } 568 569 mMasterVolume = value; 570 mMasterVolumeSW = swmv; 571 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 572 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 573 574 return NO_ERROR; 575} 576 577status_t AudioFlinger::setMode(audio_mode_t mode) 578{ 579 status_t ret = initCheck(); 580 if (ret != NO_ERROR) { 581 return ret; 582 } 583 584 // check calling permissions 585 if (!settingsAllowed()) { 586 return PERMISSION_DENIED; 587 } 588 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 589 ALOGW("Illegal value: setMode(%d)", mode); 590 return BAD_VALUE; 591 } 592 593 { // scope for the lock 594 AutoMutex lock(mHardwareLock); 595 mHardwareStatus = AUDIO_HW_SET_MODE; 596 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 597 mHardwareStatus = AUDIO_HW_IDLE; 598 } 599 600 if (NO_ERROR == ret) { 601 Mutex::Autolock _l(mLock); 602 mMode = mode; 603 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 604 mPlaybackThreads.valueAt(i)->setMode(mode); 605 } 606 607 return ret; 608} 609 610status_t AudioFlinger::setMicMute(bool state) 611{ 612 status_t ret = initCheck(); 613 if (ret != NO_ERROR) { 614 return ret; 615 } 616 617 // check calling permissions 618 if (!settingsAllowed()) { 619 return PERMISSION_DENIED; 620 } 621 622 AutoMutex lock(mHardwareLock); 623 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 624 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 625 mHardwareStatus = AUDIO_HW_IDLE; 626 return ret; 627} 628 629bool AudioFlinger::getMicMute() const 630{ 631 status_t ret = initCheck(); 632 if (ret != NO_ERROR) { 633 return false; 634 } 635 636 bool state = AUDIO_MODE_INVALID; 637 AutoMutex lock(mHardwareLock); 638 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 639 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 640 mHardwareStatus = AUDIO_HW_IDLE; 641 return state; 642} 643 644status_t AudioFlinger::setMasterMute(bool muted) 645{ 646 // check calling permissions 647 if (!settingsAllowed()) { 648 return PERMISSION_DENIED; 649 } 650 651 Mutex::Autolock _l(mLock); 652 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 653 mMasterMute = muted; 654 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 655 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 656 657 return NO_ERROR; 658} 659 660float AudioFlinger::masterVolume() const 661{ 662 Mutex::Autolock _l(mLock); 663 return masterVolume_l(); 664} 665 666float AudioFlinger::masterVolumeSW() const 667{ 668 Mutex::Autolock _l(mLock); 669 return masterVolumeSW_l(); 670} 671 672bool AudioFlinger::masterMute() const 673{ 674 Mutex::Autolock _l(mLock); 675 return masterMute_l(); 676} 677 678float AudioFlinger::masterVolume_l() const 679{ 680 if (MVS_FULL == mMasterVolumeSupportLvl) { 681 float ret_val; 682 AutoMutex lock(mHardwareLock); 683 684 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 685 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 686 (NULL != mPrimaryHardwareDev->get_master_volume), 687 "can't get master volume"); 688 689 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 690 mHardwareStatus = AUDIO_HW_IDLE; 691 return ret_val; 692 } 693 694 return mMasterVolume; 695} 696 697status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 698 audio_io_handle_t output) 699{ 700 // check calling permissions 701 if (!settingsAllowed()) { 702 return PERMISSION_DENIED; 703 } 704 705 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 706 ALOGE("setStreamVolume() invalid stream %d", stream); 707 return BAD_VALUE; 708 } 709 710 AutoMutex lock(mLock); 711 PlaybackThread *thread = NULL; 712 if (output) { 713 thread = checkPlaybackThread_l(output); 714 if (thread == NULL) { 715 return BAD_VALUE; 716 } 717 } 718 719 mStreamTypes[stream].volume = value; 720 721 if (thread == NULL) { 722 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 723 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 724 } 725 } else { 726 thread->setStreamVolume(stream, value); 727 } 728 729 return NO_ERROR; 730} 731 732status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 733{ 734 // check calling permissions 735 if (!settingsAllowed()) { 736 return PERMISSION_DENIED; 737 } 738 739 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 740 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 741 ALOGE("setStreamMute() invalid stream %d", stream); 742 return BAD_VALUE; 743 } 744 745 AutoMutex lock(mLock); 746 mStreamTypes[stream].mute = muted; 747 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 748 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 749 750 return NO_ERROR; 751} 752 753float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 754{ 755 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 756 return 0.0f; 757 } 758 759 AutoMutex lock(mLock); 760 float volume; 761 if (output) { 762 PlaybackThread *thread = checkPlaybackThread_l(output); 763 if (thread == NULL) { 764 return 0.0f; 765 } 766 volume = thread->streamVolume(stream); 767 } else { 768 volume = streamVolume_l(stream); 769 } 770 771 return volume; 772} 773 774bool AudioFlinger::streamMute(audio_stream_type_t stream) const 775{ 776 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 777 return true; 778 } 779 780 AutoMutex lock(mLock); 781 return streamMute_l(stream); 782} 783 784status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 785{ 786 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 787 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 788 // check calling permissions 789 if (!settingsAllowed()) { 790 return PERMISSION_DENIED; 791 } 792 793 // ioHandle == 0 means the parameters are global to the audio hardware interface 794 if (ioHandle == 0) { 795 Mutex::Autolock _l(mLock); 796 status_t final_result = NO_ERROR; 797 { 798 AutoMutex lock(mHardwareLock); 799 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 800 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 801 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 802 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 803 final_result = result ?: final_result; 804 } 805 mHardwareStatus = AUDIO_HW_IDLE; 806 } 807 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 808 AudioParameter param = AudioParameter(keyValuePairs); 809 String8 value; 810 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 811 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 812 if (mBtNrecIsOff != btNrecIsOff) { 813 for (size_t i = 0; i < mRecordThreads.size(); i++) { 814 sp<RecordThread> thread = mRecordThreads.valueAt(i); 815 RecordThread::RecordTrack *track = thread->track(); 816 if (track != NULL) { 817 audio_devices_t device = (audio_devices_t)( 818 thread->device() & AUDIO_DEVICE_IN_ALL); 819 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 820 thread->setEffectSuspended(FX_IID_AEC, 821 suspend, 822 track->sessionId()); 823 thread->setEffectSuspended(FX_IID_NS, 824 suspend, 825 track->sessionId()); 826 } 827 } 828 mBtNrecIsOff = btNrecIsOff; 829 } 830 } 831 return final_result; 832 } 833 834 // hold a strong ref on thread in case closeOutput() or closeInput() is called 835 // and the thread is exited once the lock is released 836 sp<ThreadBase> thread; 837 { 838 Mutex::Autolock _l(mLock); 839 thread = checkPlaybackThread_l(ioHandle); 840 if (thread == NULL) { 841 thread = checkRecordThread_l(ioHandle); 842 } else if (thread == primaryPlaybackThread_l()) { 843 // indicate output device change to all input threads for pre processing 844 AudioParameter param = AudioParameter(keyValuePairs); 845 int value; 846 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 847 (value != 0)) { 848 for (size_t i = 0; i < mRecordThreads.size(); i++) { 849 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 850 } 851 } 852 } 853 } 854 if (thread != 0) { 855 return thread->setParameters(keyValuePairs); 856 } 857 return BAD_VALUE; 858} 859 860String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 861{ 862// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 863// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 864 865 Mutex::Autolock _l(mLock); 866 867 if (ioHandle == 0) { 868 String8 out_s8; 869 870 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 871 char *s; 872 { 873 AutoMutex lock(mHardwareLock); 874 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 875 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 876 s = dev->get_parameters(dev, keys.string()); 877 mHardwareStatus = AUDIO_HW_IDLE; 878 } 879 out_s8 += String8(s ? s : ""); 880 free(s); 881 } 882 return out_s8; 883 } 884 885 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 886 if (playbackThread != NULL) { 887 return playbackThread->getParameters(keys); 888 } 889 RecordThread *recordThread = checkRecordThread_l(ioHandle); 890 if (recordThread != NULL) { 891 return recordThread->getParameters(keys); 892 } 893 return String8(""); 894} 895 896size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 897{ 898 status_t ret = initCheck(); 899 if (ret != NO_ERROR) { 900 return 0; 901 } 902 903 AutoMutex lock(mHardwareLock); 904 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 905 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 906 mHardwareStatus = AUDIO_HW_IDLE; 907 return size; 908} 909 910unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 911{ 912 if (ioHandle == 0) { 913 return 0; 914 } 915 916 Mutex::Autolock _l(mLock); 917 918 RecordThread *recordThread = checkRecordThread_l(ioHandle); 919 if (recordThread != NULL) { 920 return recordThread->getInputFramesLost(); 921 } 922 return 0; 923} 924 925status_t AudioFlinger::setVoiceVolume(float value) 926{ 927 status_t ret = initCheck(); 928 if (ret != NO_ERROR) { 929 return ret; 930 } 931 932 // check calling permissions 933 if (!settingsAllowed()) { 934 return PERMISSION_DENIED; 935 } 936 937 AutoMutex lock(mHardwareLock); 938 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 939 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 940 mHardwareStatus = AUDIO_HW_IDLE; 941 942 return ret; 943} 944 945status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 946 audio_io_handle_t output) const 947{ 948 status_t status; 949 950 Mutex::Autolock _l(mLock); 951 952 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 953 if (playbackThread != NULL) { 954 return playbackThread->getRenderPosition(halFrames, dspFrames); 955 } 956 957 return BAD_VALUE; 958} 959 960void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 961{ 962 963 Mutex::Autolock _l(mLock); 964 965 pid_t pid = IPCThreadState::self()->getCallingPid(); 966 if (mNotificationClients.indexOfKey(pid) < 0) { 967 sp<NotificationClient> notificationClient = new NotificationClient(this, 968 client, 969 pid); 970 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 971 972 mNotificationClients.add(pid, notificationClient); 973 974 sp<IBinder> binder = client->asBinder(); 975 binder->linkToDeath(notificationClient); 976 977 // the config change is always sent from playback or record threads to avoid deadlock 978 // with AudioSystem::gLock 979 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 980 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 981 } 982 983 for (size_t i = 0; i < mRecordThreads.size(); i++) { 984 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 985 } 986 } 987} 988 989void AudioFlinger::removeNotificationClient(pid_t pid) 990{ 991 Mutex::Autolock _l(mLock); 992 993 mNotificationClients.removeItem(pid); 994 995 ALOGV("%d died, releasing its sessions", pid); 996 size_t num = mAudioSessionRefs.size(); 997 bool removed = false; 998 for (size_t i = 0; i< num; ) { 999 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1000 ALOGV(" pid %d @ %d", ref->mPid, i); 1001 if (ref->mPid == pid) { 1002 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1003 mAudioSessionRefs.removeAt(i); 1004 delete ref; 1005 removed = true; 1006 num--; 1007 } else { 1008 i++; 1009 } 1010 } 1011 if (removed) { 1012 purgeStaleEffects_l(); 1013 } 1014} 1015 1016// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1017void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1018{ 1019 size_t size = mNotificationClients.size(); 1020 for (size_t i = 0; i < size; i++) { 1021 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1022 param2); 1023 } 1024} 1025 1026// removeClient_l() must be called with AudioFlinger::mLock held 1027void AudioFlinger::removeClient_l(pid_t pid) 1028{ 1029 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1030 mClients.removeItem(pid); 1031} 1032 1033 1034// ---------------------------------------------------------------------------- 1035 1036AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1037 uint32_t device, type_t type) 1038 : Thread(false), 1039 mType(type), 1040 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1041 // mChannelMask 1042 mChannelCount(0), 1043 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1044 mParamStatus(NO_ERROR), 1045 mStandby(false), mId(id), 1046 mDevice(device), 1047 mDeathRecipient(new PMDeathRecipient(this)) 1048{ 1049} 1050 1051AudioFlinger::ThreadBase::~ThreadBase() 1052{ 1053 mParamCond.broadcast(); 1054 // do not lock the mutex in destructor 1055 releaseWakeLock_l(); 1056 if (mPowerManager != 0) { 1057 sp<IBinder> binder = mPowerManager->asBinder(); 1058 binder->unlinkToDeath(mDeathRecipient); 1059 } 1060} 1061 1062void AudioFlinger::ThreadBase::exit() 1063{ 1064 ALOGV("ThreadBase::exit"); 1065 { 1066 // This lock prevents the following race in thread (uniprocessor for illustration): 1067 // if (!exitPending()) { 1068 // // context switch from here to exit() 1069 // // exit() calls requestExit(), what exitPending() observes 1070 // // exit() calls signal(), which is dropped since no waiters 1071 // // context switch back from exit() to here 1072 // mWaitWorkCV.wait(...); 1073 // // now thread is hung 1074 // } 1075 AutoMutex lock(mLock); 1076 requestExit(); 1077 mWaitWorkCV.signal(); 1078 } 1079 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1080 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1081 requestExitAndWait(); 1082} 1083 1084status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1085{ 1086 status_t status; 1087 1088 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1089 Mutex::Autolock _l(mLock); 1090 1091 mNewParameters.add(keyValuePairs); 1092 mWaitWorkCV.signal(); 1093 // wait condition with timeout in case the thread loop has exited 1094 // before the request could be processed 1095 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1096 status = mParamStatus; 1097 mWaitWorkCV.signal(); 1098 } else { 1099 status = TIMED_OUT; 1100 } 1101 return status; 1102} 1103 1104void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1105{ 1106 Mutex::Autolock _l(mLock); 1107 sendConfigEvent_l(event, param); 1108} 1109 1110// sendConfigEvent_l() must be called with ThreadBase::mLock held 1111void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1112{ 1113 ConfigEvent configEvent; 1114 configEvent.mEvent = event; 1115 configEvent.mParam = param; 1116 mConfigEvents.add(configEvent); 1117 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1118 mWaitWorkCV.signal(); 1119} 1120 1121void AudioFlinger::ThreadBase::processConfigEvents() 1122{ 1123 mLock.lock(); 1124 while (!mConfigEvents.isEmpty()) { 1125 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1126 ConfigEvent configEvent = mConfigEvents[0]; 1127 mConfigEvents.removeAt(0); 1128 // release mLock before locking AudioFlinger mLock: lock order is always 1129 // AudioFlinger then ThreadBase to avoid cross deadlock 1130 mLock.unlock(); 1131 mAudioFlinger->mLock.lock(); 1132 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1133 mAudioFlinger->mLock.unlock(); 1134 mLock.lock(); 1135 } 1136 mLock.unlock(); 1137} 1138 1139status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1140{ 1141 const size_t SIZE = 256; 1142 char buffer[SIZE]; 1143 String8 result; 1144 1145 bool locked = tryLock(mLock); 1146 if (!locked) { 1147 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1148 write(fd, buffer, strlen(buffer)); 1149 } 1150 1151 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1152 result.append(buffer); 1153 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1154 result.append(buffer); 1155 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1156 result.append(buffer); 1157 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1158 result.append(buffer); 1159 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1160 result.append(buffer); 1161 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1162 result.append(buffer); 1163 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1164 result.append(buffer); 1165 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1166 result.append(buffer); 1167 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1168 result.append(buffer); 1169 1170 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1171 result.append(buffer); 1172 result.append(" Index Command"); 1173 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1174 snprintf(buffer, SIZE, "\n %02d ", i); 1175 result.append(buffer); 1176 result.append(mNewParameters[i]); 1177 } 1178 1179 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1180 result.append(buffer); 1181 snprintf(buffer, SIZE, " Index event param\n"); 1182 result.append(buffer); 1183 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1184 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1185 result.append(buffer); 1186 } 1187 result.append("\n"); 1188 1189 write(fd, result.string(), result.size()); 1190 1191 if (locked) { 1192 mLock.unlock(); 1193 } 1194 return NO_ERROR; 1195} 1196 1197status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1198{ 1199 const size_t SIZE = 256; 1200 char buffer[SIZE]; 1201 String8 result; 1202 1203 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1204 write(fd, buffer, strlen(buffer)); 1205 1206 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1207 sp<EffectChain> chain = mEffectChains[i]; 1208 if (chain != 0) { 1209 chain->dump(fd, args); 1210 } 1211 } 1212 return NO_ERROR; 1213} 1214 1215void AudioFlinger::ThreadBase::acquireWakeLock() 1216{ 1217 Mutex::Autolock _l(mLock); 1218 acquireWakeLock_l(); 1219} 1220 1221void AudioFlinger::ThreadBase::acquireWakeLock_l() 1222{ 1223 if (mPowerManager == 0) { 1224 // use checkService() to avoid blocking if power service is not up yet 1225 sp<IBinder> binder = 1226 defaultServiceManager()->checkService(String16("power")); 1227 if (binder == 0) { 1228 ALOGW("Thread %s cannot connect to the power manager service", mName); 1229 } else { 1230 mPowerManager = interface_cast<IPowerManager>(binder); 1231 binder->linkToDeath(mDeathRecipient); 1232 } 1233 } 1234 if (mPowerManager != 0) { 1235 sp<IBinder> binder = new BBinder(); 1236 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1237 binder, 1238 String16(mName)); 1239 if (status == NO_ERROR) { 1240 mWakeLockToken = binder; 1241 } 1242 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1243 } 1244} 1245 1246void AudioFlinger::ThreadBase::releaseWakeLock() 1247{ 1248 Mutex::Autolock _l(mLock); 1249 releaseWakeLock_l(); 1250} 1251 1252void AudioFlinger::ThreadBase::releaseWakeLock_l() 1253{ 1254 if (mWakeLockToken != 0) { 1255 ALOGV("releaseWakeLock_l() %s", mName); 1256 if (mPowerManager != 0) { 1257 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1258 } 1259 mWakeLockToken.clear(); 1260 } 1261} 1262 1263void AudioFlinger::ThreadBase::clearPowerManager() 1264{ 1265 Mutex::Autolock _l(mLock); 1266 releaseWakeLock_l(); 1267 mPowerManager.clear(); 1268} 1269 1270void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1271{ 1272 sp<ThreadBase> thread = mThread.promote(); 1273 if (thread != 0) { 1274 thread->clearPowerManager(); 1275 } 1276 ALOGW("power manager service died !!!"); 1277} 1278 1279void AudioFlinger::ThreadBase::setEffectSuspended( 1280 const effect_uuid_t *type, bool suspend, int sessionId) 1281{ 1282 Mutex::Autolock _l(mLock); 1283 setEffectSuspended_l(type, suspend, sessionId); 1284} 1285 1286void AudioFlinger::ThreadBase::setEffectSuspended_l( 1287 const effect_uuid_t *type, bool suspend, int sessionId) 1288{ 1289 sp<EffectChain> chain = getEffectChain_l(sessionId); 1290 if (chain != 0) { 1291 if (type != NULL) { 1292 chain->setEffectSuspended_l(type, suspend); 1293 } else { 1294 chain->setEffectSuspendedAll_l(suspend); 1295 } 1296 } 1297 1298 updateSuspendedSessions_l(type, suspend, sessionId); 1299} 1300 1301void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1302{ 1303 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1304 if (index < 0) { 1305 return; 1306 } 1307 1308 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1309 mSuspendedSessions.editValueAt(index); 1310 1311 for (size_t i = 0; i < sessionEffects.size(); i++) { 1312 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1313 for (int j = 0; j < desc->mRefCount; j++) { 1314 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1315 chain->setEffectSuspendedAll_l(true); 1316 } else { 1317 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1318 desc->mType.timeLow); 1319 chain->setEffectSuspended_l(&desc->mType, true); 1320 } 1321 } 1322 } 1323} 1324 1325void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1326 bool suspend, 1327 int sessionId) 1328{ 1329 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1330 1331 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1332 1333 if (suspend) { 1334 if (index >= 0) { 1335 sessionEffects = mSuspendedSessions.editValueAt(index); 1336 } else { 1337 mSuspendedSessions.add(sessionId, sessionEffects); 1338 } 1339 } else { 1340 if (index < 0) { 1341 return; 1342 } 1343 sessionEffects = mSuspendedSessions.editValueAt(index); 1344 } 1345 1346 1347 int key = EffectChain::kKeyForSuspendAll; 1348 if (type != NULL) { 1349 key = type->timeLow; 1350 } 1351 index = sessionEffects.indexOfKey(key); 1352 1353 sp<SuspendedSessionDesc> desc; 1354 if (suspend) { 1355 if (index >= 0) { 1356 desc = sessionEffects.valueAt(index); 1357 } else { 1358 desc = new SuspendedSessionDesc(); 1359 if (type != NULL) { 1360 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1361 } 1362 sessionEffects.add(key, desc); 1363 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1364 } 1365 desc->mRefCount++; 1366 } else { 1367 if (index < 0) { 1368 return; 1369 } 1370 desc = sessionEffects.valueAt(index); 1371 if (--desc->mRefCount == 0) { 1372 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1373 sessionEffects.removeItemsAt(index); 1374 if (sessionEffects.isEmpty()) { 1375 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1376 sessionId); 1377 mSuspendedSessions.removeItem(sessionId); 1378 } 1379 } 1380 } 1381 if (!sessionEffects.isEmpty()) { 1382 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1383 } 1384} 1385 1386void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1387 bool enabled, 1388 int sessionId) 1389{ 1390 Mutex::Autolock _l(mLock); 1391 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1392} 1393 1394void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1395 bool enabled, 1396 int sessionId) 1397{ 1398 if (mType != RECORD) { 1399 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1400 // another session. This gives the priority to well behaved effect control panels 1401 // and applications not using global effects. 1402 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1403 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1404 } 1405 } 1406 1407 sp<EffectChain> chain = getEffectChain_l(sessionId); 1408 if (chain != 0) { 1409 chain->checkSuspendOnEffectEnabled(effect, enabled); 1410 } 1411} 1412 1413// ---------------------------------------------------------------------------- 1414 1415AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1416 AudioStreamOut* output, 1417 audio_io_handle_t id, 1418 uint32_t device, 1419 type_t type) 1420 : ThreadBase(audioFlinger, id, device, type), 1421 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1422 // Assumes constructor is called by AudioFlinger with it's mLock held, 1423 // but it would be safer to explicitly pass initial masterMute as parameter 1424 mMasterMute(audioFlinger->masterMute_l()), 1425 // mStreamTypes[] initialized in constructor body 1426 mOutput(output), 1427 // Assumes constructor is called by AudioFlinger with it's mLock held, 1428 // but it would be safer to explicitly pass initial masterVolume as parameter 1429 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1430 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1431 mMixerStatus(MIXER_IDLE), 1432 mPrevMixerStatus(MIXER_IDLE), 1433 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1434{ 1435 snprintf(mName, kNameLength, "AudioOut_%X", id); 1436 1437 readOutputParameters(); 1438 1439 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1440 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1441 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1442 stream = (audio_stream_type_t) (stream + 1)) { 1443 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1444 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1445 } 1446 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1447 // because mAudioFlinger doesn't have one to copy from 1448} 1449 1450AudioFlinger::PlaybackThread::~PlaybackThread() 1451{ 1452 delete [] mMixBuffer; 1453} 1454 1455status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1456{ 1457 dumpInternals(fd, args); 1458 dumpTracks(fd, args); 1459 dumpEffectChains(fd, args); 1460 return NO_ERROR; 1461} 1462 1463status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1464{ 1465 const size_t SIZE = 256; 1466 char buffer[SIZE]; 1467 String8 result; 1468 1469 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1470 result.append(buffer); 1471 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1472 for (size_t i = 0; i < mTracks.size(); ++i) { 1473 sp<Track> track = mTracks[i]; 1474 if (track != 0) { 1475 track->dump(buffer, SIZE); 1476 result.append(buffer); 1477 } 1478 } 1479 1480 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1481 result.append(buffer); 1482 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1483 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1484 sp<Track> track = mActiveTracks[i].promote(); 1485 if (track != 0) { 1486 track->dump(buffer, SIZE); 1487 result.append(buffer); 1488 } 1489 } 1490 write(fd, result.string(), result.size()); 1491 return NO_ERROR; 1492} 1493 1494status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1495{ 1496 const size_t SIZE = 256; 1497 char buffer[SIZE]; 1498 String8 result; 1499 1500 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1501 result.append(buffer); 1502 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1503 result.append(buffer); 1504 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1505 result.append(buffer); 1506 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1507 result.append(buffer); 1508 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1509 result.append(buffer); 1510 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1511 result.append(buffer); 1512 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1513 result.append(buffer); 1514 write(fd, result.string(), result.size()); 1515 1516 dumpBase(fd, args); 1517 1518 return NO_ERROR; 1519} 1520 1521// Thread virtuals 1522status_t AudioFlinger::PlaybackThread::readyToRun() 1523{ 1524 status_t status = initCheck(); 1525 if (status == NO_ERROR) { 1526 ALOGI("AudioFlinger's thread %p ready to run", this); 1527 } else { 1528 ALOGE("No working audio driver found."); 1529 } 1530 return status; 1531} 1532 1533void AudioFlinger::PlaybackThread::onFirstRef() 1534{ 1535 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1536} 1537 1538// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1539sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1540 const sp<AudioFlinger::Client>& client, 1541 audio_stream_type_t streamType, 1542 uint32_t sampleRate, 1543 audio_format_t format, 1544 uint32_t channelMask, 1545 int frameCount, 1546 const sp<IMemory>& sharedBuffer, 1547 int sessionId, 1548 IAudioFlinger::track_flags_t flags, 1549 status_t *status) 1550{ 1551 sp<Track> track; 1552 status_t lStatus; 1553 1554 bool isTimed = (flags & IAudioFlinger::TRACK_TIMED) != 0; 1555 1556 // client expresses a preference for FAST, but we get the final say 1557 if ((flags & IAudioFlinger::TRACK_FAST) && 1558 !( 1559 // not timed 1560 (!isTimed) && 1561 // either of these use cases: 1562 ( 1563 // use case 1: shared buffer with any frame count 1564 ( 1565 (sharedBuffer != 0) 1566 ) || 1567 // use case 2: callback handler and small power-of-2 frame count 1568 ( 1569 // unfortunately we can't verify that there's a callback until start() 1570 // FIXME supported frame counts should not be hard-coded 1571 ( 1572 (frameCount == 128) || 1573 (frameCount == 256) || 1574 (frameCount == 512) 1575 ) 1576 ) 1577 ) && 1578 // PCM data 1579 audio_is_linear_pcm(format) && 1580 // mono or stereo 1581 ( (channelMask == AUDIO_CHANNEL_OUT_MONO) || 1582 (channelMask == AUDIO_CHANNEL_OUT_STEREO) ) && 1583 // hardware sample rate 1584 (sampleRate == mSampleRate) 1585 // FIXME test that MixerThread for this fast track has a capable output HAL 1586 // FIXME add a permission test also? 1587 ) ) { 1588 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 1589 flags &= ~IAudioFlinger::TRACK_FAST; 1590 } 1591 1592 if (mType == DIRECT) { 1593 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1594 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1595 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1596 "for output %p with format %d", 1597 sampleRate, format, channelMask, mOutput, mFormat); 1598 lStatus = BAD_VALUE; 1599 goto Exit; 1600 } 1601 } 1602 } else { 1603 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1604 if (sampleRate > mSampleRate*2) { 1605 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1606 lStatus = BAD_VALUE; 1607 goto Exit; 1608 } 1609 } 1610 1611 lStatus = initCheck(); 1612 if (lStatus != NO_ERROR) { 1613 ALOGE("Audio driver not initialized."); 1614 goto Exit; 1615 } 1616 1617 { // scope for mLock 1618 Mutex::Autolock _l(mLock); 1619 1620 // all tracks in same audio session must share the same routing strategy otherwise 1621 // conflicts will happen when tracks are moved from one output to another by audio policy 1622 // manager 1623 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1624 for (size_t i = 0; i < mTracks.size(); ++i) { 1625 sp<Track> t = mTracks[i]; 1626 if (t != 0 && !t->isOutputTrack()) { 1627 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1628 if (sessionId == t->sessionId() && strategy != actual) { 1629 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1630 strategy, actual); 1631 lStatus = BAD_VALUE; 1632 goto Exit; 1633 } 1634 } 1635 } 1636 1637 if (!isTimed) { 1638 track = new Track(this, client, streamType, sampleRate, format, 1639 channelMask, frameCount, sharedBuffer, sessionId, flags); 1640 } else { 1641 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1642 channelMask, frameCount, sharedBuffer, sessionId); 1643 } 1644 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1645 lStatus = NO_MEMORY; 1646 goto Exit; 1647 } 1648 mTracks.add(track); 1649 1650 sp<EffectChain> chain = getEffectChain_l(sessionId); 1651 if (chain != 0) { 1652 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1653 track->setMainBuffer(chain->inBuffer()); 1654 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1655 chain->incTrackCnt(); 1656 } 1657 } 1658 lStatus = NO_ERROR; 1659 1660Exit: 1661 if (status) { 1662 *status = lStatus; 1663 } 1664 return track; 1665} 1666 1667uint32_t AudioFlinger::PlaybackThread::latency() const 1668{ 1669 Mutex::Autolock _l(mLock); 1670 if (initCheck() == NO_ERROR) { 1671 return mOutput->stream->get_latency(mOutput->stream); 1672 } else { 1673 return 0; 1674 } 1675} 1676 1677void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1678{ 1679 Mutex::Autolock _l(mLock); 1680 mMasterVolume = value; 1681} 1682 1683void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1684{ 1685 Mutex::Autolock _l(mLock); 1686 setMasterMute_l(muted); 1687} 1688 1689void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1690{ 1691 Mutex::Autolock _l(mLock); 1692 mStreamTypes[stream].volume = value; 1693} 1694 1695void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1696{ 1697 Mutex::Autolock _l(mLock); 1698 mStreamTypes[stream].mute = muted; 1699} 1700 1701float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1702{ 1703 Mutex::Autolock _l(mLock); 1704 return mStreamTypes[stream].volume; 1705} 1706 1707// addTrack_l() must be called with ThreadBase::mLock held 1708status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1709{ 1710 status_t status = ALREADY_EXISTS; 1711 1712 // set retry count for buffer fill 1713 track->mRetryCount = kMaxTrackStartupRetries; 1714 if (mActiveTracks.indexOf(track) < 0) { 1715 // the track is newly added, make sure it fills up all its 1716 // buffers before playing. This is to ensure the client will 1717 // effectively get the latency it requested. 1718 track->mFillingUpStatus = Track::FS_FILLING; 1719 track->mResetDone = false; 1720 mActiveTracks.add(track); 1721 if (track->mainBuffer() != mMixBuffer) { 1722 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1723 if (chain != 0) { 1724 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1725 chain->incActiveTrackCnt(); 1726 } 1727 } 1728 1729 status = NO_ERROR; 1730 } 1731 1732 ALOGV("mWaitWorkCV.broadcast"); 1733 mWaitWorkCV.broadcast(); 1734 1735 return status; 1736} 1737 1738// destroyTrack_l() must be called with ThreadBase::mLock held 1739void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1740{ 1741 track->mState = TrackBase::TERMINATED; 1742 if (mActiveTracks.indexOf(track) < 0) { 1743 removeTrack_l(track); 1744 } 1745} 1746 1747void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1748{ 1749 mTracks.remove(track); 1750 deleteTrackName_l(track->name()); 1751 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1752 if (chain != 0) { 1753 chain->decTrackCnt(); 1754 } 1755} 1756 1757String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1758{ 1759 String8 out_s8 = String8(""); 1760 char *s; 1761 1762 Mutex::Autolock _l(mLock); 1763 if (initCheck() != NO_ERROR) { 1764 return out_s8; 1765 } 1766 1767 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1768 out_s8 = String8(s); 1769 free(s); 1770 return out_s8; 1771} 1772 1773// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1774void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1775 AudioSystem::OutputDescriptor desc; 1776 void *param2 = NULL; 1777 1778 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1779 1780 switch (event) { 1781 case AudioSystem::OUTPUT_OPENED: 1782 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1783 desc.channels = mChannelMask; 1784 desc.samplingRate = mSampleRate; 1785 desc.format = mFormat; 1786 desc.frameCount = mFrameCount; 1787 desc.latency = latency(); 1788 param2 = &desc; 1789 break; 1790 1791 case AudioSystem::STREAM_CONFIG_CHANGED: 1792 param2 = ¶m; 1793 case AudioSystem::OUTPUT_CLOSED: 1794 default: 1795 break; 1796 } 1797 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1798} 1799 1800void AudioFlinger::PlaybackThread::readOutputParameters() 1801{ 1802 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1803 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1804 mChannelCount = (uint16_t)popcount(mChannelMask); 1805 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1806 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1807 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1808 1809 // FIXME - Current mixer implementation only supports stereo output: Always 1810 // Allocate a stereo buffer even if HW output is mono. 1811 delete[] mMixBuffer; 1812 mMixBuffer = new int16_t[mFrameCount * 2]; 1813 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1814 1815 // force reconfiguration of effect chains and engines to take new buffer size and audio 1816 // parameters into account 1817 // Note that mLock is not held when readOutputParameters() is called from the constructor 1818 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1819 // matter. 1820 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1821 Vector< sp<EffectChain> > effectChains = mEffectChains; 1822 for (size_t i = 0; i < effectChains.size(); i ++) { 1823 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1824 } 1825} 1826 1827status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1828{ 1829 if (halFrames == NULL || dspFrames == NULL) { 1830 return BAD_VALUE; 1831 } 1832 Mutex::Autolock _l(mLock); 1833 if (initCheck() != NO_ERROR) { 1834 return INVALID_OPERATION; 1835 } 1836 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1837 1838 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1839} 1840 1841uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1842{ 1843 Mutex::Autolock _l(mLock); 1844 uint32_t result = 0; 1845 if (getEffectChain_l(sessionId) != 0) { 1846 result = EFFECT_SESSION; 1847 } 1848 1849 for (size_t i = 0; i < mTracks.size(); ++i) { 1850 sp<Track> track = mTracks[i]; 1851 if (sessionId == track->sessionId() && 1852 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1853 result |= TRACK_SESSION; 1854 break; 1855 } 1856 } 1857 1858 return result; 1859} 1860 1861uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1862{ 1863 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1864 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1865 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1866 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1867 } 1868 for (size_t i = 0; i < mTracks.size(); i++) { 1869 sp<Track> track = mTracks[i]; 1870 if (sessionId == track->sessionId() && 1871 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1872 return AudioSystem::getStrategyForStream(track->streamType()); 1873 } 1874 } 1875 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1876} 1877 1878 1879AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1880{ 1881 Mutex::Autolock _l(mLock); 1882 return mOutput; 1883} 1884 1885AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1886{ 1887 Mutex::Autolock _l(mLock); 1888 AudioStreamOut *output = mOutput; 1889 mOutput = NULL; 1890 return output; 1891} 1892 1893// this method must always be called either with ThreadBase mLock held or inside the thread loop 1894audio_stream_t* AudioFlinger::PlaybackThread::stream() const 1895{ 1896 if (mOutput == NULL) { 1897 return NULL; 1898 } 1899 return &mOutput->stream->common; 1900} 1901 1902uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() const 1903{ 1904 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1905 // decoding and transfer time. So sleeping for half of the latency would likely cause 1906 // underruns 1907 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1908 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1909 } else { 1910 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1911 } 1912} 1913 1914status_t AudioFlinger::PlaybackThread::setSyncEvent(const sp<SyncEvent>& event) 1915{ 1916 if (!isValidSyncEvent(event)) { 1917 return BAD_VALUE; 1918 } 1919 1920 Mutex::Autolock _l(mLock); 1921 1922 for (size_t i = 0; i < mTracks.size(); ++i) { 1923 sp<Track> track = mTracks[i]; 1924 if (event->triggerSession() == track->sessionId()) { 1925 track->setSyncEvent(event); 1926 return NO_ERROR; 1927 } 1928 } 1929 1930 return NAME_NOT_FOUND; 1931} 1932 1933bool AudioFlinger::PlaybackThread::isValidSyncEvent(const sp<SyncEvent>& event) 1934{ 1935 switch (event->type()) { 1936 case AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE: 1937 return true; 1938 default: 1939 break; 1940 } 1941 return false; 1942} 1943 1944// ---------------------------------------------------------------------------- 1945 1946AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1947 audio_io_handle_t id, uint32_t device, type_t type) 1948 : PlaybackThread(audioFlinger, output, id, device, type) 1949{ 1950 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1951 // FIXME - Current mixer implementation only supports stereo output 1952 if (mChannelCount == 1) { 1953 ALOGE("Invalid audio hardware channel count"); 1954 } 1955} 1956 1957AudioFlinger::MixerThread::~MixerThread() 1958{ 1959 delete mAudioMixer; 1960} 1961 1962class CpuStats { 1963public: 1964 CpuStats(); 1965 void sample(const String8 &title); 1966#ifdef DEBUG_CPU_USAGE 1967private: 1968 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 1969 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 1970 1971 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 1972 1973 int mCpuNum; // thread's current CPU number 1974 int mCpukHz; // frequency of thread's current CPU in kHz 1975#endif 1976}; 1977 1978CpuStats::CpuStats() 1979#ifdef DEBUG_CPU_USAGE 1980 : mCpuNum(-1), mCpukHz(-1) 1981#endif 1982{ 1983} 1984 1985void CpuStats::sample(const String8 &title) { 1986#ifdef DEBUG_CPU_USAGE 1987 // get current thread's delta CPU time in wall clock ns 1988 double wcNs; 1989 bool valid = mCpuUsage.sampleAndEnable(wcNs); 1990 1991 // record sample for wall clock statistics 1992 if (valid) { 1993 mWcStats.sample(wcNs); 1994 } 1995 1996 // get the current CPU number 1997 int cpuNum = sched_getcpu(); 1998 1999 // get the current CPU frequency in kHz 2000 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 2001 2002 // check if either CPU number or frequency changed 2003 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 2004 mCpuNum = cpuNum; 2005 mCpukHz = cpukHz; 2006 // ignore sample for purposes of cycles 2007 valid = false; 2008 } 2009 2010 // if no change in CPU number or frequency, then record sample for cycle statistics 2011 if (valid && mCpukHz > 0) { 2012 double cycles = wcNs * cpukHz * 0.000001; 2013 mHzStats.sample(cycles); 2014 } 2015 2016 unsigned n = mWcStats.n(); 2017 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2018 if ((n & 127) == 1) { 2019 long long elapsed = mCpuUsage.elapsed(); 2020 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2021 double perLoop = elapsed / (double) n; 2022 double perLoop100 = perLoop * 0.01; 2023 double perLoop1k = perLoop * 0.001; 2024 double mean = mWcStats.mean(); 2025 double stddev = mWcStats.stddev(); 2026 double minimum = mWcStats.minimum(); 2027 double maximum = mWcStats.maximum(); 2028 double meanCycles = mHzStats.mean(); 2029 double stddevCycles = mHzStats.stddev(); 2030 double minCycles = mHzStats.minimum(); 2031 double maxCycles = mHzStats.maximum(); 2032 mCpuUsage.resetElapsed(); 2033 mWcStats.reset(); 2034 mHzStats.reset(); 2035 ALOGD("CPU usage for %s over past %.1f secs\n" 2036 " (%u mixer loops at %.1f mean ms per loop):\n" 2037 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2038 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2039 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2040 title.string(), 2041 elapsed * .000000001, n, perLoop * .000001, 2042 mean * .001, 2043 stddev * .001, 2044 minimum * .001, 2045 maximum * .001, 2046 mean / perLoop100, 2047 stddev / perLoop100, 2048 minimum / perLoop100, 2049 maximum / perLoop100, 2050 meanCycles / perLoop1k, 2051 stddevCycles / perLoop1k, 2052 minCycles / perLoop1k, 2053 maxCycles / perLoop1k); 2054 2055 } 2056 } 2057#endif 2058}; 2059 2060void AudioFlinger::PlaybackThread::checkSilentMode_l() 2061{ 2062 if (!mMasterMute) { 2063 char value[PROPERTY_VALUE_MAX]; 2064 if (property_get("ro.audio.silent", value, "0") > 0) { 2065 char *endptr; 2066 unsigned long ul = strtoul(value, &endptr, 0); 2067 if (*endptr == '\0' && ul != 0) { 2068 ALOGD("Silence is golden"); 2069 // The setprop command will not allow a property to be changed after 2070 // the first time it is set, so we don't have to worry about un-muting. 2071 setMasterMute_l(true); 2072 } 2073 } 2074 } 2075} 2076 2077bool AudioFlinger::PlaybackThread::threadLoop() 2078{ 2079 Vector< sp<Track> > tracksToRemove; 2080 2081 standbyTime = systemTime(); 2082 2083 // MIXER 2084 nsecs_t lastWarning = 0; 2085if (mType == MIXER) { 2086 longStandbyExit = false; 2087} 2088 2089 // DUPLICATING 2090 // FIXME could this be made local to while loop? 2091 writeFrames = 0; 2092 2093 cacheParameters_l(); 2094 sleepTime = idleSleepTime; 2095 2096if (mType == MIXER) { 2097 sleepTimeShift = 0; 2098} 2099 2100 CpuStats cpuStats; 2101 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2102 2103 acquireWakeLock(); 2104 2105 while (!exitPending()) 2106 { 2107 cpuStats.sample(myName); 2108 2109 Vector< sp<EffectChain> > effectChains; 2110 2111 processConfigEvents(); 2112 2113 { // scope for mLock 2114 2115 Mutex::Autolock _l(mLock); 2116 2117 if (checkForNewParameters_l()) { 2118 cacheParameters_l(); 2119 } 2120 2121 saveOutputTracks(); 2122 2123 // put audio hardware into standby after short delay 2124 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2125 mSuspended > 0)) { 2126 if (!mStandby) { 2127 2128 threadLoop_standby(); 2129 2130 mStandby = true; 2131 mBytesWritten = 0; 2132 } 2133 2134 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2135 // we're about to wait, flush the binder command buffer 2136 IPCThreadState::self()->flushCommands(); 2137 2138 clearOutputTracks(); 2139 2140 if (exitPending()) break; 2141 2142 releaseWakeLock_l(); 2143 // wait until we have something to do... 2144 ALOGV("%s going to sleep", myName.string()); 2145 mWaitWorkCV.wait(mLock); 2146 ALOGV("%s waking up", myName.string()); 2147 acquireWakeLock_l(); 2148 2149 mPrevMixerStatus = MIXER_IDLE; 2150 2151 checkSilentMode_l(); 2152 2153 standbyTime = systemTime() + standbyDelay; 2154 sleepTime = idleSleepTime; 2155 if (mType == MIXER) { 2156 sleepTimeShift = 0; 2157 } 2158 2159 continue; 2160 } 2161 } 2162 2163 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2164 // Shift in the new status; this could be a queue if it's 2165 // useful to filter the mixer status over several cycles. 2166 mPrevMixerStatus = mMixerStatus; 2167 mMixerStatus = newMixerStatus; 2168 2169 // prevent any changes in effect chain list and in each effect chain 2170 // during mixing and effect process as the audio buffers could be deleted 2171 // or modified if an effect is created or deleted 2172 lockEffectChains_l(effectChains); 2173 } 2174 2175 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2176 threadLoop_mix(); 2177 } else { 2178 threadLoop_sleepTime(); 2179 } 2180 2181 if (mSuspended > 0) { 2182 sleepTime = suspendSleepTimeUs(); 2183 } 2184 2185 // only process effects if we're going to write 2186 if (sleepTime == 0) { 2187 for (size_t i = 0; i < effectChains.size(); i ++) { 2188 effectChains[i]->process_l(); 2189 } 2190 } 2191 2192 // enable changes in effect chain 2193 unlockEffectChains(effectChains); 2194 2195 // sleepTime == 0 means we must write to audio hardware 2196 if (sleepTime == 0) { 2197 2198 threadLoop_write(); 2199 2200if (mType == MIXER) { 2201 // write blocked detection 2202 nsecs_t now = systemTime(); 2203 nsecs_t delta = now - mLastWriteTime; 2204 if (!mStandby && delta > maxPeriod) { 2205 mNumDelayedWrites++; 2206 if ((now - lastWarning) > kWarningThrottleNs) { 2207 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2208 ns2ms(delta), mNumDelayedWrites, this); 2209 lastWarning = now; 2210 } 2211 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2212 // a different threshold. Or completely removed for what it is worth anyway... 2213 if (mStandby) { 2214 longStandbyExit = true; 2215 } 2216 } 2217} 2218 2219 mStandby = false; 2220 } else { 2221 usleep(sleepTime); 2222 } 2223 2224 // finally let go of removed track(s), without the lock held 2225 // since we can't guarantee the destructors won't acquire that 2226 // same lock. 2227 tracksToRemove.clear(); 2228 2229 // FIXME I don't understand the need for this here; 2230 // it was in the original code but maybe the 2231 // assignment in saveOutputTracks() makes this unnecessary? 2232 clearOutputTracks(); 2233 2234 // Effect chains will be actually deleted here if they were removed from 2235 // mEffectChains list during mixing or effects processing 2236 effectChains.clear(); 2237 2238 // FIXME Note that the above .clear() is no longer necessary since effectChains 2239 // is now local to this block, but will keep it for now (at least until merge done). 2240 } 2241 2242if (mType == MIXER || mType == DIRECT) { 2243 // put output stream into standby mode 2244 if (!mStandby) { 2245 mOutput->stream->common.standby(&mOutput->stream->common); 2246 } 2247} 2248if (mType == DUPLICATING) { 2249 // for DuplicatingThread, standby mode is handled by the outputTracks 2250} 2251 2252 releaseWakeLock(); 2253 2254 ALOGV("Thread %p type %d exiting", this, mType); 2255 return false; 2256} 2257 2258// shared by MIXER and DIRECT, overridden by DUPLICATING 2259void AudioFlinger::PlaybackThread::threadLoop_write() 2260{ 2261 // FIXME rewrite to reduce number of system calls 2262 mLastWriteTime = systemTime(); 2263 mInWrite = true; 2264 mBytesWritten += mixBufferSize; 2265 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2266 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2267 mNumWrites++; 2268 mInWrite = false; 2269} 2270 2271// shared by MIXER and DIRECT, overridden by DUPLICATING 2272void AudioFlinger::PlaybackThread::threadLoop_standby() 2273{ 2274 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2275 mOutput->stream->common.standby(&mOutput->stream->common); 2276} 2277 2278void AudioFlinger::MixerThread::threadLoop_mix() 2279{ 2280 // obtain the presentation timestamp of the next output buffer 2281 int64_t pts; 2282 status_t status = INVALID_OPERATION; 2283 2284 if (NULL != mOutput->stream->get_next_write_timestamp) { 2285 status = mOutput->stream->get_next_write_timestamp( 2286 mOutput->stream, &pts); 2287 } 2288 2289 if (status != NO_ERROR) { 2290 pts = AudioBufferProvider::kInvalidPTS; 2291 } 2292 2293 // mix buffers... 2294 mAudioMixer->process(pts); 2295 // increase sleep time progressively when application underrun condition clears. 2296 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2297 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2298 // such that we would underrun the audio HAL. 2299 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2300 sleepTimeShift--; 2301 } 2302 sleepTime = 0; 2303 standbyTime = systemTime() + standbyDelay; 2304 //TODO: delay standby when effects have a tail 2305} 2306 2307void AudioFlinger::MixerThread::threadLoop_sleepTime() 2308{ 2309 // If no tracks are ready, sleep once for the duration of an output 2310 // buffer size, then write 0s to the output 2311 if (sleepTime == 0) { 2312 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2313 sleepTime = activeSleepTime >> sleepTimeShift; 2314 if (sleepTime < kMinThreadSleepTimeUs) { 2315 sleepTime = kMinThreadSleepTimeUs; 2316 } 2317 // reduce sleep time in case of consecutive application underruns to avoid 2318 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2319 // duration we would end up writing less data than needed by the audio HAL if 2320 // the condition persists. 2321 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2322 sleepTimeShift++; 2323 } 2324 } else { 2325 sleepTime = idleSleepTime; 2326 } 2327 } else if (mBytesWritten != 0 || 2328 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2329 memset (mMixBuffer, 0, mixBufferSize); 2330 sleepTime = 0; 2331 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2332 } 2333 // TODO add standby time extension fct of effect tail 2334} 2335 2336// prepareTracks_l() must be called with ThreadBase::mLock held 2337AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2338 Vector< sp<Track> > *tracksToRemove) 2339{ 2340 2341 mixer_state mixerStatus = MIXER_IDLE; 2342 // find out which tracks need to be processed 2343 size_t count = mActiveTracks.size(); 2344 size_t mixedTracks = 0; 2345 size_t tracksWithEffect = 0; 2346 2347 float masterVolume = mMasterVolume; 2348 bool masterMute = mMasterMute; 2349 2350 if (masterMute) { 2351 masterVolume = 0; 2352 } 2353 // Delegate master volume control to effect in output mix effect chain if needed 2354 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2355 if (chain != 0) { 2356 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2357 chain->setVolume_l(&v, &v); 2358 masterVolume = (float)((v + (1 << 23)) >> 24); 2359 chain.clear(); 2360 } 2361 2362 for (size_t i=0 ; i<count ; i++) { 2363 sp<Track> t = mActiveTracks[i].promote(); 2364 if (t == 0) continue; 2365 2366 // this const just means the local variable doesn't change 2367 Track* const track = t.get(); 2368 audio_track_cblk_t* cblk = track->cblk(); 2369 2370 // The first time a track is added we wait 2371 // for all its buffers to be filled before processing it 2372 int name = track->name(); 2373 // make sure that we have enough frames to mix one full buffer. 2374 // enforce this condition only once to enable draining the buffer in case the client 2375 // app does not call stop() and relies on underrun to stop: 2376 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2377 // during last round 2378 uint32_t minFrames = 1; 2379 if (!track->isStopped() && !track->isPausing() && 2380 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2381 if (t->sampleRate() == (int)mSampleRate) { 2382 minFrames = mFrameCount; 2383 } else { 2384 // +1 for rounding and +1 for additional sample needed for interpolation 2385 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2386 // add frames already consumed but not yet released by the resampler 2387 // because cblk->framesReady() will include these frames 2388 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2389 // the minimum track buffer size is normally twice the number of frames necessary 2390 // to fill one buffer and the resampler should not leave more than one buffer worth 2391 // of unreleased frames after each pass, but just in case... 2392 ALOG_ASSERT(minFrames <= cblk->frameCount); 2393 } 2394 } 2395 if ((track->framesReady() >= minFrames) && track->isReady() && 2396 !track->isPaused() && !track->isTerminated()) 2397 { 2398 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2399 2400 mixedTracks++; 2401 2402 // track->mainBuffer() != mMixBuffer means there is an effect chain 2403 // connected to the track 2404 chain.clear(); 2405 if (track->mainBuffer() != mMixBuffer) { 2406 chain = getEffectChain_l(track->sessionId()); 2407 // Delegate volume control to effect in track effect chain if needed 2408 if (chain != 0) { 2409 tracksWithEffect++; 2410 } else { 2411 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2412 name, track->sessionId()); 2413 } 2414 } 2415 2416 2417 int param = AudioMixer::VOLUME; 2418 if (track->mFillingUpStatus == Track::FS_FILLED) { 2419 // no ramp for the first volume setting 2420 track->mFillingUpStatus = Track::FS_ACTIVE; 2421 if (track->mState == TrackBase::RESUMING) { 2422 track->mState = TrackBase::ACTIVE; 2423 param = AudioMixer::RAMP_VOLUME; 2424 } 2425 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2426 } else if (cblk->server != 0) { 2427 // If the track is stopped before the first frame was mixed, 2428 // do not apply ramp 2429 param = AudioMixer::RAMP_VOLUME; 2430 } 2431 2432 // compute volume for this track 2433 uint32_t vl, vr, va; 2434 if (track->isMuted() || track->isPausing() || 2435 mStreamTypes[track->streamType()].mute) { 2436 vl = vr = va = 0; 2437 if (track->isPausing()) { 2438 track->setPaused(); 2439 } 2440 } else { 2441 2442 // read original volumes with volume control 2443 float typeVolume = mStreamTypes[track->streamType()].volume; 2444 float v = masterVolume * typeVolume; 2445 uint32_t vlr = cblk->getVolumeLR(); 2446 vl = vlr & 0xFFFF; 2447 vr = vlr >> 16; 2448 // track volumes come from shared memory, so can't be trusted and must be clamped 2449 if (vl > MAX_GAIN_INT) { 2450 ALOGV("Track left volume out of range: %04X", vl); 2451 vl = MAX_GAIN_INT; 2452 } 2453 if (vr > MAX_GAIN_INT) { 2454 ALOGV("Track right volume out of range: %04X", vr); 2455 vr = MAX_GAIN_INT; 2456 } 2457 // now apply the master volume and stream type volume 2458 vl = (uint32_t)(v * vl) << 12; 2459 vr = (uint32_t)(v * vr) << 12; 2460 // assuming master volume and stream type volume each go up to 1.0, 2461 // vl and vr are now in 8.24 format 2462 2463 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2464 // send level comes from shared memory and so may be corrupt 2465 if (sendLevel > MAX_GAIN_INT) { 2466 ALOGV("Track send level out of range: %04X", sendLevel); 2467 sendLevel = MAX_GAIN_INT; 2468 } 2469 va = (uint32_t)(v * sendLevel); 2470 } 2471 // Delegate volume control to effect in track effect chain if needed 2472 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2473 // Do not ramp volume if volume is controlled by effect 2474 param = AudioMixer::VOLUME; 2475 track->mHasVolumeController = true; 2476 } else { 2477 // force no volume ramp when volume controller was just disabled or removed 2478 // from effect chain to avoid volume spike 2479 if (track->mHasVolumeController) { 2480 param = AudioMixer::VOLUME; 2481 } 2482 track->mHasVolumeController = false; 2483 } 2484 2485 // Convert volumes from 8.24 to 4.12 format 2486 // This additional clamping is needed in case chain->setVolume_l() overshot 2487 vl = (vl + (1 << 11)) >> 12; 2488 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2489 vr = (vr + (1 << 11)) >> 12; 2490 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2491 2492 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2493 2494 // XXX: these things DON'T need to be done each time 2495 mAudioMixer->setBufferProvider(name, track); 2496 mAudioMixer->enable(name); 2497 2498 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2499 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2500 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2501 mAudioMixer->setParameter( 2502 name, 2503 AudioMixer::TRACK, 2504 AudioMixer::FORMAT, (void *)track->format()); 2505 mAudioMixer->setParameter( 2506 name, 2507 AudioMixer::TRACK, 2508 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2509 mAudioMixer->setParameter( 2510 name, 2511 AudioMixer::RESAMPLE, 2512 AudioMixer::SAMPLE_RATE, 2513 (void *)(cblk->sampleRate)); 2514 mAudioMixer->setParameter( 2515 name, 2516 AudioMixer::TRACK, 2517 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2518 mAudioMixer->setParameter( 2519 name, 2520 AudioMixer::TRACK, 2521 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2522 2523 // reset retry count 2524 track->mRetryCount = kMaxTrackRetries; 2525 2526 // If one track is ready, set the mixer ready if: 2527 // - the mixer was not ready during previous round OR 2528 // - no other track is not ready 2529 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2530 mixerStatus != MIXER_TRACKS_ENABLED) { 2531 mixerStatus = MIXER_TRACKS_READY; 2532 } 2533 } else { 2534 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2535 if (track->isStopped()) { 2536 track->reset(); 2537 } 2538 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2539 // We have consumed all the buffers of this track. 2540 // Remove it from the list of active tracks. 2541 // TODO: use actual buffer filling status instead of latency when available from 2542 // audio HAL 2543 size_t audioHALFrames = 2544 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2545 size_t framesWritten = 2546 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2547 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2548 tracksToRemove->add(track); 2549 } 2550 } else { 2551 // No buffers for this track. Give it a few chances to 2552 // fill a buffer, then remove it from active list. 2553 if (--(track->mRetryCount) <= 0) { 2554 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2555 tracksToRemove->add(track); 2556 // indicate to client process that the track was disabled because of underrun 2557 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2558 // If one track is not ready, mark the mixer also not ready if: 2559 // - the mixer was ready during previous round OR 2560 // - no other track is ready 2561 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2562 mixerStatus != MIXER_TRACKS_READY) { 2563 mixerStatus = MIXER_TRACKS_ENABLED; 2564 } 2565 } 2566 mAudioMixer->disable(name); 2567 } 2568 } 2569 2570 // remove all the tracks that need to be... 2571 count = tracksToRemove->size(); 2572 if (CC_UNLIKELY(count)) { 2573 for (size_t i=0 ; i<count ; i++) { 2574 const sp<Track>& track = tracksToRemove->itemAt(i); 2575 mActiveTracks.remove(track); 2576 if (track->mainBuffer() != mMixBuffer) { 2577 chain = getEffectChain_l(track->sessionId()); 2578 if (chain != 0) { 2579 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2580 chain->decActiveTrackCnt(); 2581 } 2582 } 2583 if (track->isTerminated()) { 2584 removeTrack_l(track); 2585 } 2586 } 2587 } 2588 2589 // mix buffer must be cleared if all tracks are connected to an 2590 // effect chain as in this case the mixer will not write to 2591 // mix buffer and track effects will accumulate into it 2592 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2593 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2594 } 2595 2596 return mixerStatus; 2597} 2598 2599/* 2600The derived values that are cached: 2601 - mixBufferSize from frame count * frame size 2602 - activeSleepTime from activeSleepTimeUs() 2603 - idleSleepTime from idleSleepTimeUs() 2604 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2605 - maxPeriod from frame count and sample rate (MIXER only) 2606 2607The parameters that affect these derived values are: 2608 - frame count 2609 - frame size 2610 - sample rate 2611 - device type: A2DP or not 2612 - device latency 2613 - format: PCM or not 2614 - active sleep time 2615 - idle sleep time 2616*/ 2617 2618void AudioFlinger::PlaybackThread::cacheParameters_l() 2619{ 2620 mixBufferSize = mFrameCount * mFrameSize; 2621 activeSleepTime = activeSleepTimeUs(); 2622 idleSleepTime = idleSleepTimeUs(); 2623} 2624 2625void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2626{ 2627 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2628 this, streamType, mTracks.size()); 2629 Mutex::Autolock _l(mLock); 2630 2631 size_t size = mTracks.size(); 2632 for (size_t i = 0; i < size; i++) { 2633 sp<Track> t = mTracks[i]; 2634 if (t->streamType() == streamType) { 2635 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2636 t->mCblk->cv.signal(); 2637 } 2638 } 2639} 2640 2641// getTrackName_l() must be called with ThreadBase::mLock held 2642int AudioFlinger::MixerThread::getTrackName_l(audio_channel_mask_t channelMask) 2643{ 2644 int name = mAudioMixer->getTrackName(); 2645 if (name >= 0) { 2646 mAudioMixer->setParameter(name, 2647 AudioMixer::TRACK, 2648 AudioMixer::CHANNEL_MASK, (void *)channelMask); 2649 } 2650 return name; 2651} 2652 2653// deleteTrackName_l() must be called with ThreadBase::mLock held 2654void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2655{ 2656 ALOGV("remove track (%d) and delete from mixer", name); 2657 mAudioMixer->deleteTrackName(name); 2658} 2659 2660// checkForNewParameters_l() must be called with ThreadBase::mLock held 2661bool AudioFlinger::MixerThread::checkForNewParameters_l() 2662{ 2663 bool reconfig = false; 2664 2665 while (!mNewParameters.isEmpty()) { 2666 status_t status = NO_ERROR; 2667 String8 keyValuePair = mNewParameters[0]; 2668 AudioParameter param = AudioParameter(keyValuePair); 2669 int value; 2670 2671 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2672 reconfig = true; 2673 } 2674 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2675 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2676 status = BAD_VALUE; 2677 } else { 2678 reconfig = true; 2679 } 2680 } 2681 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2682 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2683 status = BAD_VALUE; 2684 } else { 2685 reconfig = true; 2686 } 2687 } 2688 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2689 // do not accept frame count changes if tracks are open as the track buffer 2690 // size depends on frame count and correct behavior would not be guaranteed 2691 // if frame count is changed after track creation 2692 if (!mTracks.isEmpty()) { 2693 status = INVALID_OPERATION; 2694 } else { 2695 reconfig = true; 2696 } 2697 } 2698 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2699#ifdef ADD_BATTERY_DATA 2700 // when changing the audio output device, call addBatteryData to notify 2701 // the change 2702 if ((int)mDevice != value) { 2703 uint32_t params = 0; 2704 // check whether speaker is on 2705 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2706 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2707 } 2708 2709 int deviceWithoutSpeaker 2710 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2711 // check if any other device (except speaker) is on 2712 if (value & deviceWithoutSpeaker ) { 2713 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2714 } 2715 2716 if (params != 0) { 2717 addBatteryData(params); 2718 } 2719 } 2720#endif 2721 2722 // forward device change to effects that have requested to be 2723 // aware of attached audio device. 2724 mDevice = (uint32_t)value; 2725 for (size_t i = 0; i < mEffectChains.size(); i++) { 2726 mEffectChains[i]->setDevice_l(mDevice); 2727 } 2728 } 2729 2730 if (status == NO_ERROR) { 2731 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2732 keyValuePair.string()); 2733 if (!mStandby && status == INVALID_OPERATION) { 2734 mOutput->stream->common.standby(&mOutput->stream->common); 2735 mStandby = true; 2736 mBytesWritten = 0; 2737 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2738 keyValuePair.string()); 2739 } 2740 if (status == NO_ERROR && reconfig) { 2741 delete mAudioMixer; 2742 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2743 mAudioMixer = NULL; 2744 readOutputParameters(); 2745 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2746 for (size_t i = 0; i < mTracks.size() ; i++) { 2747 int name = getTrackName_l((audio_channel_mask_t)mTracks[i]->mChannelMask); 2748 if (name < 0) break; 2749 mTracks[i]->mName = name; 2750 // limit track sample rate to 2 x new output sample rate 2751 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2752 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2753 } 2754 } 2755 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2756 } 2757 } 2758 2759 mNewParameters.removeAt(0); 2760 2761 mParamStatus = status; 2762 mParamCond.signal(); 2763 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2764 // already timed out waiting for the status and will never signal the condition. 2765 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2766 } 2767 return reconfig; 2768} 2769 2770status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2771{ 2772 const size_t SIZE = 256; 2773 char buffer[SIZE]; 2774 String8 result; 2775 2776 PlaybackThread::dumpInternals(fd, args); 2777 2778 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2779 result.append(buffer); 2780 write(fd, result.string(), result.size()); 2781 return NO_ERROR; 2782} 2783 2784uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const 2785{ 2786 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2787} 2788 2789uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() const 2790{ 2791 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2792} 2793 2794void AudioFlinger::MixerThread::cacheParameters_l() 2795{ 2796 PlaybackThread::cacheParameters_l(); 2797 2798 // FIXME: Relaxed timing because of a certain device that can't meet latency 2799 // Should be reduced to 2x after the vendor fixes the driver issue 2800 // increase threshold again due to low power audio mode. The way this warning 2801 // threshold is calculated and its usefulness should be reconsidered anyway. 2802 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2803} 2804 2805// ---------------------------------------------------------------------------- 2806AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2807 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2808 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2809 // mLeftVolFloat, mRightVolFloat 2810 // mLeftVolShort, mRightVolShort 2811{ 2812} 2813 2814AudioFlinger::DirectOutputThread::~DirectOutputThread() 2815{ 2816} 2817 2818AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2819 Vector< sp<Track> > *tracksToRemove 2820) 2821{ 2822 sp<Track> trackToRemove; 2823 2824 mixer_state mixerStatus = MIXER_IDLE; 2825 2826 // find out which tracks need to be processed 2827 if (mActiveTracks.size() != 0) { 2828 sp<Track> t = mActiveTracks[0].promote(); 2829 // The track died recently 2830 if (t == 0) return MIXER_IDLE; 2831 2832 Track* const track = t.get(); 2833 audio_track_cblk_t* cblk = track->cblk(); 2834 2835 // The first time a track is added we wait 2836 // for all its buffers to be filled before processing it 2837 if (cblk->framesReady() && track->isReady() && 2838 !track->isPaused() && !track->isTerminated()) 2839 { 2840 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2841 2842 if (track->mFillingUpStatus == Track::FS_FILLED) { 2843 track->mFillingUpStatus = Track::FS_ACTIVE; 2844 mLeftVolFloat = mRightVolFloat = 0; 2845 mLeftVolShort = mRightVolShort = 0; 2846 if (track->mState == TrackBase::RESUMING) { 2847 track->mState = TrackBase::ACTIVE; 2848 rampVolume = true; 2849 } 2850 } else if (cblk->server != 0) { 2851 // If the track is stopped before the first frame was mixed, 2852 // do not apply ramp 2853 rampVolume = true; 2854 } 2855 // compute volume for this track 2856 float left, right; 2857 if (track->isMuted() || mMasterMute || track->isPausing() || 2858 mStreamTypes[track->streamType()].mute) { 2859 left = right = 0; 2860 if (track->isPausing()) { 2861 track->setPaused(); 2862 } 2863 } else { 2864 float typeVolume = mStreamTypes[track->streamType()].volume; 2865 float v = mMasterVolume * typeVolume; 2866 uint32_t vlr = cblk->getVolumeLR(); 2867 float v_clamped = v * (vlr & 0xFFFF); 2868 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2869 left = v_clamped/MAX_GAIN; 2870 v_clamped = v * (vlr >> 16); 2871 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2872 right = v_clamped/MAX_GAIN; 2873 } 2874 2875 if (left != mLeftVolFloat || right != mRightVolFloat) { 2876 mLeftVolFloat = left; 2877 mRightVolFloat = right; 2878 2879 // If audio HAL implements volume control, 2880 // force software volume to nominal value 2881 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2882 left = 1.0f; 2883 right = 1.0f; 2884 } 2885 2886 // Convert volumes from float to 8.24 2887 uint32_t vl = (uint32_t)(left * (1 << 24)); 2888 uint32_t vr = (uint32_t)(right * (1 << 24)); 2889 2890 // Delegate volume control to effect in track effect chain if needed 2891 // only one effect chain can be present on DirectOutputThread, so if 2892 // there is one, the track is connected to it 2893 if (!mEffectChains.isEmpty()) { 2894 // Do not ramp volume if volume is controlled by effect 2895 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2896 rampVolume = false; 2897 } 2898 } 2899 2900 // Convert volumes from 8.24 to 4.12 format 2901 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2902 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2903 leftVol = (uint16_t)v_clamped; 2904 v_clamped = (vr + (1 << 11)) >> 12; 2905 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2906 rightVol = (uint16_t)v_clamped; 2907 } else { 2908 leftVol = mLeftVolShort; 2909 rightVol = mRightVolShort; 2910 rampVolume = false; 2911 } 2912 2913 // reset retry count 2914 track->mRetryCount = kMaxTrackRetriesDirect; 2915 mActiveTrack = t; 2916 mixerStatus = MIXER_TRACKS_READY; 2917 } else { 2918 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2919 if (track->isStopped()) { 2920 track->reset(); 2921 } 2922 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2923 // We have consumed all the buffers of this track. 2924 // Remove it from the list of active tracks. 2925 // TODO: implement behavior for compressed audio 2926 size_t audioHALFrames = 2927 (mOutput->stream->get_latency(mOutput->stream)*mSampleRate) / 1000; 2928 size_t framesWritten = 2929 mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 2930 if (track->presentationComplete(framesWritten, audioHALFrames)) { 2931 trackToRemove = track; 2932 } 2933 } else { 2934 // No buffers for this track. Give it a few chances to 2935 // fill a buffer, then remove it from active list. 2936 if (--(track->mRetryCount) <= 0) { 2937 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2938 trackToRemove = track; 2939 } else { 2940 mixerStatus = MIXER_TRACKS_ENABLED; 2941 } 2942 } 2943 } 2944 } 2945 2946 // FIXME merge this with similar code for removing multiple tracks 2947 // remove all the tracks that need to be... 2948 if (CC_UNLIKELY(trackToRemove != 0)) { 2949 tracksToRemove->add(trackToRemove); 2950 mActiveTracks.remove(trackToRemove); 2951 if (!mEffectChains.isEmpty()) { 2952 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2953 trackToRemove->sessionId()); 2954 mEffectChains[0]->decActiveTrackCnt(); 2955 } 2956 if (trackToRemove->isTerminated()) { 2957 removeTrack_l(trackToRemove); 2958 } 2959 } 2960 2961 return mixerStatus; 2962} 2963 2964void AudioFlinger::DirectOutputThread::threadLoop_mix() 2965{ 2966 AudioBufferProvider::Buffer buffer; 2967 size_t frameCount = mFrameCount; 2968 int8_t *curBuf = (int8_t *)mMixBuffer; 2969 // output audio to hardware 2970 while (frameCount) { 2971 buffer.frameCount = frameCount; 2972 mActiveTrack->getNextBuffer(&buffer); 2973 if (CC_UNLIKELY(buffer.raw == NULL)) { 2974 memset(curBuf, 0, frameCount * mFrameSize); 2975 break; 2976 } 2977 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2978 frameCount -= buffer.frameCount; 2979 curBuf += buffer.frameCount * mFrameSize; 2980 mActiveTrack->releaseBuffer(&buffer); 2981 } 2982 sleepTime = 0; 2983 standbyTime = systemTime() + standbyDelay; 2984 mActiveTrack.clear(); 2985 2986 // apply volume 2987 2988 // Do not apply volume on compressed audio 2989 if (!audio_is_linear_pcm(mFormat)) { 2990 return; 2991 } 2992 2993 // convert to signed 16 bit before volume calculation 2994 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2995 size_t count = mFrameCount * mChannelCount; 2996 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2997 int16_t *dst = mMixBuffer + count-1; 2998 while (count--) { 2999 *dst-- = (int16_t)(*src--^0x80) << 8; 3000 } 3001 } 3002 3003 frameCount = mFrameCount; 3004 int16_t *out = mMixBuffer; 3005 if (rampVolume) { 3006 if (mChannelCount == 1) { 3007 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3008 int32_t vlInc = d / (int32_t)frameCount; 3009 int32_t vl = ((int32_t)mLeftVolShort << 16); 3010 do { 3011 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3012 out++; 3013 vl += vlInc; 3014 } while (--frameCount); 3015 3016 } else { 3017 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 3018 int32_t vlInc = d / (int32_t)frameCount; 3019 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3020 int32_t vrInc = d / (int32_t)frameCount; 3021 int32_t vl = ((int32_t)mLeftVolShort << 16); 3022 int32_t vr = ((int32_t)mRightVolShort << 16); 3023 do { 3024 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3025 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3026 out += 2; 3027 vl += vlInc; 3028 vr += vrInc; 3029 } while (--frameCount); 3030 } 3031 } else { 3032 if (mChannelCount == 1) { 3033 do { 3034 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3035 out++; 3036 } while (--frameCount); 3037 } else { 3038 do { 3039 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3040 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3041 out += 2; 3042 } while (--frameCount); 3043 } 3044 } 3045 3046 // convert back to unsigned 8 bit after volume calculation 3047 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3048 size_t count = mFrameCount * mChannelCount; 3049 int16_t *src = mMixBuffer; 3050 uint8_t *dst = (uint8_t *)mMixBuffer; 3051 while (count--) { 3052 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3053 } 3054 } 3055 3056 mLeftVolShort = leftVol; 3057 mRightVolShort = rightVol; 3058} 3059 3060void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3061{ 3062 if (sleepTime == 0) { 3063 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3064 sleepTime = activeSleepTime; 3065 } else { 3066 sleepTime = idleSleepTime; 3067 } 3068 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3069 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3070 sleepTime = 0; 3071 } 3072} 3073 3074// getTrackName_l() must be called with ThreadBase::mLock held 3075int AudioFlinger::DirectOutputThread::getTrackName_l(audio_channel_mask_t channelMask) 3076{ 3077 return 0; 3078} 3079 3080// deleteTrackName_l() must be called with ThreadBase::mLock held 3081void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3082{ 3083} 3084 3085// checkForNewParameters_l() must be called with ThreadBase::mLock held 3086bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3087{ 3088 bool reconfig = false; 3089 3090 while (!mNewParameters.isEmpty()) { 3091 status_t status = NO_ERROR; 3092 String8 keyValuePair = mNewParameters[0]; 3093 AudioParameter param = AudioParameter(keyValuePair); 3094 int value; 3095 3096 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3097 // do not accept frame count changes if tracks are open as the track buffer 3098 // size depends on frame count and correct behavior would not be garantied 3099 // if frame count is changed after track creation 3100 if (!mTracks.isEmpty()) { 3101 status = INVALID_OPERATION; 3102 } else { 3103 reconfig = true; 3104 } 3105 } 3106 if (status == NO_ERROR) { 3107 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3108 keyValuePair.string()); 3109 if (!mStandby && status == INVALID_OPERATION) { 3110 mOutput->stream->common.standby(&mOutput->stream->common); 3111 mStandby = true; 3112 mBytesWritten = 0; 3113 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3114 keyValuePair.string()); 3115 } 3116 if (status == NO_ERROR && reconfig) { 3117 readOutputParameters(); 3118 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3119 } 3120 } 3121 3122 mNewParameters.removeAt(0); 3123 3124 mParamStatus = status; 3125 mParamCond.signal(); 3126 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3127 // already timed out waiting for the status and will never signal the condition. 3128 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3129 } 3130 return reconfig; 3131} 3132 3133uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() const 3134{ 3135 uint32_t time; 3136 if (audio_is_linear_pcm(mFormat)) { 3137 time = PlaybackThread::activeSleepTimeUs(); 3138 } else { 3139 time = 10000; 3140 } 3141 return time; 3142} 3143 3144uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() const 3145{ 3146 uint32_t time; 3147 if (audio_is_linear_pcm(mFormat)) { 3148 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3149 } else { 3150 time = 10000; 3151 } 3152 return time; 3153} 3154 3155uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() const 3156{ 3157 uint32_t time; 3158 if (audio_is_linear_pcm(mFormat)) { 3159 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3160 } else { 3161 time = 10000; 3162 } 3163 return time; 3164} 3165 3166void AudioFlinger::DirectOutputThread::cacheParameters_l() 3167{ 3168 PlaybackThread::cacheParameters_l(); 3169 3170 // use shorter standby delay as on normal output to release 3171 // hardware resources as soon as possible 3172 standbyDelay = microseconds(activeSleepTime*2); 3173} 3174 3175// ---------------------------------------------------------------------------- 3176 3177AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3178 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3179 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3180 mWaitTimeMs(UINT_MAX) 3181{ 3182 addOutputTrack(mainThread); 3183} 3184 3185AudioFlinger::DuplicatingThread::~DuplicatingThread() 3186{ 3187 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3188 mOutputTracks[i]->destroy(); 3189 } 3190} 3191 3192void AudioFlinger::DuplicatingThread::threadLoop_mix() 3193{ 3194 // mix buffers... 3195 if (outputsReady(outputTracks)) { 3196 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3197 } else { 3198 memset(mMixBuffer, 0, mixBufferSize); 3199 } 3200 sleepTime = 0; 3201 writeFrames = mFrameCount; 3202} 3203 3204void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3205{ 3206 if (sleepTime == 0) { 3207 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3208 sleepTime = activeSleepTime; 3209 } else { 3210 sleepTime = idleSleepTime; 3211 } 3212 } else if (mBytesWritten != 0) { 3213 // flush remaining overflow buffers in output tracks 3214 for (size_t i = 0; i < outputTracks.size(); i++) { 3215 if (outputTracks[i]->isActive()) { 3216 sleepTime = 0; 3217 writeFrames = 0; 3218 memset(mMixBuffer, 0, mixBufferSize); 3219 break; 3220 } 3221 } 3222 } 3223} 3224 3225void AudioFlinger::DuplicatingThread::threadLoop_write() 3226{ 3227 standbyTime = systemTime() + standbyDelay; 3228 for (size_t i = 0; i < outputTracks.size(); i++) { 3229 outputTracks[i]->write(mMixBuffer, writeFrames); 3230 } 3231 mBytesWritten += mixBufferSize; 3232} 3233 3234void AudioFlinger::DuplicatingThread::threadLoop_standby() 3235{ 3236 // DuplicatingThread implements standby by stopping all tracks 3237 for (size_t i = 0; i < outputTracks.size(); i++) { 3238 outputTracks[i]->stop(); 3239 } 3240} 3241 3242void AudioFlinger::DuplicatingThread::saveOutputTracks() 3243{ 3244 outputTracks = mOutputTracks; 3245} 3246 3247void AudioFlinger::DuplicatingThread::clearOutputTracks() 3248{ 3249 outputTracks.clear(); 3250} 3251 3252void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3253{ 3254 Mutex::Autolock _l(mLock); 3255 // FIXME explain this formula 3256 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3257 OutputTrack *outputTrack = new OutputTrack(thread, 3258 this, 3259 mSampleRate, 3260 mFormat, 3261 mChannelMask, 3262 frameCount); 3263 if (outputTrack->cblk() != NULL) { 3264 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3265 mOutputTracks.add(outputTrack); 3266 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3267 updateWaitTime_l(); 3268 } 3269} 3270 3271void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3272{ 3273 Mutex::Autolock _l(mLock); 3274 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3275 if (mOutputTracks[i]->thread() == thread) { 3276 mOutputTracks[i]->destroy(); 3277 mOutputTracks.removeAt(i); 3278 updateWaitTime_l(); 3279 return; 3280 } 3281 } 3282 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3283} 3284 3285// caller must hold mLock 3286void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3287{ 3288 mWaitTimeMs = UINT_MAX; 3289 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3290 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3291 if (strong != 0) { 3292 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3293 if (waitTimeMs < mWaitTimeMs) { 3294 mWaitTimeMs = waitTimeMs; 3295 } 3296 } 3297 } 3298} 3299 3300 3301bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3302{ 3303 for (size_t i = 0; i < outputTracks.size(); i++) { 3304 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3305 if (thread == 0) { 3306 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3307 return false; 3308 } 3309 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3310 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3311 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3312 return false; 3313 } 3314 } 3315 return true; 3316} 3317 3318uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() const 3319{ 3320 return (mWaitTimeMs * 1000) / 2; 3321} 3322 3323void AudioFlinger::DuplicatingThread::cacheParameters_l() 3324{ 3325 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3326 updateWaitTime_l(); 3327 3328 MixerThread::cacheParameters_l(); 3329} 3330 3331// ---------------------------------------------------------------------------- 3332 3333// TrackBase constructor must be called with AudioFlinger::mLock held 3334AudioFlinger::ThreadBase::TrackBase::TrackBase( 3335 ThreadBase *thread, 3336 const sp<Client>& client, 3337 uint32_t sampleRate, 3338 audio_format_t format, 3339 uint32_t channelMask, 3340 int frameCount, 3341 const sp<IMemory>& sharedBuffer, 3342 int sessionId) 3343 : RefBase(), 3344 mThread(thread), 3345 mClient(client), 3346 mCblk(NULL), 3347 // mBuffer 3348 // mBufferEnd 3349 mFrameCount(0), 3350 mState(IDLE), 3351 mFormat(format), 3352 mStepServerFailed(false), 3353 mSessionId(sessionId) 3354 // mChannelCount 3355 // mChannelMask 3356{ 3357 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3358 3359 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3360 size_t size = sizeof(audio_track_cblk_t); 3361 uint8_t channelCount = popcount(channelMask); 3362 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3363 if (sharedBuffer == 0) { 3364 size += bufferSize; 3365 } 3366 3367 if (client != NULL) { 3368 mCblkMemory = client->heap()->allocate(size); 3369 if (mCblkMemory != 0) { 3370 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3371 if (mCblk != NULL) { // construct the shared structure in-place. 3372 new(mCblk) audio_track_cblk_t(); 3373 // clear all buffers 3374 mCblk->frameCount = frameCount; 3375 mCblk->sampleRate = sampleRate; 3376// uncomment the following lines to quickly test 32-bit wraparound 3377// mCblk->user = 0xffff0000; 3378// mCblk->server = 0xffff0000; 3379// mCblk->userBase = 0xffff0000; 3380// mCblk->serverBase = 0xffff0000; 3381 mChannelCount = channelCount; 3382 mChannelMask = channelMask; 3383 if (sharedBuffer == 0) { 3384 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3385 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3386 // Force underrun condition to avoid false underrun callback until first data is 3387 // written to buffer (other flags are cleared) 3388 mCblk->flags = CBLK_UNDERRUN_ON; 3389 } else { 3390 mBuffer = sharedBuffer->pointer(); 3391 } 3392 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3393 } 3394 } else { 3395 ALOGE("not enough memory for AudioTrack size=%u", size); 3396 client->heap()->dump("AudioTrack"); 3397 return; 3398 } 3399 } else { 3400 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3401 // construct the shared structure in-place. 3402 new(mCblk) audio_track_cblk_t(); 3403 // clear all buffers 3404 mCblk->frameCount = frameCount; 3405 mCblk->sampleRate = sampleRate; 3406// uncomment the following lines to quickly test 32-bit wraparound 3407// mCblk->user = 0xffff0000; 3408// mCblk->server = 0xffff0000; 3409// mCblk->userBase = 0xffff0000; 3410// mCblk->serverBase = 0xffff0000; 3411 mChannelCount = channelCount; 3412 mChannelMask = channelMask; 3413 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3414 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3415 // Force underrun condition to avoid false underrun callback until first data is 3416 // written to buffer (other flags are cleared) 3417 mCblk->flags = CBLK_UNDERRUN_ON; 3418 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3419 } 3420} 3421 3422AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3423{ 3424 if (mCblk != NULL) { 3425 if (mClient == 0) { 3426 delete mCblk; 3427 } else { 3428 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3429 } 3430 } 3431 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3432 if (mClient != 0) { 3433 // Client destructor must run with AudioFlinger mutex locked 3434 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3435 // If the client's reference count drops to zero, the associated destructor 3436 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3437 // relying on the automatic clear() at end of scope. 3438 mClient.clear(); 3439 } 3440} 3441 3442// AudioBufferProvider interface 3443// getNextBuffer() = 0; 3444// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3445void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3446{ 3447 buffer->raw = NULL; 3448 mFrameCount = buffer->frameCount; 3449 (void) step(); // ignore return value of step() 3450 buffer->frameCount = 0; 3451} 3452 3453bool AudioFlinger::ThreadBase::TrackBase::step() { 3454 bool result; 3455 audio_track_cblk_t* cblk = this->cblk(); 3456 3457 result = cblk->stepServer(mFrameCount); 3458 if (!result) { 3459 ALOGV("stepServer failed acquiring cblk mutex"); 3460 mStepServerFailed = true; 3461 } 3462 return result; 3463} 3464 3465void AudioFlinger::ThreadBase::TrackBase::reset() { 3466 audio_track_cblk_t* cblk = this->cblk(); 3467 3468 cblk->user = 0; 3469 cblk->server = 0; 3470 cblk->userBase = 0; 3471 cblk->serverBase = 0; 3472 mStepServerFailed = false; 3473 ALOGV("TrackBase::reset"); 3474} 3475 3476int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3477 return (int)mCblk->sampleRate; 3478} 3479 3480void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3481 audio_track_cblk_t* cblk = this->cblk(); 3482 size_t frameSize = cblk->frameSize; 3483 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3484 int8_t *bufferEnd = bufferStart + frames * frameSize; 3485 3486 // Check validity of returned pointer in case the track control block would have been corrupted. 3487 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3488 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3489 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3490 server %u, serverBase %u, user %u, userBase %u", 3491 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3492 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3493 return NULL; 3494 } 3495 3496 return bufferStart; 3497} 3498 3499status_t AudioFlinger::ThreadBase::TrackBase::setSyncEvent(const sp<SyncEvent>& event) 3500{ 3501 mSyncEvents.add(event); 3502 return NO_ERROR; 3503} 3504 3505// ---------------------------------------------------------------------------- 3506 3507// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3508AudioFlinger::PlaybackThread::Track::Track( 3509 PlaybackThread *thread, 3510 const sp<Client>& client, 3511 audio_stream_type_t streamType, 3512 uint32_t sampleRate, 3513 audio_format_t format, 3514 uint32_t channelMask, 3515 int frameCount, 3516 const sp<IMemory>& sharedBuffer, 3517 int sessionId, 3518 IAudioFlinger::track_flags_t flags) 3519 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3520 mMute(false), 3521 // mFillingUpStatus ? 3522 // mRetryCount initialized later when needed 3523 mSharedBuffer(sharedBuffer), 3524 mStreamType(streamType), 3525 mName(-1), // see note below 3526 mMainBuffer(thread->mixBuffer()), 3527 mAuxBuffer(NULL), 3528 mAuxEffectId(0), mHasVolumeController(false), 3529 mPresentationCompleteFrames(0), 3530 mFlags(flags) 3531{ 3532 if (mCblk != NULL) { 3533 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3534 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3535 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3536 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3537 mName = thread->getTrackName_l((audio_channel_mask_t)channelMask); 3538 if (mName < 0) { 3539 ALOGE("no more track names available"); 3540 } 3541 } 3542 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3543} 3544 3545AudioFlinger::PlaybackThread::Track::~Track() 3546{ 3547 ALOGV("PlaybackThread::Track destructor"); 3548 sp<ThreadBase> thread = mThread.promote(); 3549 if (thread != 0) { 3550 Mutex::Autolock _l(thread->mLock); 3551 mState = TERMINATED; 3552 } 3553} 3554 3555void AudioFlinger::PlaybackThread::Track::destroy() 3556{ 3557 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3558 // by removing it from mTracks vector, so there is a risk that this Tracks's 3559 // destructor is called. As the destructor needs to lock mLock, 3560 // we must acquire a strong reference on this Track before locking mLock 3561 // here so that the destructor is called only when exiting this function. 3562 // On the other hand, as long as Track::destroy() is only called by 3563 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3564 // this Track with its member mTrack. 3565 sp<Track> keep(this); 3566 { // scope for mLock 3567 sp<ThreadBase> thread = mThread.promote(); 3568 if (thread != 0) { 3569 if (!isOutputTrack()) { 3570 if (mState == ACTIVE || mState == RESUMING) { 3571 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3572 3573#ifdef ADD_BATTERY_DATA 3574 // to track the speaker usage 3575 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3576#endif 3577 } 3578 AudioSystem::releaseOutput(thread->id()); 3579 } 3580 Mutex::Autolock _l(thread->mLock); 3581 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3582 playbackThread->destroyTrack_l(this); 3583 } 3584 } 3585} 3586 3587void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3588{ 3589 uint32_t vlr = mCblk->getVolumeLR(); 3590 snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 3591 mName - AudioMixer::TRACK0, 3592 (mClient == 0) ? getpid_cached : mClient->pid(), 3593 mStreamType, 3594 mFormat, 3595 mChannelMask, 3596 mSessionId, 3597 mFrameCount, 3598 mState, 3599 mMute, 3600 mFillingUpStatus, 3601 mCblk->sampleRate, 3602 vlr & 0xFFFF, 3603 vlr >> 16, 3604 mCblk->server, 3605 mCblk->user, 3606 (int)mMainBuffer, 3607 (int)mAuxBuffer); 3608} 3609 3610// AudioBufferProvider interface 3611status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3612 AudioBufferProvider::Buffer* buffer, int64_t pts) 3613{ 3614 audio_track_cblk_t* cblk = this->cblk(); 3615 uint32_t framesReady; 3616 uint32_t framesReq = buffer->frameCount; 3617 3618 // Check if last stepServer failed, try to step now 3619 if (mStepServerFailed) { 3620 if (!step()) goto getNextBuffer_exit; 3621 ALOGV("stepServer recovered"); 3622 mStepServerFailed = false; 3623 } 3624 3625 framesReady = cblk->framesReady(); 3626 3627 if (CC_LIKELY(framesReady)) { 3628 uint32_t s = cblk->server; 3629 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3630 3631 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3632 if (framesReq > framesReady) { 3633 framesReq = framesReady; 3634 } 3635 if (framesReq > bufferEnd - s) { 3636 framesReq = bufferEnd - s; 3637 } 3638 3639 buffer->raw = getBuffer(s, framesReq); 3640 if (buffer->raw == NULL) goto getNextBuffer_exit; 3641 3642 buffer->frameCount = framesReq; 3643 return NO_ERROR; 3644 } 3645 3646getNextBuffer_exit: 3647 buffer->raw = NULL; 3648 buffer->frameCount = 0; 3649 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3650 return NOT_ENOUGH_DATA; 3651} 3652 3653uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3654 return mCblk->framesReady(); 3655} 3656 3657bool AudioFlinger::PlaybackThread::Track::isReady() const { 3658 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3659 3660 if (framesReady() >= mCblk->frameCount || 3661 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3662 mFillingUpStatus = FS_FILLED; 3663 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3664 return true; 3665 } 3666 return false; 3667} 3668 3669status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid, 3670 AudioSystem::sync_event_t event, 3671 int triggerSession) 3672{ 3673 status_t status = NO_ERROR; 3674 ALOGV("start(%d), calling pid %d session %d tid %d", 3675 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3676 // check for use case 2 with missing callback 3677 if (isFastTrack() && (mSharedBuffer == 0) && (tid == 0)) { 3678 ALOGW("AUDIO_POLICY_OUTPUT_FLAG_FAST denied"); 3679 mFlags &= ~IAudioFlinger::TRACK_FAST; 3680 // FIXME the track must be invalidated and moved to another thread or 3681 // attached directly to the normal mixer now 3682 } 3683 sp<ThreadBase> thread = mThread.promote(); 3684 if (thread != 0) { 3685 Mutex::Autolock _l(thread->mLock); 3686 track_state state = mState; 3687 // here the track could be either new, or restarted 3688 // in both cases "unstop" the track 3689 if (mState == PAUSED) { 3690 mState = TrackBase::RESUMING; 3691 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3692 } else { 3693 mState = TrackBase::ACTIVE; 3694 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3695 } 3696 3697 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3698 thread->mLock.unlock(); 3699 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3700 thread->mLock.lock(); 3701 3702#ifdef ADD_BATTERY_DATA 3703 // to track the speaker usage 3704 if (status == NO_ERROR) { 3705 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3706 } 3707#endif 3708 } 3709 if (status == NO_ERROR) { 3710 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3711 playbackThread->addTrack_l(this); 3712 } else { 3713 mState = state; 3714 } 3715 } else { 3716 status = BAD_VALUE; 3717 } 3718 return status; 3719} 3720 3721void AudioFlinger::PlaybackThread::Track::stop() 3722{ 3723 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3724 sp<ThreadBase> thread = mThread.promote(); 3725 if (thread != 0) { 3726 Mutex::Autolock _l(thread->mLock); 3727 track_state state = mState; 3728 if (mState > STOPPED) { 3729 mState = STOPPED; 3730 // If the track is not active (PAUSED and buffers full), flush buffers 3731 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3732 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3733 reset(); 3734 } 3735 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3736 } 3737 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3738 thread->mLock.unlock(); 3739 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3740 thread->mLock.lock(); 3741 3742#ifdef ADD_BATTERY_DATA 3743 // to track the speaker usage 3744 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3745#endif 3746 } 3747 } 3748} 3749 3750void AudioFlinger::PlaybackThread::Track::pause() 3751{ 3752 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3753 sp<ThreadBase> thread = mThread.promote(); 3754 if (thread != 0) { 3755 Mutex::Autolock _l(thread->mLock); 3756 if (mState == ACTIVE || mState == RESUMING) { 3757 mState = PAUSING; 3758 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3759 if (!isOutputTrack()) { 3760 thread->mLock.unlock(); 3761 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3762 thread->mLock.lock(); 3763 3764#ifdef ADD_BATTERY_DATA 3765 // to track the speaker usage 3766 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3767#endif 3768 } 3769 } 3770 } 3771} 3772 3773void AudioFlinger::PlaybackThread::Track::flush() 3774{ 3775 ALOGV("flush(%d)", mName); 3776 sp<ThreadBase> thread = mThread.promote(); 3777 if (thread != 0) { 3778 Mutex::Autolock _l(thread->mLock); 3779 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3780 return; 3781 } 3782 // No point remaining in PAUSED state after a flush => go to 3783 // STOPPED state 3784 mState = STOPPED; 3785 3786 // do not reset the track if it is still in the process of being stopped or paused. 3787 // this will be done by prepareTracks_l() when the track is stopped. 3788 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3789 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3790 reset(); 3791 } 3792 } 3793} 3794 3795void AudioFlinger::PlaybackThread::Track::reset() 3796{ 3797 // Do not reset twice to avoid discarding data written just after a flush and before 3798 // the audioflinger thread detects the track is stopped. 3799 if (!mResetDone) { 3800 TrackBase::reset(); 3801 // Force underrun condition to avoid false underrun callback until first data is 3802 // written to buffer 3803 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3804 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3805 mFillingUpStatus = FS_FILLING; 3806 mResetDone = true; 3807 mPresentationCompleteFrames = 0; 3808 } 3809} 3810 3811void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3812{ 3813 mMute = muted; 3814} 3815 3816status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3817{ 3818 status_t status = DEAD_OBJECT; 3819 sp<ThreadBase> thread = mThread.promote(); 3820 if (thread != 0) { 3821 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3822 status = playbackThread->attachAuxEffect(this, EffectId); 3823 } 3824 return status; 3825} 3826 3827void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3828{ 3829 mAuxEffectId = EffectId; 3830 mAuxBuffer = buffer; 3831} 3832 3833bool AudioFlinger::PlaybackThread::Track::presentationComplete(size_t framesWritten, 3834 size_t audioHalFrames) 3835{ 3836 // a track is considered presented when the total number of frames written to audio HAL 3837 // corresponds to the number of frames written when presentationComplete() is called for the 3838 // first time (mPresentationCompleteFrames == 0) plus the buffer filling status at that time. 3839 if (mPresentationCompleteFrames == 0) { 3840 mPresentationCompleteFrames = framesWritten + audioHalFrames; 3841 ALOGV("presentationComplete() reset: mPresentationCompleteFrames %d audioHalFrames %d", 3842 mPresentationCompleteFrames, audioHalFrames); 3843 } 3844 if (framesWritten >= mPresentationCompleteFrames) { 3845 ALOGV("presentationComplete() session %d complete: framesWritten %d", 3846 mSessionId, framesWritten); 3847 triggerEvents(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE); 3848 mPresentationCompleteFrames = 0; 3849 return true; 3850 } 3851 return false; 3852} 3853 3854void AudioFlinger::PlaybackThread::Track::triggerEvents(AudioSystem::sync_event_t type) 3855{ 3856 for (int i = 0; i < (int)mSyncEvents.size(); i++) { 3857 if (mSyncEvents[i]->type() == type) { 3858 mSyncEvents[i]->trigger(); 3859 mSyncEvents.removeAt(i); 3860 i--; 3861 } 3862 } 3863} 3864 3865 3866// timed audio tracks 3867 3868sp<AudioFlinger::PlaybackThread::TimedTrack> 3869AudioFlinger::PlaybackThread::TimedTrack::create( 3870 PlaybackThread *thread, 3871 const sp<Client>& client, 3872 audio_stream_type_t streamType, 3873 uint32_t sampleRate, 3874 audio_format_t format, 3875 uint32_t channelMask, 3876 int frameCount, 3877 const sp<IMemory>& sharedBuffer, 3878 int sessionId) { 3879 if (!client->reserveTimedTrack()) 3880 return NULL; 3881 3882 return new TimedTrack( 3883 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3884 sharedBuffer, sessionId); 3885} 3886 3887AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3888 PlaybackThread *thread, 3889 const sp<Client>& client, 3890 audio_stream_type_t streamType, 3891 uint32_t sampleRate, 3892 audio_format_t format, 3893 uint32_t channelMask, 3894 int frameCount, 3895 const sp<IMemory>& sharedBuffer, 3896 int sessionId) 3897 : Track(thread, client, streamType, sampleRate, format, channelMask, 3898 frameCount, sharedBuffer, sessionId, IAudioFlinger::TRACK_TIMED), 3899 mTimedSilenceBuffer(NULL), 3900 mTimedSilenceBufferSize(0), 3901 mTimedAudioOutputOnTime(false), 3902 mMediaTimeTransformValid(false) 3903{ 3904 LocalClock lc; 3905 mLocalTimeFreq = lc.getLocalFreq(); 3906 3907 mLocalTimeToSampleTransform.a_zero = 0; 3908 mLocalTimeToSampleTransform.b_zero = 0; 3909 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3910 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3911 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3912 &mLocalTimeToSampleTransform.a_to_b_denom); 3913} 3914 3915AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3916 mClient->releaseTimedTrack(); 3917 delete [] mTimedSilenceBuffer; 3918} 3919 3920status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3921 size_t size, sp<IMemory>* buffer) { 3922 3923 Mutex::Autolock _l(mTimedBufferQueueLock); 3924 3925 trimTimedBufferQueue_l(); 3926 3927 // lazily initialize the shared memory heap for timed buffers 3928 if (mTimedMemoryDealer == NULL) { 3929 const int kTimedBufferHeapSize = 512 << 10; 3930 3931 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3932 "AudioFlingerTimed"); 3933 if (mTimedMemoryDealer == NULL) 3934 return NO_MEMORY; 3935 } 3936 3937 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3938 if (newBuffer == NULL) { 3939 newBuffer = mTimedMemoryDealer->allocate(size); 3940 if (newBuffer == NULL) 3941 return NO_MEMORY; 3942 } 3943 3944 *buffer = newBuffer; 3945 return NO_ERROR; 3946} 3947 3948// caller must hold mTimedBufferQueueLock 3949void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3950 int64_t mediaTimeNow; 3951 { 3952 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3953 if (!mMediaTimeTransformValid) 3954 return; 3955 3956 int64_t targetTimeNow; 3957 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3958 ? mCCHelper.getCommonTime(&targetTimeNow) 3959 : mCCHelper.getLocalTime(&targetTimeNow); 3960 3961 if (OK != res) 3962 return; 3963 3964 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3965 &mediaTimeNow)) { 3966 return; 3967 } 3968 } 3969 3970 size_t trimIndex; 3971 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3972 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3973 break; 3974 } 3975 3976 if (trimIndex) { 3977 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3978 } 3979} 3980 3981status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3982 const sp<IMemory>& buffer, int64_t pts) { 3983 3984 { 3985 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3986 if (!mMediaTimeTransformValid) 3987 return INVALID_OPERATION; 3988 } 3989 3990 Mutex::Autolock _l(mTimedBufferQueueLock); 3991 3992 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3993 3994 return NO_ERROR; 3995} 3996 3997status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3998 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3999 4000 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 4001 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 4002 target); 4003 4004 if (!(target == TimedAudioTrack::LOCAL_TIME || 4005 target == TimedAudioTrack::COMMON_TIME)) { 4006 return BAD_VALUE; 4007 } 4008 4009 Mutex::Autolock lock(mMediaTimeTransformLock); 4010 mMediaTimeTransform = xform; 4011 mMediaTimeTransformTarget = target; 4012 mMediaTimeTransformValid = true; 4013 4014 return NO_ERROR; 4015} 4016 4017#define min(a, b) ((a) < (b) ? (a) : (b)) 4018 4019// implementation of getNextBuffer for tracks whose buffers have timestamps 4020status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 4021 AudioBufferProvider::Buffer* buffer, int64_t pts) 4022{ 4023 if (pts == AudioBufferProvider::kInvalidPTS) { 4024 buffer->raw = 0; 4025 buffer->frameCount = 0; 4026 return INVALID_OPERATION; 4027 } 4028 4029 Mutex::Autolock _l(mTimedBufferQueueLock); 4030 4031 while (true) { 4032 4033 // if we have no timed buffers, then fail 4034 if (mTimedBufferQueue.isEmpty()) { 4035 buffer->raw = 0; 4036 buffer->frameCount = 0; 4037 return NOT_ENOUGH_DATA; 4038 } 4039 4040 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4041 4042 // calculate the PTS of the head of the timed buffer queue expressed in 4043 // local time 4044 int64_t headLocalPTS; 4045 { 4046 Mutex::Autolock mttLock(mMediaTimeTransformLock); 4047 4048 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 4049 4050 if (mMediaTimeTransform.a_to_b_denom == 0) { 4051 // the transform represents a pause, so yield silence 4052 timedYieldSilence(buffer->frameCount, buffer); 4053 return NO_ERROR; 4054 } 4055 4056 int64_t transformedPTS; 4057 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 4058 &transformedPTS)) { 4059 // the transform failed. this shouldn't happen, but if it does 4060 // then just drop this buffer 4061 ALOGW("timedGetNextBuffer transform failed"); 4062 buffer->raw = 0; 4063 buffer->frameCount = 0; 4064 mTimedBufferQueue.removeAt(0); 4065 return NO_ERROR; 4066 } 4067 4068 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 4069 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 4070 &headLocalPTS)) { 4071 buffer->raw = 0; 4072 buffer->frameCount = 0; 4073 return INVALID_OPERATION; 4074 } 4075 } else { 4076 headLocalPTS = transformedPTS; 4077 } 4078 } 4079 4080 // adjust the head buffer's PTS to reflect the portion of the head buffer 4081 // that has already been consumed 4082 int64_t effectivePTS = headLocalPTS + 4083 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4084 4085 // Calculate the delta in samples between the head of the input buffer 4086 // queue and the start of the next output buffer that will be written. 4087 // If the transformation fails because of over or underflow, it means 4088 // that the sample's position in the output stream is so far out of 4089 // whack that it should just be dropped. 4090 int64_t sampleDelta; 4091 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4092 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4093 mTimedBufferQueue.removeAt(0); 4094 continue; 4095 } 4096 if (!mLocalTimeToSampleTransform.doForwardTransform( 4097 (effectivePTS - pts) << 32, &sampleDelta)) { 4098 ALOGV("*** too late during sample rate transform: dropped buffer"); 4099 mTimedBufferQueue.removeAt(0); 4100 continue; 4101 } 4102 4103 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4104 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4105 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4106 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4107 4108 // if the delta between the ideal placement for the next input sample and 4109 // the current output position is within this threshold, then we will 4110 // concatenate the next input samples to the previous output 4111 const int64_t kSampleContinuityThreshold = 4112 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4113 4114 // if this is the first buffer of audio that we're emitting from this track 4115 // then it should be almost exactly on time. 4116 const int64_t kSampleStartupThreshold = 1LL << 32; 4117 4118 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4119 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4120 // the next input is close enough to being on time, so concatenate it 4121 // with the last output 4122 timedYieldSamples(buffer); 4123 4124 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4125 return NO_ERROR; 4126 } else if (sampleDelta > 0) { 4127 // the gap between the current output position and the proper start of 4128 // the next input sample is too big, so fill it with silence 4129 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4130 4131 timedYieldSilence(framesUntilNextInput, buffer); 4132 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4133 return NO_ERROR; 4134 } else { 4135 // the next input sample is late 4136 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4137 size_t onTimeSamplePosition = 4138 head.position() + lateFrames * mCblk->frameSize; 4139 4140 if (onTimeSamplePosition > head.buffer()->size()) { 4141 // all the remaining samples in the head are too late, so 4142 // drop it and move on 4143 ALOGV("*** too late: dropped buffer"); 4144 mTimedBufferQueue.removeAt(0); 4145 continue; 4146 } else { 4147 // skip over the late samples 4148 head.setPosition(onTimeSamplePosition); 4149 4150 // yield the available samples 4151 timedYieldSamples(buffer); 4152 4153 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4154 return NO_ERROR; 4155 } 4156 } 4157 } 4158} 4159 4160// Yield samples from the timed buffer queue head up to the given output 4161// buffer's capacity. 4162// 4163// Caller must hold mTimedBufferQueueLock 4164void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4165 AudioBufferProvider::Buffer* buffer) { 4166 4167 const TimedBuffer& head = mTimedBufferQueue[0]; 4168 4169 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4170 head.position()); 4171 4172 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4173 mCblk->frameSize); 4174 size_t framesRequested = buffer->frameCount; 4175 buffer->frameCount = min(framesLeftInHead, framesRequested); 4176 4177 mTimedAudioOutputOnTime = true; 4178} 4179 4180// Yield samples of silence up to the given output buffer's capacity 4181// 4182// Caller must hold mTimedBufferQueueLock 4183void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4184 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4185 4186 // lazily allocate a buffer filled with silence 4187 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4188 delete [] mTimedSilenceBuffer; 4189 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4190 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4191 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4192 } 4193 4194 buffer->raw = mTimedSilenceBuffer; 4195 size_t framesRequested = buffer->frameCount; 4196 buffer->frameCount = min(numFrames, framesRequested); 4197 4198 mTimedAudioOutputOnTime = false; 4199} 4200 4201// AudioBufferProvider interface 4202void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4203 AudioBufferProvider::Buffer* buffer) { 4204 4205 Mutex::Autolock _l(mTimedBufferQueueLock); 4206 4207 // If the buffer which was just released is part of the buffer at the head 4208 // of the queue, be sure to update the amt of the buffer which has been 4209 // consumed. If the buffer being returned is not part of the head of the 4210 // queue, its either because the buffer is part of the silence buffer, or 4211 // because the head of the timed queue was trimmed after the mixer called 4212 // getNextBuffer but before the mixer called releaseBuffer. 4213 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4214 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4215 4216 void* start = head.buffer()->pointer(); 4217 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4218 4219 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4220 head.setPosition(head.position() + 4221 (buffer->frameCount * mCblk->frameSize)); 4222 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4223 mTimedBufferQueue.removeAt(0); 4224 } 4225 } 4226 } 4227 4228 buffer->raw = 0; 4229 buffer->frameCount = 0; 4230} 4231 4232uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4233 Mutex::Autolock _l(mTimedBufferQueueLock); 4234 4235 uint32_t frames = 0; 4236 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4237 const TimedBuffer& tb = mTimedBufferQueue[i]; 4238 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4239 } 4240 4241 return frames; 4242} 4243 4244AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4245 : mPTS(0), mPosition(0) {} 4246 4247AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4248 const sp<IMemory>& buffer, int64_t pts) 4249 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4250 4251// ---------------------------------------------------------------------------- 4252 4253// RecordTrack constructor must be called with AudioFlinger::mLock held 4254AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4255 RecordThread *thread, 4256 const sp<Client>& client, 4257 uint32_t sampleRate, 4258 audio_format_t format, 4259 uint32_t channelMask, 4260 int frameCount, 4261 int sessionId) 4262 : TrackBase(thread, client, sampleRate, format, 4263 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4264 mOverflow(false) 4265{ 4266 if (mCblk != NULL) { 4267 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4268 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4269 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4270 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4271 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4272 } else { 4273 mCblk->frameSize = sizeof(int8_t); 4274 } 4275 } 4276} 4277 4278AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4279{ 4280 sp<ThreadBase> thread = mThread.promote(); 4281 if (thread != 0) { 4282 AudioSystem::releaseInput(thread->id()); 4283 } 4284} 4285 4286// AudioBufferProvider interface 4287status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4288{ 4289 audio_track_cblk_t* cblk = this->cblk(); 4290 uint32_t framesAvail; 4291 uint32_t framesReq = buffer->frameCount; 4292 4293 // Check if last stepServer failed, try to step now 4294 if (mStepServerFailed) { 4295 if (!step()) goto getNextBuffer_exit; 4296 ALOGV("stepServer recovered"); 4297 mStepServerFailed = false; 4298 } 4299 4300 framesAvail = cblk->framesAvailable_l(); 4301 4302 if (CC_LIKELY(framesAvail)) { 4303 uint32_t s = cblk->server; 4304 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4305 4306 if (framesReq > framesAvail) { 4307 framesReq = framesAvail; 4308 } 4309 if (framesReq > bufferEnd - s) { 4310 framesReq = bufferEnd - s; 4311 } 4312 4313 buffer->raw = getBuffer(s, framesReq); 4314 if (buffer->raw == NULL) goto getNextBuffer_exit; 4315 4316 buffer->frameCount = framesReq; 4317 return NO_ERROR; 4318 } 4319 4320getNextBuffer_exit: 4321 buffer->raw = NULL; 4322 buffer->frameCount = 0; 4323 return NOT_ENOUGH_DATA; 4324} 4325 4326status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid, 4327 AudioSystem::sync_event_t event, 4328 int triggerSession) 4329{ 4330 sp<ThreadBase> thread = mThread.promote(); 4331 if (thread != 0) { 4332 RecordThread *recordThread = (RecordThread *)thread.get(); 4333 return recordThread->start(this, tid, event, triggerSession); 4334 } else { 4335 return BAD_VALUE; 4336 } 4337} 4338 4339void AudioFlinger::RecordThread::RecordTrack::stop() 4340{ 4341 sp<ThreadBase> thread = mThread.promote(); 4342 if (thread != 0) { 4343 RecordThread *recordThread = (RecordThread *)thread.get(); 4344 recordThread->stop(this); 4345 TrackBase::reset(); 4346 // Force overrun condition to avoid false overrun callback until first data is 4347 // read from buffer 4348 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4349 } 4350} 4351 4352void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4353{ 4354 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4355 (mClient == 0) ? getpid_cached : mClient->pid(), 4356 mFormat, 4357 mChannelMask, 4358 mSessionId, 4359 mFrameCount, 4360 mState, 4361 mCblk->sampleRate, 4362 mCblk->server, 4363 mCblk->user); 4364} 4365 4366 4367// ---------------------------------------------------------------------------- 4368 4369AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4370 PlaybackThread *playbackThread, 4371 DuplicatingThread *sourceThread, 4372 uint32_t sampleRate, 4373 audio_format_t format, 4374 uint32_t channelMask, 4375 int frameCount) 4376 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, 4377 NULL, 0, IAudioFlinger::TRACK_DEFAULT), 4378 mActive(false), mSourceThread(sourceThread) 4379{ 4380 4381 if (mCblk != NULL) { 4382 mCblk->flags |= CBLK_DIRECTION_OUT; 4383 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4384 mOutBuffer.frameCount = 0; 4385 playbackThread->mTracks.add(this); 4386 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4387 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4388 mCblk, mBuffer, mCblk->buffers, 4389 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4390 } else { 4391 ALOGW("Error creating output track on thread %p", playbackThread); 4392 } 4393} 4394 4395AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4396{ 4397 clearBufferQueue(); 4398} 4399 4400status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid, 4401 AudioSystem::sync_event_t event, 4402 int triggerSession) 4403{ 4404 status_t status = Track::start(tid, event, triggerSession); 4405 if (status != NO_ERROR) { 4406 return status; 4407 } 4408 4409 mActive = true; 4410 mRetryCount = 127; 4411 return status; 4412} 4413 4414void AudioFlinger::PlaybackThread::OutputTrack::stop() 4415{ 4416 Track::stop(); 4417 clearBufferQueue(); 4418 mOutBuffer.frameCount = 0; 4419 mActive = false; 4420} 4421 4422bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4423{ 4424 Buffer *pInBuffer; 4425 Buffer inBuffer; 4426 uint32_t channelCount = mChannelCount; 4427 bool outputBufferFull = false; 4428 inBuffer.frameCount = frames; 4429 inBuffer.i16 = data; 4430 4431 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4432 4433 if (!mActive && frames != 0) { 4434 start(0); 4435 sp<ThreadBase> thread = mThread.promote(); 4436 if (thread != 0) { 4437 MixerThread *mixerThread = (MixerThread *)thread.get(); 4438 if (mCblk->frameCount > frames){ 4439 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4440 uint32_t startFrames = (mCblk->frameCount - frames); 4441 pInBuffer = new Buffer; 4442 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4443 pInBuffer->frameCount = startFrames; 4444 pInBuffer->i16 = pInBuffer->mBuffer; 4445 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4446 mBufferQueue.add(pInBuffer); 4447 } else { 4448 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4449 } 4450 } 4451 } 4452 } 4453 4454 while (waitTimeLeftMs) { 4455 // First write pending buffers, then new data 4456 if (mBufferQueue.size()) { 4457 pInBuffer = mBufferQueue.itemAt(0); 4458 } else { 4459 pInBuffer = &inBuffer; 4460 } 4461 4462 if (pInBuffer->frameCount == 0) { 4463 break; 4464 } 4465 4466 if (mOutBuffer.frameCount == 0) { 4467 mOutBuffer.frameCount = pInBuffer->frameCount; 4468 nsecs_t startTime = systemTime(); 4469 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4470 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4471 outputBufferFull = true; 4472 break; 4473 } 4474 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4475 if (waitTimeLeftMs >= waitTimeMs) { 4476 waitTimeLeftMs -= waitTimeMs; 4477 } else { 4478 waitTimeLeftMs = 0; 4479 } 4480 } 4481 4482 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4483 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4484 mCblk->stepUser(outFrames); 4485 pInBuffer->frameCount -= outFrames; 4486 pInBuffer->i16 += outFrames * channelCount; 4487 mOutBuffer.frameCount -= outFrames; 4488 mOutBuffer.i16 += outFrames * channelCount; 4489 4490 if (pInBuffer->frameCount == 0) { 4491 if (mBufferQueue.size()) { 4492 mBufferQueue.removeAt(0); 4493 delete [] pInBuffer->mBuffer; 4494 delete pInBuffer; 4495 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4496 } else { 4497 break; 4498 } 4499 } 4500 } 4501 4502 // If we could not write all frames, allocate a buffer and queue it for next time. 4503 if (inBuffer.frameCount) { 4504 sp<ThreadBase> thread = mThread.promote(); 4505 if (thread != 0 && !thread->standby()) { 4506 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4507 pInBuffer = new Buffer; 4508 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4509 pInBuffer->frameCount = inBuffer.frameCount; 4510 pInBuffer->i16 = pInBuffer->mBuffer; 4511 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4512 mBufferQueue.add(pInBuffer); 4513 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4514 } else { 4515 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4516 } 4517 } 4518 } 4519 4520 // Calling write() with a 0 length buffer, means that no more data will be written: 4521 // If no more buffers are pending, fill output track buffer to make sure it is started 4522 // by output mixer. 4523 if (frames == 0 && mBufferQueue.size() == 0) { 4524 if (mCblk->user < mCblk->frameCount) { 4525 frames = mCblk->frameCount - mCblk->user; 4526 pInBuffer = new Buffer; 4527 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4528 pInBuffer->frameCount = frames; 4529 pInBuffer->i16 = pInBuffer->mBuffer; 4530 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4531 mBufferQueue.add(pInBuffer); 4532 } else if (mActive) { 4533 stop(); 4534 } 4535 } 4536 4537 return outputBufferFull; 4538} 4539 4540status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4541{ 4542 int active; 4543 status_t result; 4544 audio_track_cblk_t* cblk = mCblk; 4545 uint32_t framesReq = buffer->frameCount; 4546 4547// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4548 buffer->frameCount = 0; 4549 4550 uint32_t framesAvail = cblk->framesAvailable(); 4551 4552 4553 if (framesAvail == 0) { 4554 Mutex::Autolock _l(cblk->lock); 4555 goto start_loop_here; 4556 while (framesAvail == 0) { 4557 active = mActive; 4558 if (CC_UNLIKELY(!active)) { 4559 ALOGV("Not active and NO_MORE_BUFFERS"); 4560 return NO_MORE_BUFFERS; 4561 } 4562 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4563 if (result != NO_ERROR) { 4564 return NO_MORE_BUFFERS; 4565 } 4566 // read the server count again 4567 start_loop_here: 4568 framesAvail = cblk->framesAvailable_l(); 4569 } 4570 } 4571 4572// if (framesAvail < framesReq) { 4573// return NO_MORE_BUFFERS; 4574// } 4575 4576 if (framesReq > framesAvail) { 4577 framesReq = framesAvail; 4578 } 4579 4580 uint32_t u = cblk->user; 4581 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4582 4583 if (framesReq > bufferEnd - u) { 4584 framesReq = bufferEnd - u; 4585 } 4586 4587 buffer->frameCount = framesReq; 4588 buffer->raw = (void *)cblk->buffer(u); 4589 return NO_ERROR; 4590} 4591 4592 4593void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4594{ 4595 size_t size = mBufferQueue.size(); 4596 4597 for (size_t i = 0; i < size; i++) { 4598 Buffer *pBuffer = mBufferQueue.itemAt(i); 4599 delete [] pBuffer->mBuffer; 4600 delete pBuffer; 4601 } 4602 mBufferQueue.clear(); 4603} 4604 4605// ---------------------------------------------------------------------------- 4606 4607AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4608 : RefBase(), 4609 mAudioFlinger(audioFlinger), 4610 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4611 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4612 mPid(pid), 4613 mTimedTrackCount(0) 4614{ 4615 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4616} 4617 4618// Client destructor must be called with AudioFlinger::mLock held 4619AudioFlinger::Client::~Client() 4620{ 4621 mAudioFlinger->removeClient_l(mPid); 4622} 4623 4624sp<MemoryDealer> AudioFlinger::Client::heap() const 4625{ 4626 return mMemoryDealer; 4627} 4628 4629// Reserve one of the limited slots for a timed audio track associated 4630// with this client 4631bool AudioFlinger::Client::reserveTimedTrack() 4632{ 4633 const int kMaxTimedTracksPerClient = 4; 4634 4635 Mutex::Autolock _l(mTimedTrackLock); 4636 4637 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4638 ALOGW("can not create timed track - pid %d has exceeded the limit", 4639 mPid); 4640 return false; 4641 } 4642 4643 mTimedTrackCount++; 4644 return true; 4645} 4646 4647// Release a slot for a timed audio track 4648void AudioFlinger::Client::releaseTimedTrack() 4649{ 4650 Mutex::Autolock _l(mTimedTrackLock); 4651 mTimedTrackCount--; 4652} 4653 4654// ---------------------------------------------------------------------------- 4655 4656AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4657 const sp<IAudioFlingerClient>& client, 4658 pid_t pid) 4659 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4660{ 4661} 4662 4663AudioFlinger::NotificationClient::~NotificationClient() 4664{ 4665} 4666 4667void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4668{ 4669 sp<NotificationClient> keep(this); 4670 mAudioFlinger->removeNotificationClient(mPid); 4671} 4672 4673// ---------------------------------------------------------------------------- 4674 4675AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4676 : BnAudioTrack(), 4677 mTrack(track) 4678{ 4679} 4680 4681AudioFlinger::TrackHandle::~TrackHandle() { 4682 // just stop the track on deletion, associated resources 4683 // will be freed from the main thread once all pending buffers have 4684 // been played. Unless it's not in the active track list, in which 4685 // case we free everything now... 4686 mTrack->destroy(); 4687} 4688 4689sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4690 return mTrack->getCblk(); 4691} 4692 4693status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4694 return mTrack->start(tid); 4695} 4696 4697void AudioFlinger::TrackHandle::stop() { 4698 mTrack->stop(); 4699} 4700 4701void AudioFlinger::TrackHandle::flush() { 4702 mTrack->flush(); 4703} 4704 4705void AudioFlinger::TrackHandle::mute(bool e) { 4706 mTrack->mute(e); 4707} 4708 4709void AudioFlinger::TrackHandle::pause() { 4710 mTrack->pause(); 4711} 4712 4713status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4714{ 4715 return mTrack->attachAuxEffect(EffectId); 4716} 4717 4718status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4719 sp<IMemory>* buffer) { 4720 if (!mTrack->isTimedTrack()) 4721 return INVALID_OPERATION; 4722 4723 PlaybackThread::TimedTrack* tt = 4724 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4725 return tt->allocateTimedBuffer(size, buffer); 4726} 4727 4728status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4729 int64_t pts) { 4730 if (!mTrack->isTimedTrack()) 4731 return INVALID_OPERATION; 4732 4733 PlaybackThread::TimedTrack* tt = 4734 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4735 return tt->queueTimedBuffer(buffer, pts); 4736} 4737 4738status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4739 const LinearTransform& xform, int target) { 4740 4741 if (!mTrack->isTimedTrack()) 4742 return INVALID_OPERATION; 4743 4744 PlaybackThread::TimedTrack* tt = 4745 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4746 return tt->setMediaTimeTransform( 4747 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4748} 4749 4750status_t AudioFlinger::TrackHandle::onTransact( 4751 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4752{ 4753 return BnAudioTrack::onTransact(code, data, reply, flags); 4754} 4755 4756// ---------------------------------------------------------------------------- 4757 4758sp<IAudioRecord> AudioFlinger::openRecord( 4759 pid_t pid, 4760 audio_io_handle_t input, 4761 uint32_t sampleRate, 4762 audio_format_t format, 4763 uint32_t channelMask, 4764 int frameCount, 4765 IAudioFlinger::track_flags_t flags, 4766 int *sessionId, 4767 status_t *status) 4768{ 4769 sp<RecordThread::RecordTrack> recordTrack; 4770 sp<RecordHandle> recordHandle; 4771 sp<Client> client; 4772 status_t lStatus; 4773 RecordThread *thread; 4774 size_t inFrameCount; 4775 int lSessionId; 4776 4777 // check calling permissions 4778 if (!recordingAllowed()) { 4779 lStatus = PERMISSION_DENIED; 4780 goto Exit; 4781 } 4782 4783 // add client to list 4784 { // scope for mLock 4785 Mutex::Autolock _l(mLock); 4786 thread = checkRecordThread_l(input); 4787 if (thread == NULL) { 4788 lStatus = BAD_VALUE; 4789 goto Exit; 4790 } 4791 4792 client = registerPid_l(pid); 4793 4794 // If no audio session id is provided, create one here 4795 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4796 lSessionId = *sessionId; 4797 } else { 4798 lSessionId = nextUniqueId(); 4799 if (sessionId != NULL) { 4800 *sessionId = lSessionId; 4801 } 4802 } 4803 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4804 recordTrack = thread->createRecordTrack_l(client, 4805 sampleRate, 4806 format, 4807 channelMask, 4808 frameCount, 4809 lSessionId, 4810 &lStatus); 4811 } 4812 if (lStatus != NO_ERROR) { 4813 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4814 // destructor is called by the TrackBase destructor with mLock held 4815 client.clear(); 4816 recordTrack.clear(); 4817 goto Exit; 4818 } 4819 4820 // return to handle to client 4821 recordHandle = new RecordHandle(recordTrack); 4822 lStatus = NO_ERROR; 4823 4824Exit: 4825 if (status) { 4826 *status = lStatus; 4827 } 4828 return recordHandle; 4829} 4830 4831// ---------------------------------------------------------------------------- 4832 4833AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4834 : BnAudioRecord(), 4835 mRecordTrack(recordTrack) 4836{ 4837} 4838 4839AudioFlinger::RecordHandle::~RecordHandle() { 4840 stop(); 4841} 4842 4843sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4844 return mRecordTrack->getCblk(); 4845} 4846 4847status_t AudioFlinger::RecordHandle::start(pid_t tid, int event, int triggerSession) { 4848 ALOGV("RecordHandle::start()"); 4849 return mRecordTrack->start(tid, (AudioSystem::sync_event_t)event, triggerSession); 4850} 4851 4852void AudioFlinger::RecordHandle::stop() { 4853 ALOGV("RecordHandle::stop()"); 4854 mRecordTrack->stop(); 4855} 4856 4857status_t AudioFlinger::RecordHandle::onTransact( 4858 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4859{ 4860 return BnAudioRecord::onTransact(code, data, reply, flags); 4861} 4862 4863// ---------------------------------------------------------------------------- 4864 4865AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4866 AudioStreamIn *input, 4867 uint32_t sampleRate, 4868 uint32_t channels, 4869 audio_io_handle_t id, 4870 uint32_t device) : 4871 ThreadBase(audioFlinger, id, device, RECORD), 4872 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4873 // mRsmpInIndex and mInputBytes set by readInputParameters() 4874 mReqChannelCount(popcount(channels)), 4875 mReqSampleRate(sampleRate) 4876 // mBytesRead is only meaningful while active, and so is cleared in start() 4877 // (but might be better to also clear here for dump?) 4878{ 4879 snprintf(mName, kNameLength, "AudioIn_%X", id); 4880 4881 readInputParameters(); 4882} 4883 4884 4885AudioFlinger::RecordThread::~RecordThread() 4886{ 4887 delete[] mRsmpInBuffer; 4888 delete mResampler; 4889 delete[] mRsmpOutBuffer; 4890} 4891 4892void AudioFlinger::RecordThread::onFirstRef() 4893{ 4894 run(mName, PRIORITY_URGENT_AUDIO); 4895} 4896 4897status_t AudioFlinger::RecordThread::readyToRun() 4898{ 4899 status_t status = initCheck(); 4900 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4901 return status; 4902} 4903 4904bool AudioFlinger::RecordThread::threadLoop() 4905{ 4906 AudioBufferProvider::Buffer buffer; 4907 sp<RecordTrack> activeTrack; 4908 Vector< sp<EffectChain> > effectChains; 4909 4910 nsecs_t lastWarning = 0; 4911 4912 acquireWakeLock(); 4913 4914 // start recording 4915 while (!exitPending()) { 4916 4917 processConfigEvents(); 4918 4919 { // scope for mLock 4920 Mutex::Autolock _l(mLock); 4921 checkForNewParameters_l(); 4922 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4923 if (!mStandby) { 4924 mInput->stream->common.standby(&mInput->stream->common); 4925 mStandby = true; 4926 } 4927 4928 if (exitPending()) break; 4929 4930 releaseWakeLock_l(); 4931 ALOGV("RecordThread: loop stopping"); 4932 // go to sleep 4933 mWaitWorkCV.wait(mLock); 4934 ALOGV("RecordThread: loop starting"); 4935 acquireWakeLock_l(); 4936 continue; 4937 } 4938 if (mActiveTrack != 0) { 4939 if (mActiveTrack->mState == TrackBase::PAUSING) { 4940 if (!mStandby) { 4941 mInput->stream->common.standby(&mInput->stream->common); 4942 mStandby = true; 4943 } 4944 mActiveTrack.clear(); 4945 mStartStopCond.broadcast(); 4946 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4947 if (mReqChannelCount != mActiveTrack->channelCount()) { 4948 mActiveTrack.clear(); 4949 mStartStopCond.broadcast(); 4950 } else if (mBytesRead != 0) { 4951 // record start succeeds only if first read from audio input 4952 // succeeds 4953 if (mBytesRead > 0) { 4954 mActiveTrack->mState = TrackBase::ACTIVE; 4955 } else { 4956 mActiveTrack.clear(); 4957 } 4958 mStartStopCond.broadcast(); 4959 } 4960 mStandby = false; 4961 } 4962 } 4963 lockEffectChains_l(effectChains); 4964 } 4965 4966 if (mActiveTrack != 0) { 4967 if (mActiveTrack->mState != TrackBase::ACTIVE && 4968 mActiveTrack->mState != TrackBase::RESUMING) { 4969 unlockEffectChains(effectChains); 4970 usleep(kRecordThreadSleepUs); 4971 continue; 4972 } 4973 for (size_t i = 0; i < effectChains.size(); i ++) { 4974 effectChains[i]->process_l(); 4975 } 4976 4977 buffer.frameCount = mFrameCount; 4978 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4979 size_t framesOut = buffer.frameCount; 4980 if (mResampler == NULL) { 4981 // no resampling 4982 while (framesOut) { 4983 size_t framesIn = mFrameCount - mRsmpInIndex; 4984 if (framesIn) { 4985 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4986 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4987 if (framesIn > framesOut) 4988 framesIn = framesOut; 4989 mRsmpInIndex += framesIn; 4990 framesOut -= framesIn; 4991 if ((int)mChannelCount == mReqChannelCount || 4992 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4993 memcpy(dst, src, framesIn * mFrameSize); 4994 } else { 4995 int16_t *src16 = (int16_t *)src; 4996 int16_t *dst16 = (int16_t *)dst; 4997 if (mChannelCount == 1) { 4998 while (framesIn--) { 4999 *dst16++ = *src16; 5000 *dst16++ = *src16++; 5001 } 5002 } else { 5003 while (framesIn--) { 5004 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 5005 src16 += 2; 5006 } 5007 } 5008 } 5009 } 5010 if (framesOut && mFrameCount == mRsmpInIndex) { 5011 if (framesOut == mFrameCount && 5012 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 5013 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 5014 framesOut = 0; 5015 } else { 5016 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5017 mRsmpInIndex = 0; 5018 } 5019 if (mBytesRead < 0) { 5020 ALOGE("Error reading audio input"); 5021 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5022 // Force input into standby so that it tries to 5023 // recover at next read attempt 5024 mInput->stream->common.standby(&mInput->stream->common); 5025 usleep(kRecordThreadSleepUs); 5026 } 5027 mRsmpInIndex = mFrameCount; 5028 framesOut = 0; 5029 buffer.frameCount = 0; 5030 } 5031 } 5032 } 5033 } else { 5034 // resampling 5035 5036 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 5037 // alter output frame count as if we were expecting stereo samples 5038 if (mChannelCount == 1 && mReqChannelCount == 1) { 5039 framesOut >>= 1; 5040 } 5041 mResampler->resample(mRsmpOutBuffer, framesOut, this); 5042 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 5043 // are 32 bit aligned which should be always true. 5044 if (mChannelCount == 2 && mReqChannelCount == 1) { 5045 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 5046 // the resampler always outputs stereo samples: do post stereo to mono conversion 5047 int16_t *src = (int16_t *)mRsmpOutBuffer; 5048 int16_t *dst = buffer.i16; 5049 while (framesOut--) { 5050 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 5051 src += 2; 5052 } 5053 } else { 5054 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 5055 } 5056 5057 } 5058 if (mFramestoDrop == 0) { 5059 mActiveTrack->releaseBuffer(&buffer); 5060 } else { 5061 if (mFramestoDrop > 0) { 5062 mFramestoDrop -= buffer.frameCount; 5063 if (mFramestoDrop < 0) { 5064 mFramestoDrop = 0; 5065 } 5066 } 5067 } 5068 mActiveTrack->overflow(); 5069 } 5070 // client isn't retrieving buffers fast enough 5071 else { 5072 if (!mActiveTrack->setOverflow()) { 5073 nsecs_t now = systemTime(); 5074 if ((now - lastWarning) > kWarningThrottleNs) { 5075 ALOGW("RecordThread: buffer overflow"); 5076 lastWarning = now; 5077 } 5078 } 5079 // Release the processor for a while before asking for a new buffer. 5080 // This will give the application more chance to read from the buffer and 5081 // clear the overflow. 5082 usleep(kRecordThreadSleepUs); 5083 } 5084 } 5085 // enable changes in effect chain 5086 unlockEffectChains(effectChains); 5087 effectChains.clear(); 5088 } 5089 5090 if (!mStandby) { 5091 mInput->stream->common.standby(&mInput->stream->common); 5092 } 5093 mActiveTrack.clear(); 5094 5095 mStartStopCond.broadcast(); 5096 5097 releaseWakeLock(); 5098 5099 ALOGV("RecordThread %p exiting", this); 5100 return false; 5101} 5102 5103 5104sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5105 const sp<AudioFlinger::Client>& client, 5106 uint32_t sampleRate, 5107 audio_format_t format, 5108 int channelMask, 5109 int frameCount, 5110 int sessionId, 5111 status_t *status) 5112{ 5113 sp<RecordTrack> track; 5114 status_t lStatus; 5115 5116 lStatus = initCheck(); 5117 if (lStatus != NO_ERROR) { 5118 ALOGE("Audio driver not initialized."); 5119 goto Exit; 5120 } 5121 5122 { // scope for mLock 5123 Mutex::Autolock _l(mLock); 5124 5125 track = new RecordTrack(this, client, sampleRate, 5126 format, channelMask, frameCount, sessionId); 5127 5128 if (track->getCblk() == 0) { 5129 lStatus = NO_MEMORY; 5130 goto Exit; 5131 } 5132 5133 mTrack = track.get(); 5134 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5135 bool suspend = audio_is_bluetooth_sco_device( 5136 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5137 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5138 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5139 } 5140 lStatus = NO_ERROR; 5141 5142Exit: 5143 if (status) { 5144 *status = lStatus; 5145 } 5146 return track; 5147} 5148 5149status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, 5150 pid_t tid, AudioSystem::sync_event_t event, 5151 int triggerSession) 5152{ 5153 ALOGV("RecordThread::start tid=%d, event %d, triggerSession %d", tid, event, triggerSession); 5154 sp<ThreadBase> strongMe = this; 5155 status_t status = NO_ERROR; 5156 5157 if (event == AudioSystem::SYNC_EVENT_NONE) { 5158 mSyncStartEvent.clear(); 5159 mFramestoDrop = 0; 5160 } else if (event != AudioSystem::SYNC_EVENT_SAME) { 5161 mSyncStartEvent = mAudioFlinger->createSyncEvent(event, 5162 triggerSession, 5163 recordTrack->sessionId(), 5164 syncStartEventCallback, 5165 this); 5166 mFramestoDrop = -1; 5167 } 5168 5169 { 5170 AutoMutex lock(mLock); 5171 if (mActiveTrack != 0) { 5172 if (recordTrack != mActiveTrack.get()) { 5173 status = -EBUSY; 5174 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5175 mActiveTrack->mState = TrackBase::ACTIVE; 5176 } 5177 return status; 5178 } 5179 5180 recordTrack->mState = TrackBase::IDLE; 5181 mActiveTrack = recordTrack; 5182 mLock.unlock(); 5183 status_t status = AudioSystem::startInput(mId); 5184 mLock.lock(); 5185 if (status != NO_ERROR) { 5186 mActiveTrack.clear(); 5187 clearSyncStartEvent(); 5188 return status; 5189 } 5190 mRsmpInIndex = mFrameCount; 5191 mBytesRead = 0; 5192 if (mResampler != NULL) { 5193 mResampler->reset(); 5194 } 5195 mActiveTrack->mState = TrackBase::RESUMING; 5196 // signal thread to start 5197 ALOGV("Signal record thread"); 5198 mWaitWorkCV.signal(); 5199 // do not wait for mStartStopCond if exiting 5200 if (exitPending()) { 5201 mActiveTrack.clear(); 5202 status = INVALID_OPERATION; 5203 goto startError; 5204 } 5205 mStartStopCond.wait(mLock); 5206 if (mActiveTrack == 0) { 5207 ALOGV("Record failed to start"); 5208 status = BAD_VALUE; 5209 goto startError; 5210 } 5211 ALOGV("Record started OK"); 5212 return status; 5213 } 5214startError: 5215 AudioSystem::stopInput(mId); 5216 clearSyncStartEvent(); 5217 return status; 5218} 5219 5220void AudioFlinger::RecordThread::clearSyncStartEvent() 5221{ 5222 if (mSyncStartEvent != 0) { 5223 mSyncStartEvent->cancel(); 5224 } 5225 mSyncStartEvent.clear(); 5226} 5227 5228void AudioFlinger::RecordThread::syncStartEventCallback(const wp<SyncEvent>& event) 5229{ 5230 sp<SyncEvent> strongEvent = event.promote(); 5231 5232 if (strongEvent != 0) { 5233 RecordThread *me = (RecordThread *)strongEvent->cookie(); 5234 me->handleSyncStartEvent(strongEvent); 5235 } 5236} 5237 5238void AudioFlinger::RecordThread::handleSyncStartEvent(const sp<SyncEvent>& event) 5239{ 5240 ALOGV("handleSyncStartEvent() mActiveTrack %p session %d event->listenerSession() %d", 5241 mActiveTrack.get(), 5242 mActiveTrack.get() ? mActiveTrack->sessionId() : 0, 5243 event->listenerSession()); 5244 5245 if (mActiveTrack != 0 && 5246 event == mSyncStartEvent) { 5247 // TODO: use actual buffer filling status instead of 2 buffers when info is available 5248 // from audio HAL 5249 mFramestoDrop = mFrameCount * 2; 5250 mSyncStartEvent.clear(); 5251 } 5252} 5253 5254void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5255 ALOGV("RecordThread::stop"); 5256 sp<ThreadBase> strongMe = this; 5257 { 5258 AutoMutex lock(mLock); 5259 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5260 mActiveTrack->mState = TrackBase::PAUSING; 5261 // do not wait for mStartStopCond if exiting 5262 if (exitPending()) { 5263 return; 5264 } 5265 mStartStopCond.wait(mLock); 5266 // if we have been restarted, recordTrack == mActiveTrack.get() here 5267 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5268 mLock.unlock(); 5269 AudioSystem::stopInput(mId); 5270 mLock.lock(); 5271 ALOGV("Record stopped OK"); 5272 } 5273 } 5274 } 5275} 5276 5277bool AudioFlinger::RecordThread::isValidSyncEvent(const sp<SyncEvent>& event) 5278{ 5279 return false; 5280} 5281 5282status_t AudioFlinger::RecordThread::setSyncEvent(const sp<SyncEvent>& event) 5283{ 5284 if (!isValidSyncEvent(event)) { 5285 return BAD_VALUE; 5286 } 5287 5288 Mutex::Autolock _l(mLock); 5289 5290 if (mTrack != NULL && event->triggerSession() == mTrack->sessionId()) { 5291 mTrack->setSyncEvent(event); 5292 return NO_ERROR; 5293 } 5294 return NAME_NOT_FOUND; 5295} 5296 5297status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5298{ 5299 const size_t SIZE = 256; 5300 char buffer[SIZE]; 5301 String8 result; 5302 5303 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5304 result.append(buffer); 5305 5306 if (mActiveTrack != 0) { 5307 result.append("Active Track:\n"); 5308 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5309 mActiveTrack->dump(buffer, SIZE); 5310 result.append(buffer); 5311 5312 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5313 result.append(buffer); 5314 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5315 result.append(buffer); 5316 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5317 result.append(buffer); 5318 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5319 result.append(buffer); 5320 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5321 result.append(buffer); 5322 5323 5324 } else { 5325 result.append("No record client\n"); 5326 } 5327 write(fd, result.string(), result.size()); 5328 5329 dumpBase(fd, args); 5330 dumpEffectChains(fd, args); 5331 5332 return NO_ERROR; 5333} 5334 5335// AudioBufferProvider interface 5336status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5337{ 5338 size_t framesReq = buffer->frameCount; 5339 size_t framesReady = mFrameCount - mRsmpInIndex; 5340 int channelCount; 5341 5342 if (framesReady == 0) { 5343 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5344 if (mBytesRead < 0) { 5345 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5346 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5347 // Force input into standby so that it tries to 5348 // recover at next read attempt 5349 mInput->stream->common.standby(&mInput->stream->common); 5350 usleep(kRecordThreadSleepUs); 5351 } 5352 buffer->raw = NULL; 5353 buffer->frameCount = 0; 5354 return NOT_ENOUGH_DATA; 5355 } 5356 mRsmpInIndex = 0; 5357 framesReady = mFrameCount; 5358 } 5359 5360 if (framesReq > framesReady) { 5361 framesReq = framesReady; 5362 } 5363 5364 if (mChannelCount == 1 && mReqChannelCount == 2) { 5365 channelCount = 1; 5366 } else { 5367 channelCount = 2; 5368 } 5369 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5370 buffer->frameCount = framesReq; 5371 return NO_ERROR; 5372} 5373 5374// AudioBufferProvider interface 5375void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5376{ 5377 mRsmpInIndex += buffer->frameCount; 5378 buffer->frameCount = 0; 5379} 5380 5381bool AudioFlinger::RecordThread::checkForNewParameters_l() 5382{ 5383 bool reconfig = false; 5384 5385 while (!mNewParameters.isEmpty()) { 5386 status_t status = NO_ERROR; 5387 String8 keyValuePair = mNewParameters[0]; 5388 AudioParameter param = AudioParameter(keyValuePair); 5389 int value; 5390 audio_format_t reqFormat = mFormat; 5391 int reqSamplingRate = mReqSampleRate; 5392 int reqChannelCount = mReqChannelCount; 5393 5394 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5395 reqSamplingRate = value; 5396 reconfig = true; 5397 } 5398 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5399 reqFormat = (audio_format_t) value; 5400 reconfig = true; 5401 } 5402 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5403 reqChannelCount = popcount(value); 5404 reconfig = true; 5405 } 5406 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5407 // do not accept frame count changes if tracks are open as the track buffer 5408 // size depends on frame count and correct behavior would not be guaranteed 5409 // if frame count is changed after track creation 5410 if (mActiveTrack != 0) { 5411 status = INVALID_OPERATION; 5412 } else { 5413 reconfig = true; 5414 } 5415 } 5416 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5417 // forward device change to effects that have requested to be 5418 // aware of attached audio device. 5419 for (size_t i = 0; i < mEffectChains.size(); i++) { 5420 mEffectChains[i]->setDevice_l(value); 5421 } 5422 // store input device and output device but do not forward output device to audio HAL. 5423 // Note that status is ignored by the caller for output device 5424 // (see AudioFlinger::setParameters() 5425 if (value & AUDIO_DEVICE_OUT_ALL) { 5426 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5427 status = BAD_VALUE; 5428 } else { 5429 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5430 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5431 if (mTrack != NULL) { 5432 bool suspend = audio_is_bluetooth_sco_device( 5433 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5434 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5435 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5436 } 5437 } 5438 mDevice |= (uint32_t)value; 5439 } 5440 if (status == NO_ERROR) { 5441 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5442 if (status == INVALID_OPERATION) { 5443 mInput->stream->common.standby(&mInput->stream->common); 5444 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5445 keyValuePair.string()); 5446 } 5447 if (reconfig) { 5448 if (status == BAD_VALUE && 5449 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5450 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5451 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5452 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5453 (reqChannelCount <= FCC_2)) { 5454 status = NO_ERROR; 5455 } 5456 if (status == NO_ERROR) { 5457 readInputParameters(); 5458 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5459 } 5460 } 5461 } 5462 5463 mNewParameters.removeAt(0); 5464 5465 mParamStatus = status; 5466 mParamCond.signal(); 5467 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5468 // already timed out waiting for the status and will never signal the condition. 5469 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5470 } 5471 return reconfig; 5472} 5473 5474String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5475{ 5476 char *s; 5477 String8 out_s8 = String8(); 5478 5479 Mutex::Autolock _l(mLock); 5480 if (initCheck() != NO_ERROR) { 5481 return out_s8; 5482 } 5483 5484 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5485 out_s8 = String8(s); 5486 free(s); 5487 return out_s8; 5488} 5489 5490void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5491 AudioSystem::OutputDescriptor desc; 5492 void *param2 = NULL; 5493 5494 switch (event) { 5495 case AudioSystem::INPUT_OPENED: 5496 case AudioSystem::INPUT_CONFIG_CHANGED: 5497 desc.channels = mChannelMask; 5498 desc.samplingRate = mSampleRate; 5499 desc.format = mFormat; 5500 desc.frameCount = mFrameCount; 5501 desc.latency = 0; 5502 param2 = &desc; 5503 break; 5504 5505 case AudioSystem::INPUT_CLOSED: 5506 default: 5507 break; 5508 } 5509 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5510} 5511 5512void AudioFlinger::RecordThread::readInputParameters() 5513{ 5514 delete mRsmpInBuffer; 5515 // mRsmpInBuffer is always assigned a new[] below 5516 delete mRsmpOutBuffer; 5517 mRsmpOutBuffer = NULL; 5518 delete mResampler; 5519 mResampler = NULL; 5520 5521 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5522 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5523 mChannelCount = (uint16_t)popcount(mChannelMask); 5524 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5525 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5526 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5527 mFrameCount = mInputBytes / mFrameSize; 5528 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5529 5530 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5531 { 5532 int channelCount; 5533 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5534 // stereo to mono post process as the resampler always outputs stereo. 5535 if (mChannelCount == 1 && mReqChannelCount == 2) { 5536 channelCount = 1; 5537 } else { 5538 channelCount = 2; 5539 } 5540 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5541 mResampler->setSampleRate(mSampleRate); 5542 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5543 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5544 5545 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5546 if (mChannelCount == 1 && mReqChannelCount == 1) { 5547 mFrameCount >>= 1; 5548 } 5549 5550 } 5551 mRsmpInIndex = mFrameCount; 5552} 5553 5554unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5555{ 5556 Mutex::Autolock _l(mLock); 5557 if (initCheck() != NO_ERROR) { 5558 return 0; 5559 } 5560 5561 return mInput->stream->get_input_frames_lost(mInput->stream); 5562} 5563 5564uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5565{ 5566 Mutex::Autolock _l(mLock); 5567 uint32_t result = 0; 5568 if (getEffectChain_l(sessionId) != 0) { 5569 result = EFFECT_SESSION; 5570 } 5571 5572 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5573 result |= TRACK_SESSION; 5574 } 5575 5576 return result; 5577} 5578 5579AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5580{ 5581 Mutex::Autolock _l(mLock); 5582 return mTrack; 5583} 5584 5585AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5586{ 5587 Mutex::Autolock _l(mLock); 5588 return mInput; 5589} 5590 5591AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5592{ 5593 Mutex::Autolock _l(mLock); 5594 AudioStreamIn *input = mInput; 5595 mInput = NULL; 5596 return input; 5597} 5598 5599// this method must always be called either with ThreadBase mLock held or inside the thread loop 5600audio_stream_t* AudioFlinger::RecordThread::stream() const 5601{ 5602 if (mInput == NULL) { 5603 return NULL; 5604 } 5605 return &mInput->stream->common; 5606} 5607 5608 5609// ---------------------------------------------------------------------------- 5610 5611audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 5612{ 5613 if (!settingsAllowed()) { 5614 return 0; 5615 } 5616 Mutex::Autolock _l(mLock); 5617 return loadHwModule_l(name); 5618} 5619 5620// loadHwModule_l() must be called with AudioFlinger::mLock held 5621audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 5622{ 5623 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5624 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 5625 ALOGW("loadHwModule() module %s already loaded", name); 5626 return mAudioHwDevs.keyAt(i); 5627 } 5628 } 5629 5630 const hw_module_t *mod; 5631 audio_hw_device_t *dev; 5632 5633 int rc = load_audio_interface(name, &mod, &dev); 5634 if (rc) { 5635 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 5636 return 0; 5637 } 5638 5639 mHardwareStatus = AUDIO_HW_INIT; 5640 rc = dev->init_check(dev); 5641 mHardwareStatus = AUDIO_HW_IDLE; 5642 if (rc) { 5643 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 5644 return 0; 5645 } 5646 5647 if ((mMasterVolumeSupportLvl != MVS_NONE) && 5648 (NULL != dev->set_master_volume)) { 5649 AutoMutex lock(mHardwareLock); 5650 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5651 dev->set_master_volume(dev, mMasterVolume); 5652 mHardwareStatus = AUDIO_HW_IDLE; 5653 } 5654 5655 audio_module_handle_t handle = nextUniqueId(); 5656 mAudioHwDevs.add(handle, new AudioHwDevice(name, dev)); 5657 5658 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 5659 name, mod->name, mod->id, handle); 5660 5661 return handle; 5662 5663} 5664 5665audio_io_handle_t AudioFlinger::openOutput(audio_module_handle_t module, 5666 audio_devices_t *pDevices, 5667 uint32_t *pSamplingRate, 5668 audio_format_t *pFormat, 5669 audio_channel_mask_t *pChannelMask, 5670 uint32_t *pLatencyMs, 5671 audio_policy_output_flags_t flags) 5672{ 5673 status_t status; 5674 PlaybackThread *thread = NULL; 5675 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5676 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5677 audio_channel_mask_t channelMask = pChannelMask ? *pChannelMask : 0; 5678 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5679 audio_stream_out_t *outStream; 5680 audio_hw_device_t *outHwDev; 5681 5682 ALOGV("openOutput(), module %d Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5683 module, 5684 pDevices ? *pDevices : 0, 5685 samplingRate, 5686 format, 5687 channelMask, 5688 flags); 5689 5690 if (pDevices == NULL || *pDevices == 0) { 5691 return 0; 5692 } 5693 5694 Mutex::Autolock _l(mLock); 5695 5696 outHwDev = findSuitableHwDev_l(module, *pDevices); 5697 if (outHwDev == NULL) 5698 return 0; 5699 5700 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5701 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5702 &channelMask, &samplingRate, &outStream); 5703 mHardwareStatus = AUDIO_HW_IDLE; 5704 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5705 outStream, 5706 samplingRate, 5707 format, 5708 channelMask, 5709 status); 5710 5711 if (outStream != NULL) { 5712 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5713 audio_io_handle_t id = nextUniqueId(); 5714 5715 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5716 (format != AUDIO_FORMAT_PCM_16_BIT) || 5717 (channelMask != AUDIO_CHANNEL_OUT_STEREO)) { 5718 thread = new DirectOutputThread(this, output, id, *pDevices); 5719 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5720 } else { 5721 thread = new MixerThread(this, output, id, *pDevices); 5722 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5723 } 5724 mPlaybackThreads.add(id, thread); 5725 5726 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5727 if (pFormat != NULL) *pFormat = format; 5728 if (pChannelMask != NULL) *pChannelMask = channelMask; 5729 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5730 5731 // notify client processes of the new output creation 5732 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5733 5734 // the first primary output opened designates the primary hw device 5735 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_POLICY_OUTPUT_FLAG_PRIMARY)) { 5736 ALOGI("Using module %d has the primary audio interface", module); 5737 mPrimaryHardwareDev = outHwDev; 5738 5739 AutoMutex lock(mHardwareLock); 5740 mHardwareStatus = AUDIO_HW_SET_MODE; 5741 outHwDev->set_mode(outHwDev, mMode); 5742 5743 // Determine the level of master volume support the primary audio HAL has, 5744 // and set the initial master volume at the same time. 5745 float initialVolume = 1.0; 5746 mMasterVolumeSupportLvl = MVS_NONE; 5747 5748 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 5749 if ((NULL != outHwDev->get_master_volume) && 5750 (NO_ERROR == outHwDev->get_master_volume(outHwDev, &initialVolume))) { 5751 mMasterVolumeSupportLvl = MVS_FULL; 5752 } else { 5753 mMasterVolumeSupportLvl = MVS_SETONLY; 5754 initialVolume = 1.0; 5755 } 5756 5757 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 5758 if ((NULL == outHwDev->set_master_volume) || 5759 (NO_ERROR != outHwDev->set_master_volume(outHwDev, initialVolume))) { 5760 mMasterVolumeSupportLvl = MVS_NONE; 5761 } 5762 // now that we have a primary device, initialize master volume on other devices 5763 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 5764 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 5765 5766 if ((dev != mPrimaryHardwareDev) && 5767 (NULL != dev->set_master_volume)) { 5768 dev->set_master_volume(dev, initialVolume); 5769 } 5770 } 5771 mHardwareStatus = AUDIO_HW_IDLE; 5772 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 5773 ? initialVolume 5774 : 1.0; 5775 mMasterVolume = initialVolume; 5776 } 5777 return id; 5778 } 5779 5780 return 0; 5781} 5782 5783audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5784 audio_io_handle_t output2) 5785{ 5786 Mutex::Autolock _l(mLock); 5787 MixerThread *thread1 = checkMixerThread_l(output1); 5788 MixerThread *thread2 = checkMixerThread_l(output2); 5789 5790 if (thread1 == NULL || thread2 == NULL) { 5791 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5792 return 0; 5793 } 5794 5795 audio_io_handle_t id = nextUniqueId(); 5796 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5797 thread->addOutputTrack(thread2); 5798 mPlaybackThreads.add(id, thread); 5799 // notify client processes of the new output creation 5800 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5801 return id; 5802} 5803 5804status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5805{ 5806 // keep strong reference on the playback thread so that 5807 // it is not destroyed while exit() is executed 5808 sp<PlaybackThread> thread; 5809 { 5810 Mutex::Autolock _l(mLock); 5811 thread = checkPlaybackThread_l(output); 5812 if (thread == NULL) { 5813 return BAD_VALUE; 5814 } 5815 5816 ALOGV("closeOutput() %d", output); 5817 5818 if (thread->type() == ThreadBase::MIXER) { 5819 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5820 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5821 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5822 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5823 } 5824 } 5825 } 5826 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5827 mPlaybackThreads.removeItem(output); 5828 } 5829 thread->exit(); 5830 // The thread entity (active unit of execution) is no longer running here, 5831 // but the ThreadBase container still exists. 5832 5833 if (thread->type() != ThreadBase::DUPLICATING) { 5834 AudioStreamOut *out = thread->clearOutput(); 5835 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5836 // from now on thread->mOutput is NULL 5837 out->hwDev->close_output_stream(out->hwDev, out->stream); 5838 delete out; 5839 } 5840 return NO_ERROR; 5841} 5842 5843status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5844{ 5845 Mutex::Autolock _l(mLock); 5846 PlaybackThread *thread = checkPlaybackThread_l(output); 5847 5848 if (thread == NULL) { 5849 return BAD_VALUE; 5850 } 5851 5852 ALOGV("suspendOutput() %d", output); 5853 thread->suspend(); 5854 5855 return NO_ERROR; 5856} 5857 5858status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5859{ 5860 Mutex::Autolock _l(mLock); 5861 PlaybackThread *thread = checkPlaybackThread_l(output); 5862 5863 if (thread == NULL) { 5864 return BAD_VALUE; 5865 } 5866 5867 ALOGV("restoreOutput() %d", output); 5868 5869 thread->restore(); 5870 5871 return NO_ERROR; 5872} 5873 5874audio_io_handle_t AudioFlinger::openInput(audio_module_handle_t module, 5875 audio_devices_t *pDevices, 5876 uint32_t *pSamplingRate, 5877 audio_format_t *pFormat, 5878 uint32_t *pChannelMask) 5879{ 5880 status_t status; 5881 RecordThread *thread = NULL; 5882 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5883 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5884 audio_channel_mask_t channelMask = pChannelMask ? *pChannelMask : 0; 5885 uint32_t reqSamplingRate = samplingRate; 5886 audio_format_t reqFormat = format; 5887 audio_channel_mask_t reqChannels = channelMask; 5888 audio_stream_in_t *inStream; 5889 audio_hw_device_t *inHwDev; 5890 5891 if (pDevices == NULL || *pDevices == 0) { 5892 return 0; 5893 } 5894 5895 Mutex::Autolock _l(mLock); 5896 5897 inHwDev = findSuitableHwDev_l(module, *pDevices); 5898 if (inHwDev == NULL) 5899 return 0; 5900 5901 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5902 &channelMask, &samplingRate, 5903 (audio_in_acoustics_t)0, 5904 &inStream); 5905 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, status %d", 5906 inStream, 5907 samplingRate, 5908 format, 5909 channelMask, 5910 status); 5911 5912 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5913 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5914 // or stereo to mono conversions on 16 bit PCM inputs. 5915 if (inStream == NULL && status == BAD_VALUE && 5916 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5917 (samplingRate <= 2 * reqSamplingRate) && 5918 (popcount(channelMask) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5919 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5920 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5921 &channelMask, &samplingRate, 5922 (audio_in_acoustics_t)0, 5923 &inStream); 5924 } 5925 5926 if (inStream != NULL) { 5927 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5928 5929 audio_io_handle_t id = nextUniqueId(); 5930 // Start record thread 5931 // RecorThread require both input and output device indication to forward to audio 5932 // pre processing modules 5933 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5934 thread = new RecordThread(this, 5935 input, 5936 reqSamplingRate, 5937 reqChannels, 5938 id, 5939 device); 5940 mRecordThreads.add(id, thread); 5941 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5942 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5943 if (pFormat != NULL) *pFormat = format; 5944 if (pChannelMask != NULL) *pChannelMask = reqChannels; 5945 5946 input->stream->common.standby(&input->stream->common); 5947 5948 // notify client processes of the new input creation 5949 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5950 return id; 5951 } 5952 5953 return 0; 5954} 5955 5956status_t AudioFlinger::closeInput(audio_io_handle_t input) 5957{ 5958 // keep strong reference on the record thread so that 5959 // it is not destroyed while exit() is executed 5960 sp<RecordThread> thread; 5961 { 5962 Mutex::Autolock _l(mLock); 5963 thread = checkRecordThread_l(input); 5964 if (thread == NULL) { 5965 return BAD_VALUE; 5966 } 5967 5968 ALOGV("closeInput() %d", input); 5969 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5970 mRecordThreads.removeItem(input); 5971 } 5972 thread->exit(); 5973 // The thread entity (active unit of execution) is no longer running here, 5974 // but the ThreadBase container still exists. 5975 5976 AudioStreamIn *in = thread->clearInput(); 5977 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5978 // from now on thread->mInput is NULL 5979 in->hwDev->close_input_stream(in->hwDev, in->stream); 5980 delete in; 5981 5982 return NO_ERROR; 5983} 5984 5985status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5986{ 5987 Mutex::Autolock _l(mLock); 5988 MixerThread *dstThread = checkMixerThread_l(output); 5989 if (dstThread == NULL) { 5990 ALOGW("setStreamOutput() bad output id %d", output); 5991 return BAD_VALUE; 5992 } 5993 5994 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5995 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5996 5997 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5998 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5999 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 6000 MixerThread *srcThread = (MixerThread *)thread; 6001 srcThread->invalidateTracks(stream); 6002 } 6003 } 6004 6005 return NO_ERROR; 6006} 6007 6008 6009int AudioFlinger::newAudioSessionId() 6010{ 6011 return nextUniqueId(); 6012} 6013 6014void AudioFlinger::acquireAudioSessionId(int audioSession) 6015{ 6016 Mutex::Autolock _l(mLock); 6017 pid_t caller = IPCThreadState::self()->getCallingPid(); 6018 ALOGV("acquiring %d from %d", audioSession, caller); 6019 size_t num = mAudioSessionRefs.size(); 6020 for (size_t i = 0; i< num; i++) { 6021 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 6022 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6023 ref->mCnt++; 6024 ALOGV(" incremented refcount to %d", ref->mCnt); 6025 return; 6026 } 6027 } 6028 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 6029 ALOGV(" added new entry for %d", audioSession); 6030} 6031 6032void AudioFlinger::releaseAudioSessionId(int audioSession) 6033{ 6034 Mutex::Autolock _l(mLock); 6035 pid_t caller = IPCThreadState::self()->getCallingPid(); 6036 ALOGV("releasing %d from %d", audioSession, caller); 6037 size_t num = mAudioSessionRefs.size(); 6038 for (size_t i = 0; i< num; i++) { 6039 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 6040 if (ref->mSessionid == audioSession && ref->mPid == caller) { 6041 ref->mCnt--; 6042 ALOGV(" decremented refcount to %d", ref->mCnt); 6043 if (ref->mCnt == 0) { 6044 mAudioSessionRefs.removeAt(i); 6045 delete ref; 6046 purgeStaleEffects_l(); 6047 } 6048 return; 6049 } 6050 } 6051 ALOGW("session id %d not found for pid %d", audioSession, caller); 6052} 6053 6054void AudioFlinger::purgeStaleEffects_l() { 6055 6056 ALOGV("purging stale effects"); 6057 6058 Vector< sp<EffectChain> > chains; 6059 6060 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6061 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 6062 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6063 sp<EffectChain> ec = t->mEffectChains[j]; 6064 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 6065 chains.push(ec); 6066 } 6067 } 6068 } 6069 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6070 sp<RecordThread> t = mRecordThreads.valueAt(i); 6071 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 6072 sp<EffectChain> ec = t->mEffectChains[j]; 6073 chains.push(ec); 6074 } 6075 } 6076 6077 for (size_t i = 0; i < chains.size(); i++) { 6078 sp<EffectChain> ec = chains[i]; 6079 int sessionid = ec->sessionId(); 6080 sp<ThreadBase> t = ec->mThread.promote(); 6081 if (t == 0) { 6082 continue; 6083 } 6084 size_t numsessionrefs = mAudioSessionRefs.size(); 6085 bool found = false; 6086 for (size_t k = 0; k < numsessionrefs; k++) { 6087 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 6088 if (ref->mSessionid == sessionid) { 6089 ALOGV(" session %d still exists for %d with %d refs", 6090 sessionid, ref->mPid, ref->mCnt); 6091 found = true; 6092 break; 6093 } 6094 } 6095 if (!found) { 6096 // remove all effects from the chain 6097 while (ec->mEffects.size()) { 6098 sp<EffectModule> effect = ec->mEffects[0]; 6099 effect->unPin(); 6100 Mutex::Autolock _l (t->mLock); 6101 t->removeEffect_l(effect); 6102 for (size_t j = 0; j < effect->mHandles.size(); j++) { 6103 sp<EffectHandle> handle = effect->mHandles[j].promote(); 6104 if (handle != 0) { 6105 handle->mEffect.clear(); 6106 if (handle->mHasControl && handle->mEnabled) { 6107 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 6108 } 6109 } 6110 } 6111 AudioSystem::unregisterEffect(effect->id()); 6112 } 6113 } 6114 } 6115 return; 6116} 6117 6118// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 6119AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 6120{ 6121 return mPlaybackThreads.valueFor(output).get(); 6122} 6123 6124// checkMixerThread_l() must be called with AudioFlinger::mLock held 6125AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 6126{ 6127 PlaybackThread *thread = checkPlaybackThread_l(output); 6128 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 6129} 6130 6131// checkRecordThread_l() must be called with AudioFlinger::mLock held 6132AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 6133{ 6134 return mRecordThreads.valueFor(input).get(); 6135} 6136 6137uint32_t AudioFlinger::nextUniqueId() 6138{ 6139 return android_atomic_inc(&mNextUniqueId); 6140} 6141 6142AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 6143{ 6144 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6145 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 6146 AudioStreamOut *output = thread->getOutput(); 6147 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 6148 return thread; 6149 } 6150 } 6151 return NULL; 6152} 6153 6154uint32_t AudioFlinger::primaryOutputDevice_l() const 6155{ 6156 PlaybackThread *thread = primaryPlaybackThread_l(); 6157 6158 if (thread == NULL) { 6159 return 0; 6160 } 6161 6162 return thread->device(); 6163} 6164 6165sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 6166 int triggerSession, 6167 int listenerSession, 6168 sync_event_callback_t callBack, 6169 void *cookie) 6170{ 6171 Mutex::Autolock _l(mLock); 6172 6173 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 6174 status_t playStatus = NAME_NOT_FOUND; 6175 status_t recStatus = NAME_NOT_FOUND; 6176 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6177 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 6178 if (playStatus == NO_ERROR) { 6179 return event; 6180 } 6181 } 6182 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6183 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 6184 if (recStatus == NO_ERROR) { 6185 return event; 6186 } 6187 } 6188 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 6189 mPendingSyncEvents.add(event); 6190 } else { 6191 ALOGV("createSyncEvent() invalid event %d", event->type()); 6192 event.clear(); 6193 } 6194 return event; 6195} 6196 6197// ---------------------------------------------------------------------------- 6198// Effect management 6199// ---------------------------------------------------------------------------- 6200 6201 6202status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 6203{ 6204 Mutex::Autolock _l(mLock); 6205 return EffectQueryNumberEffects(numEffects); 6206} 6207 6208status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 6209{ 6210 Mutex::Autolock _l(mLock); 6211 return EffectQueryEffect(index, descriptor); 6212} 6213 6214status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 6215 effect_descriptor_t *descriptor) const 6216{ 6217 Mutex::Autolock _l(mLock); 6218 return EffectGetDescriptor(pUuid, descriptor); 6219} 6220 6221 6222sp<IEffect> AudioFlinger::createEffect(pid_t pid, 6223 effect_descriptor_t *pDesc, 6224 const sp<IEffectClient>& effectClient, 6225 int32_t priority, 6226 audio_io_handle_t io, 6227 int sessionId, 6228 status_t *status, 6229 int *id, 6230 int *enabled) 6231{ 6232 status_t lStatus = NO_ERROR; 6233 sp<EffectHandle> handle; 6234 effect_descriptor_t desc; 6235 6236 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 6237 pid, effectClient.get(), priority, sessionId, io); 6238 6239 if (pDesc == NULL) { 6240 lStatus = BAD_VALUE; 6241 goto Exit; 6242 } 6243 6244 // check audio settings permission for global effects 6245 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 6246 lStatus = PERMISSION_DENIED; 6247 goto Exit; 6248 } 6249 6250 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 6251 // that can only be created by audio policy manager (running in same process) 6252 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 6253 lStatus = PERMISSION_DENIED; 6254 goto Exit; 6255 } 6256 6257 if (io == 0) { 6258 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 6259 // output must be specified by AudioPolicyManager when using session 6260 // AUDIO_SESSION_OUTPUT_STAGE 6261 lStatus = BAD_VALUE; 6262 goto Exit; 6263 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 6264 // if the output returned by getOutputForEffect() is removed before we lock the 6265 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 6266 // and we will exit safely 6267 io = AudioSystem::getOutputForEffect(&desc); 6268 } 6269 } 6270 6271 { 6272 Mutex::Autolock _l(mLock); 6273 6274 6275 if (!EffectIsNullUuid(&pDesc->uuid)) { 6276 // if uuid is specified, request effect descriptor 6277 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 6278 if (lStatus < 0) { 6279 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 6280 goto Exit; 6281 } 6282 } else { 6283 // if uuid is not specified, look for an available implementation 6284 // of the required type in effect factory 6285 if (EffectIsNullUuid(&pDesc->type)) { 6286 ALOGW("createEffect() no effect type"); 6287 lStatus = BAD_VALUE; 6288 goto Exit; 6289 } 6290 uint32_t numEffects = 0; 6291 effect_descriptor_t d; 6292 d.flags = 0; // prevent compiler warning 6293 bool found = false; 6294 6295 lStatus = EffectQueryNumberEffects(&numEffects); 6296 if (lStatus < 0) { 6297 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6298 goto Exit; 6299 } 6300 for (uint32_t i = 0; i < numEffects; i++) { 6301 lStatus = EffectQueryEffect(i, &desc); 6302 if (lStatus < 0) { 6303 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6304 continue; 6305 } 6306 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6307 // If matching type found save effect descriptor. If the session is 6308 // 0 and the effect is not auxiliary, continue enumeration in case 6309 // an auxiliary version of this effect type is available 6310 found = true; 6311 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6312 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6313 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6314 break; 6315 } 6316 } 6317 } 6318 if (!found) { 6319 lStatus = BAD_VALUE; 6320 ALOGW("createEffect() effect not found"); 6321 goto Exit; 6322 } 6323 // For same effect type, chose auxiliary version over insert version if 6324 // connect to output mix (Compliance to OpenSL ES) 6325 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6326 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6327 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6328 } 6329 } 6330 6331 // Do not allow auxiliary effects on a session different from 0 (output mix) 6332 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6333 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6334 lStatus = INVALID_OPERATION; 6335 goto Exit; 6336 } 6337 6338 // check recording permission for visualizer 6339 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6340 !recordingAllowed()) { 6341 lStatus = PERMISSION_DENIED; 6342 goto Exit; 6343 } 6344 6345 // return effect descriptor 6346 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6347 6348 // If output is not specified try to find a matching audio session ID in one of the 6349 // output threads. 6350 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6351 // because of code checking output when entering the function. 6352 // Note: io is never 0 when creating an effect on an input 6353 if (io == 0) { 6354 // look for the thread where the specified audio session is present 6355 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6356 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6357 io = mPlaybackThreads.keyAt(i); 6358 break; 6359 } 6360 } 6361 if (io == 0) { 6362 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6363 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6364 io = mRecordThreads.keyAt(i); 6365 break; 6366 } 6367 } 6368 } 6369 // If no output thread contains the requested session ID, default to 6370 // first output. The effect chain will be moved to the correct output 6371 // thread when a track with the same session ID is created 6372 if (io == 0 && mPlaybackThreads.size()) { 6373 io = mPlaybackThreads.keyAt(0); 6374 } 6375 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6376 } 6377 ThreadBase *thread = checkRecordThread_l(io); 6378 if (thread == NULL) { 6379 thread = checkPlaybackThread_l(io); 6380 if (thread == NULL) { 6381 ALOGE("createEffect() unknown output thread"); 6382 lStatus = BAD_VALUE; 6383 goto Exit; 6384 } 6385 } 6386 6387 sp<Client> client = registerPid_l(pid); 6388 6389 // create effect on selected output thread 6390 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6391 &desc, enabled, &lStatus); 6392 if (handle != 0 && id != NULL) { 6393 *id = handle->id(); 6394 } 6395 } 6396 6397Exit: 6398 if (status != NULL) { 6399 *status = lStatus; 6400 } 6401 return handle; 6402} 6403 6404status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6405 audio_io_handle_t dstOutput) 6406{ 6407 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6408 sessionId, srcOutput, dstOutput); 6409 Mutex::Autolock _l(mLock); 6410 if (srcOutput == dstOutput) { 6411 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6412 return NO_ERROR; 6413 } 6414 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6415 if (srcThread == NULL) { 6416 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6417 return BAD_VALUE; 6418 } 6419 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6420 if (dstThread == NULL) { 6421 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6422 return BAD_VALUE; 6423 } 6424 6425 Mutex::Autolock _dl(dstThread->mLock); 6426 Mutex::Autolock _sl(srcThread->mLock); 6427 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6428 6429 return NO_ERROR; 6430} 6431 6432// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6433status_t AudioFlinger::moveEffectChain_l(int sessionId, 6434 AudioFlinger::PlaybackThread *srcThread, 6435 AudioFlinger::PlaybackThread *dstThread, 6436 bool reRegister) 6437{ 6438 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6439 sessionId, srcThread, dstThread); 6440 6441 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6442 if (chain == 0) { 6443 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6444 sessionId, srcThread); 6445 return INVALID_OPERATION; 6446 } 6447 6448 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6449 // so that a new chain is created with correct parameters when first effect is added. This is 6450 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6451 // removed. 6452 srcThread->removeEffectChain_l(chain); 6453 6454 // transfer all effects one by one so that new effect chain is created on new thread with 6455 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6456 audio_io_handle_t dstOutput = dstThread->id(); 6457 sp<EffectChain> dstChain; 6458 uint32_t strategy = 0; // prevent compiler warning 6459 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6460 while (effect != 0) { 6461 srcThread->removeEffect_l(effect); 6462 dstThread->addEffect_l(effect); 6463 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6464 if (effect->state() == EffectModule::ACTIVE || 6465 effect->state() == EffectModule::STOPPING) { 6466 effect->start(); 6467 } 6468 // if the move request is not received from audio policy manager, the effect must be 6469 // re-registered with the new strategy and output 6470 if (dstChain == 0) { 6471 dstChain = effect->chain().promote(); 6472 if (dstChain == 0) { 6473 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6474 srcThread->addEffect_l(effect); 6475 return NO_INIT; 6476 } 6477 strategy = dstChain->strategy(); 6478 } 6479 if (reRegister) { 6480 AudioSystem::unregisterEffect(effect->id()); 6481 AudioSystem::registerEffect(&effect->desc(), 6482 dstOutput, 6483 strategy, 6484 sessionId, 6485 effect->id()); 6486 } 6487 effect = chain->getEffectFromId_l(0); 6488 } 6489 6490 return NO_ERROR; 6491} 6492 6493 6494// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6495sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6496 const sp<AudioFlinger::Client>& client, 6497 const sp<IEffectClient>& effectClient, 6498 int32_t priority, 6499 int sessionId, 6500 effect_descriptor_t *desc, 6501 int *enabled, 6502 status_t *status 6503 ) 6504{ 6505 sp<EffectModule> effect; 6506 sp<EffectHandle> handle; 6507 status_t lStatus; 6508 sp<EffectChain> chain; 6509 bool chainCreated = false; 6510 bool effectCreated = false; 6511 bool effectRegistered = false; 6512 6513 lStatus = initCheck(); 6514 if (lStatus != NO_ERROR) { 6515 ALOGW("createEffect_l() Audio driver not initialized."); 6516 goto Exit; 6517 } 6518 6519 // Do not allow effects with session ID 0 on direct output or duplicating threads 6520 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6521 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6522 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6523 desc->name, sessionId); 6524 lStatus = BAD_VALUE; 6525 goto Exit; 6526 } 6527 // Only Pre processor effects are allowed on input threads and only on input threads 6528 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6529 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6530 desc->name, desc->flags, mType); 6531 lStatus = BAD_VALUE; 6532 goto Exit; 6533 } 6534 6535 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6536 6537 { // scope for mLock 6538 Mutex::Autolock _l(mLock); 6539 6540 // check for existing effect chain with the requested audio session 6541 chain = getEffectChain_l(sessionId); 6542 if (chain == 0) { 6543 // create a new chain for this session 6544 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6545 chain = new EffectChain(this, sessionId); 6546 addEffectChain_l(chain); 6547 chain->setStrategy(getStrategyForSession_l(sessionId)); 6548 chainCreated = true; 6549 } else { 6550 effect = chain->getEffectFromDesc_l(desc); 6551 } 6552 6553 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6554 6555 if (effect == 0) { 6556 int id = mAudioFlinger->nextUniqueId(); 6557 // Check CPU and memory usage 6558 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6559 if (lStatus != NO_ERROR) { 6560 goto Exit; 6561 } 6562 effectRegistered = true; 6563 // create a new effect module if none present in the chain 6564 effect = new EffectModule(this, chain, desc, id, sessionId); 6565 lStatus = effect->status(); 6566 if (lStatus != NO_ERROR) { 6567 goto Exit; 6568 } 6569 lStatus = chain->addEffect_l(effect); 6570 if (lStatus != NO_ERROR) { 6571 goto Exit; 6572 } 6573 effectCreated = true; 6574 6575 effect->setDevice(mDevice); 6576 effect->setMode(mAudioFlinger->getMode()); 6577 } 6578 // create effect handle and connect it to effect module 6579 handle = new EffectHandle(effect, client, effectClient, priority); 6580 lStatus = effect->addHandle(handle); 6581 if (enabled != NULL) { 6582 *enabled = (int)effect->isEnabled(); 6583 } 6584 } 6585 6586Exit: 6587 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6588 Mutex::Autolock _l(mLock); 6589 if (effectCreated) { 6590 chain->removeEffect_l(effect); 6591 } 6592 if (effectRegistered) { 6593 AudioSystem::unregisterEffect(effect->id()); 6594 } 6595 if (chainCreated) { 6596 removeEffectChain_l(chain); 6597 } 6598 handle.clear(); 6599 } 6600 6601 if (status != NULL) { 6602 *status = lStatus; 6603 } 6604 return handle; 6605} 6606 6607sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6608{ 6609 sp<EffectChain> chain = getEffectChain_l(sessionId); 6610 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6611} 6612 6613// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6614// PlaybackThread::mLock held 6615status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6616{ 6617 // check for existing effect chain with the requested audio session 6618 int sessionId = effect->sessionId(); 6619 sp<EffectChain> chain = getEffectChain_l(sessionId); 6620 bool chainCreated = false; 6621 6622 if (chain == 0) { 6623 // create a new chain for this session 6624 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6625 chain = new EffectChain(this, sessionId); 6626 addEffectChain_l(chain); 6627 chain->setStrategy(getStrategyForSession_l(sessionId)); 6628 chainCreated = true; 6629 } 6630 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6631 6632 if (chain->getEffectFromId_l(effect->id()) != 0) { 6633 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6634 this, effect->desc().name, chain.get()); 6635 return BAD_VALUE; 6636 } 6637 6638 status_t status = chain->addEffect_l(effect); 6639 if (status != NO_ERROR) { 6640 if (chainCreated) { 6641 removeEffectChain_l(chain); 6642 } 6643 return status; 6644 } 6645 6646 effect->setDevice(mDevice); 6647 effect->setMode(mAudioFlinger->getMode()); 6648 return NO_ERROR; 6649} 6650 6651void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6652 6653 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6654 effect_descriptor_t desc = effect->desc(); 6655 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6656 detachAuxEffect_l(effect->id()); 6657 } 6658 6659 sp<EffectChain> chain = effect->chain().promote(); 6660 if (chain != 0) { 6661 // remove effect chain if removing last effect 6662 if (chain->removeEffect_l(effect) == 0) { 6663 removeEffectChain_l(chain); 6664 } 6665 } else { 6666 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6667 } 6668} 6669 6670void AudioFlinger::ThreadBase::lockEffectChains_l( 6671 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6672{ 6673 effectChains = mEffectChains; 6674 for (size_t i = 0; i < mEffectChains.size(); i++) { 6675 mEffectChains[i]->lock(); 6676 } 6677} 6678 6679void AudioFlinger::ThreadBase::unlockEffectChains( 6680 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6681{ 6682 for (size_t i = 0; i < effectChains.size(); i++) { 6683 effectChains[i]->unlock(); 6684 } 6685} 6686 6687sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6688{ 6689 Mutex::Autolock _l(mLock); 6690 return getEffectChain_l(sessionId); 6691} 6692 6693sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6694{ 6695 size_t size = mEffectChains.size(); 6696 for (size_t i = 0; i < size; i++) { 6697 if (mEffectChains[i]->sessionId() == sessionId) { 6698 return mEffectChains[i]; 6699 } 6700 } 6701 return 0; 6702} 6703 6704void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6705{ 6706 Mutex::Autolock _l(mLock); 6707 size_t size = mEffectChains.size(); 6708 for (size_t i = 0; i < size; i++) { 6709 mEffectChains[i]->setMode_l(mode); 6710 } 6711} 6712 6713void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6714 const wp<EffectHandle>& handle, 6715 bool unpinIfLast) { 6716 6717 Mutex::Autolock _l(mLock); 6718 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6719 // delete the effect module if removing last handle on it 6720 if (effect->removeHandle(handle) == 0) { 6721 if (!effect->isPinned() || unpinIfLast) { 6722 removeEffect_l(effect); 6723 AudioSystem::unregisterEffect(effect->id()); 6724 } 6725 } 6726} 6727 6728status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6729{ 6730 int session = chain->sessionId(); 6731 int16_t *buffer = mMixBuffer; 6732 bool ownsBuffer = false; 6733 6734 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6735 if (session > 0) { 6736 // Only one effect chain can be present in direct output thread and it uses 6737 // the mix buffer as input 6738 if (mType != DIRECT) { 6739 size_t numSamples = mFrameCount * mChannelCount; 6740 buffer = new int16_t[numSamples]; 6741 memset(buffer, 0, numSamples * sizeof(int16_t)); 6742 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6743 ownsBuffer = true; 6744 } 6745 6746 // Attach all tracks with same session ID to this chain. 6747 for (size_t i = 0; i < mTracks.size(); ++i) { 6748 sp<Track> track = mTracks[i]; 6749 if (session == track->sessionId()) { 6750 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6751 track->setMainBuffer(buffer); 6752 chain->incTrackCnt(); 6753 } 6754 } 6755 6756 // indicate all active tracks in the chain 6757 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6758 sp<Track> track = mActiveTracks[i].promote(); 6759 if (track == 0) continue; 6760 if (session == track->sessionId()) { 6761 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6762 chain->incActiveTrackCnt(); 6763 } 6764 } 6765 } 6766 6767 chain->setInBuffer(buffer, ownsBuffer); 6768 chain->setOutBuffer(mMixBuffer); 6769 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6770 // chains list in order to be processed last as it contains output stage effects 6771 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6772 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6773 // after track specific effects and before output stage 6774 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6775 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6776 // Effect chain for other sessions are inserted at beginning of effect 6777 // chains list to be processed before output mix effects. Relative order between other 6778 // sessions is not important 6779 size_t size = mEffectChains.size(); 6780 size_t i = 0; 6781 for (i = 0; i < size; i++) { 6782 if (mEffectChains[i]->sessionId() < session) break; 6783 } 6784 mEffectChains.insertAt(chain, i); 6785 checkSuspendOnAddEffectChain_l(chain); 6786 6787 return NO_ERROR; 6788} 6789 6790size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6791{ 6792 int session = chain->sessionId(); 6793 6794 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6795 6796 for (size_t i = 0; i < mEffectChains.size(); i++) { 6797 if (chain == mEffectChains[i]) { 6798 mEffectChains.removeAt(i); 6799 // detach all active tracks from the chain 6800 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6801 sp<Track> track = mActiveTracks[i].promote(); 6802 if (track == 0) continue; 6803 if (session == track->sessionId()) { 6804 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6805 chain.get(), session); 6806 chain->decActiveTrackCnt(); 6807 } 6808 } 6809 6810 // detach all tracks with same session ID from this chain 6811 for (size_t i = 0; i < mTracks.size(); ++i) { 6812 sp<Track> track = mTracks[i]; 6813 if (session == track->sessionId()) { 6814 track->setMainBuffer(mMixBuffer); 6815 chain->decTrackCnt(); 6816 } 6817 } 6818 break; 6819 } 6820 } 6821 return mEffectChains.size(); 6822} 6823 6824status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6825 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6826{ 6827 Mutex::Autolock _l(mLock); 6828 return attachAuxEffect_l(track, EffectId); 6829} 6830 6831status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6832 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6833{ 6834 status_t status = NO_ERROR; 6835 6836 if (EffectId == 0) { 6837 track->setAuxBuffer(0, NULL); 6838 } else { 6839 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6840 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6841 if (effect != 0) { 6842 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6843 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6844 } else { 6845 status = INVALID_OPERATION; 6846 } 6847 } else { 6848 status = BAD_VALUE; 6849 } 6850 } 6851 return status; 6852} 6853 6854void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6855{ 6856 for (size_t i = 0; i < mTracks.size(); ++i) { 6857 sp<Track> track = mTracks[i]; 6858 if (track->auxEffectId() == effectId) { 6859 attachAuxEffect_l(track, 0); 6860 } 6861 } 6862} 6863 6864status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6865{ 6866 // only one chain per input thread 6867 if (mEffectChains.size() != 0) { 6868 return INVALID_OPERATION; 6869 } 6870 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6871 6872 chain->setInBuffer(NULL); 6873 chain->setOutBuffer(NULL); 6874 6875 checkSuspendOnAddEffectChain_l(chain); 6876 6877 mEffectChains.add(chain); 6878 6879 return NO_ERROR; 6880} 6881 6882size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6883{ 6884 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6885 ALOGW_IF(mEffectChains.size() != 1, 6886 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6887 chain.get(), mEffectChains.size(), this); 6888 if (mEffectChains.size() == 1) { 6889 mEffectChains.removeAt(0); 6890 } 6891 return 0; 6892} 6893 6894// ---------------------------------------------------------------------------- 6895// EffectModule implementation 6896// ---------------------------------------------------------------------------- 6897 6898#undef LOG_TAG 6899#define LOG_TAG "AudioFlinger::EffectModule" 6900 6901AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6902 const wp<AudioFlinger::EffectChain>& chain, 6903 effect_descriptor_t *desc, 6904 int id, 6905 int sessionId) 6906 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6907 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6908{ 6909 ALOGV("Constructor %p", this); 6910 int lStatus; 6911 if (thread == NULL) { 6912 return; 6913 } 6914 6915 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6916 6917 // create effect engine from effect factory 6918 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6919 6920 if (mStatus != NO_ERROR) { 6921 return; 6922 } 6923 lStatus = init(); 6924 if (lStatus < 0) { 6925 mStatus = lStatus; 6926 goto Error; 6927 } 6928 6929 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6930 mPinned = true; 6931 } 6932 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6933 return; 6934Error: 6935 EffectRelease(mEffectInterface); 6936 mEffectInterface = NULL; 6937 ALOGV("Constructor Error %d", mStatus); 6938} 6939 6940AudioFlinger::EffectModule::~EffectModule() 6941{ 6942 ALOGV("Destructor %p", this); 6943 if (mEffectInterface != NULL) { 6944 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6945 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6946 sp<ThreadBase> thread = mThread.promote(); 6947 if (thread != 0) { 6948 audio_stream_t *stream = thread->stream(); 6949 if (stream != NULL) { 6950 stream->remove_audio_effect(stream, mEffectInterface); 6951 } 6952 } 6953 } 6954 // release effect engine 6955 EffectRelease(mEffectInterface); 6956 } 6957} 6958 6959status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6960{ 6961 status_t status; 6962 6963 Mutex::Autolock _l(mLock); 6964 int priority = handle->priority(); 6965 size_t size = mHandles.size(); 6966 sp<EffectHandle> h; 6967 size_t i; 6968 for (i = 0; i < size; i++) { 6969 h = mHandles[i].promote(); 6970 if (h == 0) continue; 6971 if (h->priority() <= priority) break; 6972 } 6973 // if inserted in first place, move effect control from previous owner to this handle 6974 if (i == 0) { 6975 bool enabled = false; 6976 if (h != 0) { 6977 enabled = h->enabled(); 6978 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6979 } 6980 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6981 status = NO_ERROR; 6982 } else { 6983 status = ALREADY_EXISTS; 6984 } 6985 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6986 mHandles.insertAt(handle, i); 6987 return status; 6988} 6989 6990size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6991{ 6992 Mutex::Autolock _l(mLock); 6993 size_t size = mHandles.size(); 6994 size_t i; 6995 for (i = 0; i < size; i++) { 6996 if (mHandles[i] == handle) break; 6997 } 6998 if (i == size) { 6999 return size; 7000 } 7001 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 7002 7003 bool enabled = false; 7004 EffectHandle *hdl = handle.unsafe_get(); 7005 if (hdl != NULL) { 7006 ALOGV("removeHandle() unsafe_get OK"); 7007 enabled = hdl->enabled(); 7008 } 7009 mHandles.removeAt(i); 7010 size = mHandles.size(); 7011 // if removed from first place, move effect control from this handle to next in line 7012 if (i == 0 && size != 0) { 7013 sp<EffectHandle> h = mHandles[0].promote(); 7014 if (h != 0) { 7015 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 7016 } 7017 } 7018 7019 // Prevent calls to process() and other functions on effect interface from now on. 7020 // The effect engine will be released by the destructor when the last strong reference on 7021 // this object is released which can happen after next process is called. 7022 if (size == 0 && !mPinned) { 7023 mState = DESTROYED; 7024 } 7025 7026 return size; 7027} 7028 7029sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 7030{ 7031 Mutex::Autolock _l(mLock); 7032 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 7033} 7034 7035void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 7036{ 7037 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 7038 // keep a strong reference on this EffectModule to avoid calling the 7039 // destructor before we exit 7040 sp<EffectModule> keep(this); 7041 { 7042 sp<ThreadBase> thread = mThread.promote(); 7043 if (thread != 0) { 7044 thread->disconnectEffect(keep, handle, unpinIfLast); 7045 } 7046 } 7047} 7048 7049void AudioFlinger::EffectModule::updateState() { 7050 Mutex::Autolock _l(mLock); 7051 7052 switch (mState) { 7053 case RESTART: 7054 reset_l(); 7055 // FALL THROUGH 7056 7057 case STARTING: 7058 // clear auxiliary effect input buffer for next accumulation 7059 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7060 memset(mConfig.inputCfg.buffer.raw, 7061 0, 7062 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7063 } 7064 start_l(); 7065 mState = ACTIVE; 7066 break; 7067 case STOPPING: 7068 stop_l(); 7069 mDisableWaitCnt = mMaxDisableWaitCnt; 7070 mState = STOPPED; 7071 break; 7072 case STOPPED: 7073 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 7074 // turn off sequence. 7075 if (--mDisableWaitCnt == 0) { 7076 reset_l(); 7077 mState = IDLE; 7078 } 7079 break; 7080 default: //IDLE , ACTIVE, DESTROYED 7081 break; 7082 } 7083} 7084 7085void AudioFlinger::EffectModule::process() 7086{ 7087 Mutex::Autolock _l(mLock); 7088 7089 if (mState == DESTROYED || mEffectInterface == NULL || 7090 mConfig.inputCfg.buffer.raw == NULL || 7091 mConfig.outputCfg.buffer.raw == NULL) { 7092 return; 7093 } 7094 7095 if (isProcessEnabled()) { 7096 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 7097 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7098 ditherAndClamp(mConfig.inputCfg.buffer.s32, 7099 mConfig.inputCfg.buffer.s32, 7100 mConfig.inputCfg.buffer.frameCount/2); 7101 } 7102 7103 // do the actual processing in the effect engine 7104 int ret = (*mEffectInterface)->process(mEffectInterface, 7105 &mConfig.inputCfg.buffer, 7106 &mConfig.outputCfg.buffer); 7107 7108 // force transition to IDLE state when engine is ready 7109 if (mState == STOPPED && ret == -ENODATA) { 7110 mDisableWaitCnt = 1; 7111 } 7112 7113 // clear auxiliary effect input buffer for next accumulation 7114 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7115 memset(mConfig.inputCfg.buffer.raw, 0, 7116 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 7117 } 7118 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 7119 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7120 // If an insert effect is idle and input buffer is different from output buffer, 7121 // accumulate input onto output 7122 sp<EffectChain> chain = mChain.promote(); 7123 if (chain != 0 && chain->activeTrackCnt() != 0) { 7124 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 7125 int16_t *in = mConfig.inputCfg.buffer.s16; 7126 int16_t *out = mConfig.outputCfg.buffer.s16; 7127 for (size_t i = 0; i < frameCnt; i++) { 7128 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 7129 } 7130 } 7131 } 7132} 7133 7134void AudioFlinger::EffectModule::reset_l() 7135{ 7136 if (mEffectInterface == NULL) { 7137 return; 7138 } 7139 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 7140} 7141 7142status_t AudioFlinger::EffectModule::configure() 7143{ 7144 uint32_t channels; 7145 if (mEffectInterface == NULL) { 7146 return NO_INIT; 7147 } 7148 7149 sp<ThreadBase> thread = mThread.promote(); 7150 if (thread == 0) { 7151 return DEAD_OBJECT; 7152 } 7153 7154 // TODO: handle configuration of effects replacing track process 7155 if (thread->channelCount() == 1) { 7156 channels = AUDIO_CHANNEL_OUT_MONO; 7157 } else { 7158 channels = AUDIO_CHANNEL_OUT_STEREO; 7159 } 7160 7161 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7162 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 7163 } else { 7164 mConfig.inputCfg.channels = channels; 7165 } 7166 mConfig.outputCfg.channels = channels; 7167 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7168 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 7169 mConfig.inputCfg.samplingRate = thread->sampleRate(); 7170 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 7171 mConfig.inputCfg.bufferProvider.cookie = NULL; 7172 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 7173 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 7174 mConfig.outputCfg.bufferProvider.cookie = NULL; 7175 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 7176 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 7177 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 7178 // Insert effect: 7179 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 7180 // always overwrites output buffer: input buffer == output buffer 7181 // - in other sessions: 7182 // last effect in the chain accumulates in output buffer: input buffer != output buffer 7183 // other effect: overwrites output buffer: input buffer == output buffer 7184 // Auxiliary effect: 7185 // accumulates in output buffer: input buffer != output buffer 7186 // Therefore: accumulate <=> input buffer != output buffer 7187 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 7188 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 7189 } else { 7190 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 7191 } 7192 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 7193 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 7194 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 7195 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 7196 7197 ALOGV("configure() %p thread %p buffer %p framecount %d", 7198 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 7199 7200 status_t cmdStatus; 7201 uint32_t size = sizeof(int); 7202 status_t status = (*mEffectInterface)->command(mEffectInterface, 7203 EFFECT_CMD_SET_CONFIG, 7204 sizeof(effect_config_t), 7205 &mConfig, 7206 &size, 7207 &cmdStatus); 7208 if (status == 0) { 7209 status = cmdStatus; 7210 } 7211 7212 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 7213 (1000 * mConfig.outputCfg.buffer.frameCount); 7214 7215 return status; 7216} 7217 7218status_t AudioFlinger::EffectModule::init() 7219{ 7220 Mutex::Autolock _l(mLock); 7221 if (mEffectInterface == NULL) { 7222 return NO_INIT; 7223 } 7224 status_t cmdStatus; 7225 uint32_t size = sizeof(status_t); 7226 status_t status = (*mEffectInterface)->command(mEffectInterface, 7227 EFFECT_CMD_INIT, 7228 0, 7229 NULL, 7230 &size, 7231 &cmdStatus); 7232 if (status == 0) { 7233 status = cmdStatus; 7234 } 7235 return status; 7236} 7237 7238status_t AudioFlinger::EffectModule::start() 7239{ 7240 Mutex::Autolock _l(mLock); 7241 return start_l(); 7242} 7243 7244status_t AudioFlinger::EffectModule::start_l() 7245{ 7246 if (mEffectInterface == NULL) { 7247 return NO_INIT; 7248 } 7249 status_t cmdStatus; 7250 uint32_t size = sizeof(status_t); 7251 status_t status = (*mEffectInterface)->command(mEffectInterface, 7252 EFFECT_CMD_ENABLE, 7253 0, 7254 NULL, 7255 &size, 7256 &cmdStatus); 7257 if (status == 0) { 7258 status = cmdStatus; 7259 } 7260 if (status == 0 && 7261 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7262 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7263 sp<ThreadBase> thread = mThread.promote(); 7264 if (thread != 0) { 7265 audio_stream_t *stream = thread->stream(); 7266 if (stream != NULL) { 7267 stream->add_audio_effect(stream, mEffectInterface); 7268 } 7269 } 7270 } 7271 return status; 7272} 7273 7274status_t AudioFlinger::EffectModule::stop() 7275{ 7276 Mutex::Autolock _l(mLock); 7277 return stop_l(); 7278} 7279 7280status_t AudioFlinger::EffectModule::stop_l() 7281{ 7282 if (mEffectInterface == NULL) { 7283 return NO_INIT; 7284 } 7285 status_t cmdStatus; 7286 uint32_t size = sizeof(status_t); 7287 status_t status = (*mEffectInterface)->command(mEffectInterface, 7288 EFFECT_CMD_DISABLE, 7289 0, 7290 NULL, 7291 &size, 7292 &cmdStatus); 7293 if (status == 0) { 7294 status = cmdStatus; 7295 } 7296 if (status == 0 && 7297 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7298 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7299 sp<ThreadBase> thread = mThread.promote(); 7300 if (thread != 0) { 7301 audio_stream_t *stream = thread->stream(); 7302 if (stream != NULL) { 7303 stream->remove_audio_effect(stream, mEffectInterface); 7304 } 7305 } 7306 } 7307 return status; 7308} 7309 7310status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7311 uint32_t cmdSize, 7312 void *pCmdData, 7313 uint32_t *replySize, 7314 void *pReplyData) 7315{ 7316 Mutex::Autolock _l(mLock); 7317// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7318 7319 if (mState == DESTROYED || mEffectInterface == NULL) { 7320 return NO_INIT; 7321 } 7322 status_t status = (*mEffectInterface)->command(mEffectInterface, 7323 cmdCode, 7324 cmdSize, 7325 pCmdData, 7326 replySize, 7327 pReplyData); 7328 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7329 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7330 for (size_t i = 1; i < mHandles.size(); i++) { 7331 sp<EffectHandle> h = mHandles[i].promote(); 7332 if (h != 0) { 7333 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7334 } 7335 } 7336 } 7337 return status; 7338} 7339 7340status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7341{ 7342 7343 Mutex::Autolock _l(mLock); 7344 ALOGV("setEnabled %p enabled %d", this, enabled); 7345 7346 if (enabled != isEnabled()) { 7347 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7348 if (enabled && status != NO_ERROR) { 7349 return status; 7350 } 7351 7352 switch (mState) { 7353 // going from disabled to enabled 7354 case IDLE: 7355 mState = STARTING; 7356 break; 7357 case STOPPED: 7358 mState = RESTART; 7359 break; 7360 case STOPPING: 7361 mState = ACTIVE; 7362 break; 7363 7364 // going from enabled to disabled 7365 case RESTART: 7366 mState = STOPPED; 7367 break; 7368 case STARTING: 7369 mState = IDLE; 7370 break; 7371 case ACTIVE: 7372 mState = STOPPING; 7373 break; 7374 case DESTROYED: 7375 return NO_ERROR; // simply ignore as we are being destroyed 7376 } 7377 for (size_t i = 1; i < mHandles.size(); i++) { 7378 sp<EffectHandle> h = mHandles[i].promote(); 7379 if (h != 0) { 7380 h->setEnabled(enabled); 7381 } 7382 } 7383 } 7384 return NO_ERROR; 7385} 7386 7387bool AudioFlinger::EffectModule::isEnabled() const 7388{ 7389 switch (mState) { 7390 case RESTART: 7391 case STARTING: 7392 case ACTIVE: 7393 return true; 7394 case IDLE: 7395 case STOPPING: 7396 case STOPPED: 7397 case DESTROYED: 7398 default: 7399 return false; 7400 } 7401} 7402 7403bool AudioFlinger::EffectModule::isProcessEnabled() const 7404{ 7405 switch (mState) { 7406 case RESTART: 7407 case ACTIVE: 7408 case STOPPING: 7409 case STOPPED: 7410 return true; 7411 case IDLE: 7412 case STARTING: 7413 case DESTROYED: 7414 default: 7415 return false; 7416 } 7417} 7418 7419status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7420{ 7421 Mutex::Autolock _l(mLock); 7422 status_t status = NO_ERROR; 7423 7424 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7425 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7426 if (isProcessEnabled() && 7427 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7428 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7429 status_t cmdStatus; 7430 uint32_t volume[2]; 7431 uint32_t *pVolume = NULL; 7432 uint32_t size = sizeof(volume); 7433 volume[0] = *left; 7434 volume[1] = *right; 7435 if (controller) { 7436 pVolume = volume; 7437 } 7438 status = (*mEffectInterface)->command(mEffectInterface, 7439 EFFECT_CMD_SET_VOLUME, 7440 size, 7441 volume, 7442 &size, 7443 pVolume); 7444 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7445 *left = volume[0]; 7446 *right = volume[1]; 7447 } 7448 } 7449 return status; 7450} 7451 7452status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7453{ 7454 Mutex::Autolock _l(mLock); 7455 status_t status = NO_ERROR; 7456 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7457 // audio pre processing modules on RecordThread can receive both output and 7458 // input device indication in the same call 7459 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7460 if (dev) { 7461 status_t cmdStatus; 7462 uint32_t size = sizeof(status_t); 7463 7464 status = (*mEffectInterface)->command(mEffectInterface, 7465 EFFECT_CMD_SET_DEVICE, 7466 sizeof(uint32_t), 7467 &dev, 7468 &size, 7469 &cmdStatus); 7470 if (status == NO_ERROR) { 7471 status = cmdStatus; 7472 } 7473 } 7474 dev = device & AUDIO_DEVICE_IN_ALL; 7475 if (dev) { 7476 status_t cmdStatus; 7477 uint32_t size = sizeof(status_t); 7478 7479 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7480 EFFECT_CMD_SET_INPUT_DEVICE, 7481 sizeof(uint32_t), 7482 &dev, 7483 &size, 7484 &cmdStatus); 7485 if (status2 == NO_ERROR) { 7486 status2 = cmdStatus; 7487 } 7488 if (status == NO_ERROR) { 7489 status = status2; 7490 } 7491 } 7492 } 7493 return status; 7494} 7495 7496status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7497{ 7498 Mutex::Autolock _l(mLock); 7499 status_t status = NO_ERROR; 7500 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7501 status_t cmdStatus; 7502 uint32_t size = sizeof(status_t); 7503 status = (*mEffectInterface)->command(mEffectInterface, 7504 EFFECT_CMD_SET_AUDIO_MODE, 7505 sizeof(audio_mode_t), 7506 &mode, 7507 &size, 7508 &cmdStatus); 7509 if (status == NO_ERROR) { 7510 status = cmdStatus; 7511 } 7512 } 7513 return status; 7514} 7515 7516void AudioFlinger::EffectModule::setSuspended(bool suspended) 7517{ 7518 Mutex::Autolock _l(mLock); 7519 mSuspended = suspended; 7520} 7521 7522bool AudioFlinger::EffectModule::suspended() const 7523{ 7524 Mutex::Autolock _l(mLock); 7525 return mSuspended; 7526} 7527 7528status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7529{ 7530 const size_t SIZE = 256; 7531 char buffer[SIZE]; 7532 String8 result; 7533 7534 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7535 result.append(buffer); 7536 7537 bool locked = tryLock(mLock); 7538 // failed to lock - AudioFlinger is probably deadlocked 7539 if (!locked) { 7540 result.append("\t\tCould not lock Fx mutex:\n"); 7541 } 7542 7543 result.append("\t\tSession Status State Engine:\n"); 7544 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7545 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7546 result.append(buffer); 7547 7548 result.append("\t\tDescriptor:\n"); 7549 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7550 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7551 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7552 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7553 result.append(buffer); 7554 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7555 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7556 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7557 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7558 result.append(buffer); 7559 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7560 mDescriptor.apiVersion, 7561 mDescriptor.flags); 7562 result.append(buffer); 7563 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7564 mDescriptor.name); 7565 result.append(buffer); 7566 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7567 mDescriptor.implementor); 7568 result.append(buffer); 7569 7570 result.append("\t\t- Input configuration:\n"); 7571 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7572 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7573 (uint32_t)mConfig.inputCfg.buffer.raw, 7574 mConfig.inputCfg.buffer.frameCount, 7575 mConfig.inputCfg.samplingRate, 7576 mConfig.inputCfg.channels, 7577 mConfig.inputCfg.format); 7578 result.append(buffer); 7579 7580 result.append("\t\t- Output configuration:\n"); 7581 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7582 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7583 (uint32_t)mConfig.outputCfg.buffer.raw, 7584 mConfig.outputCfg.buffer.frameCount, 7585 mConfig.outputCfg.samplingRate, 7586 mConfig.outputCfg.channels, 7587 mConfig.outputCfg.format); 7588 result.append(buffer); 7589 7590 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7591 result.append(buffer); 7592 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7593 for (size_t i = 0; i < mHandles.size(); ++i) { 7594 sp<EffectHandle> handle = mHandles[i].promote(); 7595 if (handle != 0) { 7596 handle->dump(buffer, SIZE); 7597 result.append(buffer); 7598 } 7599 } 7600 7601 result.append("\n"); 7602 7603 write(fd, result.string(), result.length()); 7604 7605 if (locked) { 7606 mLock.unlock(); 7607 } 7608 7609 return NO_ERROR; 7610} 7611 7612// ---------------------------------------------------------------------------- 7613// EffectHandle implementation 7614// ---------------------------------------------------------------------------- 7615 7616#undef LOG_TAG 7617#define LOG_TAG "AudioFlinger::EffectHandle" 7618 7619AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7620 const sp<AudioFlinger::Client>& client, 7621 const sp<IEffectClient>& effectClient, 7622 int32_t priority) 7623 : BnEffect(), 7624 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7625 mPriority(priority), mHasControl(false), mEnabled(false) 7626{ 7627 ALOGV("constructor %p", this); 7628 7629 if (client == 0) { 7630 return; 7631 } 7632 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7633 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7634 if (mCblkMemory != 0) { 7635 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7636 7637 if (mCblk != NULL) { 7638 new(mCblk) effect_param_cblk_t(); 7639 mBuffer = (uint8_t *)mCblk + bufOffset; 7640 } 7641 } else { 7642 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7643 return; 7644 } 7645} 7646 7647AudioFlinger::EffectHandle::~EffectHandle() 7648{ 7649 ALOGV("Destructor %p", this); 7650 disconnect(false); 7651 ALOGV("Destructor DONE %p", this); 7652} 7653 7654status_t AudioFlinger::EffectHandle::enable() 7655{ 7656 ALOGV("enable %p", this); 7657 if (!mHasControl) return INVALID_OPERATION; 7658 if (mEffect == 0) return DEAD_OBJECT; 7659 7660 if (mEnabled) { 7661 return NO_ERROR; 7662 } 7663 7664 mEnabled = true; 7665 7666 sp<ThreadBase> thread = mEffect->thread().promote(); 7667 if (thread != 0) { 7668 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7669 } 7670 7671 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7672 if (mEffect->suspended()) { 7673 return NO_ERROR; 7674 } 7675 7676 status_t status = mEffect->setEnabled(true); 7677 if (status != NO_ERROR) { 7678 if (thread != 0) { 7679 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7680 } 7681 mEnabled = false; 7682 } 7683 return status; 7684} 7685 7686status_t AudioFlinger::EffectHandle::disable() 7687{ 7688 ALOGV("disable %p", this); 7689 if (!mHasControl) return INVALID_OPERATION; 7690 if (mEffect == 0) return DEAD_OBJECT; 7691 7692 if (!mEnabled) { 7693 return NO_ERROR; 7694 } 7695 mEnabled = false; 7696 7697 if (mEffect->suspended()) { 7698 return NO_ERROR; 7699 } 7700 7701 status_t status = mEffect->setEnabled(false); 7702 7703 sp<ThreadBase> thread = mEffect->thread().promote(); 7704 if (thread != 0) { 7705 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7706 } 7707 7708 return status; 7709} 7710 7711void AudioFlinger::EffectHandle::disconnect() 7712{ 7713 disconnect(true); 7714} 7715 7716void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7717{ 7718 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7719 if (mEffect == 0) { 7720 return; 7721 } 7722 mEffect->disconnect(this, unpinIfLast); 7723 7724 if (mHasControl && mEnabled) { 7725 sp<ThreadBase> thread = mEffect->thread().promote(); 7726 if (thread != 0) { 7727 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7728 } 7729 } 7730 7731 // release sp on module => module destructor can be called now 7732 mEffect.clear(); 7733 if (mClient != 0) { 7734 if (mCblk != NULL) { 7735 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7736 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7737 } 7738 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7739 // Client destructor must run with AudioFlinger mutex locked 7740 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7741 mClient.clear(); 7742 } 7743} 7744 7745status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7746 uint32_t cmdSize, 7747 void *pCmdData, 7748 uint32_t *replySize, 7749 void *pReplyData) 7750{ 7751// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7752// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7753 7754 // only get parameter command is permitted for applications not controlling the effect 7755 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7756 return INVALID_OPERATION; 7757 } 7758 if (mEffect == 0) return DEAD_OBJECT; 7759 if (mClient == 0) return INVALID_OPERATION; 7760 7761 // handle commands that are not forwarded transparently to effect engine 7762 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7763 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7764 // no risk to block the whole media server process or mixer threads is we are stuck here 7765 Mutex::Autolock _l(mCblk->lock); 7766 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7767 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7768 mCblk->serverIndex = 0; 7769 mCblk->clientIndex = 0; 7770 return BAD_VALUE; 7771 } 7772 status_t status = NO_ERROR; 7773 while (mCblk->serverIndex < mCblk->clientIndex) { 7774 int reply; 7775 uint32_t rsize = sizeof(int); 7776 int *p = (int *)(mBuffer + mCblk->serverIndex); 7777 int size = *p++; 7778 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7779 ALOGW("command(): invalid parameter block size"); 7780 break; 7781 } 7782 effect_param_t *param = (effect_param_t *)p; 7783 if (param->psize == 0 || param->vsize == 0) { 7784 ALOGW("command(): null parameter or value size"); 7785 mCblk->serverIndex += size; 7786 continue; 7787 } 7788 uint32_t psize = sizeof(effect_param_t) + 7789 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7790 param->vsize; 7791 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7792 psize, 7793 p, 7794 &rsize, 7795 &reply); 7796 // stop at first error encountered 7797 if (ret != NO_ERROR) { 7798 status = ret; 7799 *(int *)pReplyData = reply; 7800 break; 7801 } else if (reply != NO_ERROR) { 7802 *(int *)pReplyData = reply; 7803 break; 7804 } 7805 mCblk->serverIndex += size; 7806 } 7807 mCblk->serverIndex = 0; 7808 mCblk->clientIndex = 0; 7809 return status; 7810 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7811 *(int *)pReplyData = NO_ERROR; 7812 return enable(); 7813 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7814 *(int *)pReplyData = NO_ERROR; 7815 return disable(); 7816 } 7817 7818 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7819} 7820 7821void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7822{ 7823 ALOGV("setControl %p control %d", this, hasControl); 7824 7825 mHasControl = hasControl; 7826 mEnabled = enabled; 7827 7828 if (signal && mEffectClient != 0) { 7829 mEffectClient->controlStatusChanged(hasControl); 7830 } 7831} 7832 7833void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7834 uint32_t cmdSize, 7835 void *pCmdData, 7836 uint32_t replySize, 7837 void *pReplyData) 7838{ 7839 if (mEffectClient != 0) { 7840 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7841 } 7842} 7843 7844 7845 7846void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7847{ 7848 if (mEffectClient != 0) { 7849 mEffectClient->enableStatusChanged(enabled); 7850 } 7851} 7852 7853status_t AudioFlinger::EffectHandle::onTransact( 7854 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7855{ 7856 return BnEffect::onTransact(code, data, reply, flags); 7857} 7858 7859 7860void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7861{ 7862 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7863 7864 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7865 (mClient == 0) ? getpid_cached : mClient->pid(), 7866 mPriority, 7867 mHasControl, 7868 !locked, 7869 mCblk ? mCblk->clientIndex : 0, 7870 mCblk ? mCblk->serverIndex : 0 7871 ); 7872 7873 if (locked) { 7874 mCblk->lock.unlock(); 7875 } 7876} 7877 7878#undef LOG_TAG 7879#define LOG_TAG "AudioFlinger::EffectChain" 7880 7881AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7882 int sessionId) 7883 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7884 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7885 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7886{ 7887 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7888 if (thread == NULL) { 7889 return; 7890 } 7891 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7892 thread->frameCount(); 7893} 7894 7895AudioFlinger::EffectChain::~EffectChain() 7896{ 7897 if (mOwnInBuffer) { 7898 delete mInBuffer; 7899 } 7900 7901} 7902 7903// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7904sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7905{ 7906 size_t size = mEffects.size(); 7907 7908 for (size_t i = 0; i < size; i++) { 7909 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7910 return mEffects[i]; 7911 } 7912 } 7913 return 0; 7914} 7915 7916// getEffectFromId_l() must be called with ThreadBase::mLock held 7917sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7918{ 7919 size_t size = mEffects.size(); 7920 7921 for (size_t i = 0; i < size; i++) { 7922 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7923 if (id == 0 || mEffects[i]->id() == id) { 7924 return mEffects[i]; 7925 } 7926 } 7927 return 0; 7928} 7929 7930// getEffectFromType_l() must be called with ThreadBase::mLock held 7931sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7932 const effect_uuid_t *type) 7933{ 7934 size_t size = mEffects.size(); 7935 7936 for (size_t i = 0; i < size; i++) { 7937 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7938 return mEffects[i]; 7939 } 7940 } 7941 return 0; 7942} 7943 7944// Must be called with EffectChain::mLock locked 7945void AudioFlinger::EffectChain::process_l() 7946{ 7947 sp<ThreadBase> thread = mThread.promote(); 7948 if (thread == 0) { 7949 ALOGW("process_l(): cannot promote mixer thread"); 7950 return; 7951 } 7952 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7953 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7954 // always process effects unless no more tracks are on the session and the effect tail 7955 // has been rendered 7956 bool doProcess = true; 7957 if (!isGlobalSession) { 7958 bool tracksOnSession = (trackCnt() != 0); 7959 7960 if (!tracksOnSession && mTailBufferCount == 0) { 7961 doProcess = false; 7962 } 7963 7964 if (activeTrackCnt() == 0) { 7965 // if no track is active and the effect tail has not been rendered, 7966 // the input buffer must be cleared here as the mixer process will not do it 7967 if (tracksOnSession || mTailBufferCount > 0) { 7968 size_t numSamples = thread->frameCount() * thread->channelCount(); 7969 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7970 if (mTailBufferCount > 0) { 7971 mTailBufferCount--; 7972 } 7973 } 7974 } 7975 } 7976 7977 size_t size = mEffects.size(); 7978 if (doProcess) { 7979 for (size_t i = 0; i < size; i++) { 7980 mEffects[i]->process(); 7981 } 7982 } 7983 for (size_t i = 0; i < size; i++) { 7984 mEffects[i]->updateState(); 7985 } 7986} 7987 7988// addEffect_l() must be called with PlaybackThread::mLock held 7989status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7990{ 7991 effect_descriptor_t desc = effect->desc(); 7992 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7993 7994 Mutex::Autolock _l(mLock); 7995 effect->setChain(this); 7996 sp<ThreadBase> thread = mThread.promote(); 7997 if (thread == 0) { 7998 return NO_INIT; 7999 } 8000 effect->setThread(thread); 8001 8002 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 8003 // Auxiliary effects are inserted at the beginning of mEffects vector as 8004 // they are processed first and accumulated in chain input buffer 8005 mEffects.insertAt(effect, 0); 8006 8007 // the input buffer for auxiliary effect contains mono samples in 8008 // 32 bit format. This is to avoid saturation in AudoMixer 8009 // accumulation stage. Saturation is done in EffectModule::process() before 8010 // calling the process in effect engine 8011 size_t numSamples = thread->frameCount(); 8012 int32_t *buffer = new int32_t[numSamples]; 8013 memset(buffer, 0, numSamples * sizeof(int32_t)); 8014 effect->setInBuffer((int16_t *)buffer); 8015 // auxiliary effects output samples to chain input buffer for further processing 8016 // by insert effects 8017 effect->setOutBuffer(mInBuffer); 8018 } else { 8019 // Insert effects are inserted at the end of mEffects vector as they are processed 8020 // after track and auxiliary effects. 8021 // Insert effect order as a function of indicated preference: 8022 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 8023 // another effect is present 8024 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 8025 // last effect claiming first position 8026 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 8027 // first effect claiming last position 8028 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 8029 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 8030 // already present 8031 8032 size_t size = mEffects.size(); 8033 size_t idx_insert = size; 8034 ssize_t idx_insert_first = -1; 8035 ssize_t idx_insert_last = -1; 8036 8037 for (size_t i = 0; i < size; i++) { 8038 effect_descriptor_t d = mEffects[i]->desc(); 8039 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 8040 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 8041 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 8042 // check invalid effect chaining combinations 8043 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 8044 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 8045 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 8046 return INVALID_OPERATION; 8047 } 8048 // remember position of first insert effect and by default 8049 // select this as insert position for new effect 8050 if (idx_insert == size) { 8051 idx_insert = i; 8052 } 8053 // remember position of last insert effect claiming 8054 // first position 8055 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 8056 idx_insert_first = i; 8057 } 8058 // remember position of first insert effect claiming 8059 // last position 8060 if (iPref == EFFECT_FLAG_INSERT_LAST && 8061 idx_insert_last == -1) { 8062 idx_insert_last = i; 8063 } 8064 } 8065 } 8066 8067 // modify idx_insert from first position if needed 8068 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 8069 if (idx_insert_last != -1) { 8070 idx_insert = idx_insert_last; 8071 } else { 8072 idx_insert = size; 8073 } 8074 } else { 8075 if (idx_insert_first != -1) { 8076 idx_insert = idx_insert_first + 1; 8077 } 8078 } 8079 8080 // always read samples from chain input buffer 8081 effect->setInBuffer(mInBuffer); 8082 8083 // if last effect in the chain, output samples to chain 8084 // output buffer, otherwise to chain input buffer 8085 if (idx_insert == size) { 8086 if (idx_insert != 0) { 8087 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 8088 mEffects[idx_insert-1]->configure(); 8089 } 8090 effect->setOutBuffer(mOutBuffer); 8091 } else { 8092 effect->setOutBuffer(mInBuffer); 8093 } 8094 mEffects.insertAt(effect, idx_insert); 8095 8096 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 8097 } 8098 effect->configure(); 8099 return NO_ERROR; 8100} 8101 8102// removeEffect_l() must be called with PlaybackThread::mLock held 8103size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 8104{ 8105 Mutex::Autolock _l(mLock); 8106 size_t size = mEffects.size(); 8107 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 8108 8109 for (size_t i = 0; i < size; i++) { 8110 if (effect == mEffects[i]) { 8111 // calling stop here will remove pre-processing effect from the audio HAL. 8112 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 8113 // the middle of a read from audio HAL 8114 if (mEffects[i]->state() == EffectModule::ACTIVE || 8115 mEffects[i]->state() == EffectModule::STOPPING) { 8116 mEffects[i]->stop(); 8117 } 8118 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 8119 delete[] effect->inBuffer(); 8120 } else { 8121 if (i == size - 1 && i != 0) { 8122 mEffects[i - 1]->setOutBuffer(mOutBuffer); 8123 mEffects[i - 1]->configure(); 8124 } 8125 } 8126 mEffects.removeAt(i); 8127 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 8128 break; 8129 } 8130 } 8131 8132 return mEffects.size(); 8133} 8134 8135// setDevice_l() must be called with PlaybackThread::mLock held 8136void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 8137{ 8138 size_t size = mEffects.size(); 8139 for (size_t i = 0; i < size; i++) { 8140 mEffects[i]->setDevice(device); 8141 } 8142} 8143 8144// setMode_l() must be called with PlaybackThread::mLock held 8145void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 8146{ 8147 size_t size = mEffects.size(); 8148 for (size_t i = 0; i < size; i++) { 8149 mEffects[i]->setMode(mode); 8150 } 8151} 8152 8153// setVolume_l() must be called with PlaybackThread::mLock held 8154bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 8155{ 8156 uint32_t newLeft = *left; 8157 uint32_t newRight = *right; 8158 bool hasControl = false; 8159 int ctrlIdx = -1; 8160 size_t size = mEffects.size(); 8161 8162 // first update volume controller 8163 for (size_t i = size; i > 0; i--) { 8164 if (mEffects[i - 1]->isProcessEnabled() && 8165 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 8166 ctrlIdx = i - 1; 8167 hasControl = true; 8168 break; 8169 } 8170 } 8171 8172 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 8173 if (hasControl) { 8174 *left = mNewLeftVolume; 8175 *right = mNewRightVolume; 8176 } 8177 return hasControl; 8178 } 8179 8180 mVolumeCtrlIdx = ctrlIdx; 8181 mLeftVolume = newLeft; 8182 mRightVolume = newRight; 8183 8184 // second get volume update from volume controller 8185 if (ctrlIdx >= 0) { 8186 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 8187 mNewLeftVolume = newLeft; 8188 mNewRightVolume = newRight; 8189 } 8190 // then indicate volume to all other effects in chain. 8191 // Pass altered volume to effects before volume controller 8192 // and requested volume to effects after controller 8193 uint32_t lVol = newLeft; 8194 uint32_t rVol = newRight; 8195 8196 for (size_t i = 0; i < size; i++) { 8197 if ((int)i == ctrlIdx) continue; 8198 // this also works for ctrlIdx == -1 when there is no volume controller 8199 if ((int)i > ctrlIdx) { 8200 lVol = *left; 8201 rVol = *right; 8202 } 8203 mEffects[i]->setVolume(&lVol, &rVol, false); 8204 } 8205 *left = newLeft; 8206 *right = newRight; 8207 8208 return hasControl; 8209} 8210 8211status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 8212{ 8213 const size_t SIZE = 256; 8214 char buffer[SIZE]; 8215 String8 result; 8216 8217 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 8218 result.append(buffer); 8219 8220 bool locked = tryLock(mLock); 8221 // failed to lock - AudioFlinger is probably deadlocked 8222 if (!locked) { 8223 result.append("\tCould not lock mutex:\n"); 8224 } 8225 8226 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 8227 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 8228 mEffects.size(), 8229 (uint32_t)mInBuffer, 8230 (uint32_t)mOutBuffer, 8231 mActiveTrackCnt); 8232 result.append(buffer); 8233 write(fd, result.string(), result.size()); 8234 8235 for (size_t i = 0; i < mEffects.size(); ++i) { 8236 sp<EffectModule> effect = mEffects[i]; 8237 if (effect != 0) { 8238 effect->dump(fd, args); 8239 } 8240 } 8241 8242 if (locked) { 8243 mLock.unlock(); 8244 } 8245 8246 return NO_ERROR; 8247} 8248 8249// must be called with ThreadBase::mLock held 8250void AudioFlinger::EffectChain::setEffectSuspended_l( 8251 const effect_uuid_t *type, bool suspend) 8252{ 8253 sp<SuspendedEffectDesc> desc; 8254 // use effect type UUID timelow as key as there is no real risk of identical 8255 // timeLow fields among effect type UUIDs. 8256 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 8257 if (suspend) { 8258 if (index >= 0) { 8259 desc = mSuspendedEffects.valueAt(index); 8260 } else { 8261 desc = new SuspendedEffectDesc(); 8262 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 8263 mSuspendedEffects.add(type->timeLow, desc); 8264 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 8265 } 8266 if (desc->mRefCount++ == 0) { 8267 sp<EffectModule> effect = getEffectIfEnabled(type); 8268 if (effect != 0) { 8269 desc->mEffect = effect; 8270 effect->setSuspended(true); 8271 effect->setEnabled(false); 8272 } 8273 } 8274 } else { 8275 if (index < 0) { 8276 return; 8277 } 8278 desc = mSuspendedEffects.valueAt(index); 8279 if (desc->mRefCount <= 0) { 8280 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 8281 desc->mRefCount = 1; 8282 } 8283 if (--desc->mRefCount == 0) { 8284 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8285 if (desc->mEffect != 0) { 8286 sp<EffectModule> effect = desc->mEffect.promote(); 8287 if (effect != 0) { 8288 effect->setSuspended(false); 8289 sp<EffectHandle> handle = effect->controlHandle(); 8290 if (handle != 0) { 8291 effect->setEnabled(handle->enabled()); 8292 } 8293 } 8294 desc->mEffect.clear(); 8295 } 8296 mSuspendedEffects.removeItemsAt(index); 8297 } 8298 } 8299} 8300 8301// must be called with ThreadBase::mLock held 8302void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8303{ 8304 sp<SuspendedEffectDesc> desc; 8305 8306 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8307 if (suspend) { 8308 if (index >= 0) { 8309 desc = mSuspendedEffects.valueAt(index); 8310 } else { 8311 desc = new SuspendedEffectDesc(); 8312 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8313 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8314 } 8315 if (desc->mRefCount++ == 0) { 8316 Vector< sp<EffectModule> > effects; 8317 getSuspendEligibleEffects(effects); 8318 for (size_t i = 0; i < effects.size(); i++) { 8319 setEffectSuspended_l(&effects[i]->desc().type, true); 8320 } 8321 } 8322 } else { 8323 if (index < 0) { 8324 return; 8325 } 8326 desc = mSuspendedEffects.valueAt(index); 8327 if (desc->mRefCount <= 0) { 8328 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8329 desc->mRefCount = 1; 8330 } 8331 if (--desc->mRefCount == 0) { 8332 Vector<const effect_uuid_t *> types; 8333 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8334 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8335 continue; 8336 } 8337 types.add(&mSuspendedEffects.valueAt(i)->mType); 8338 } 8339 for (size_t i = 0; i < types.size(); i++) { 8340 setEffectSuspended_l(types[i], false); 8341 } 8342 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8343 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8344 } 8345 } 8346} 8347 8348 8349// The volume effect is used for automated tests only 8350#ifndef OPENSL_ES_H_ 8351static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8352 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8353const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8354#endif //OPENSL_ES_H_ 8355 8356bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8357{ 8358 // auxiliary effects and visualizer are never suspended on output mix 8359 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8360 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8361 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8362 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8363 return false; 8364 } 8365 return true; 8366} 8367 8368void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8369{ 8370 effects.clear(); 8371 for (size_t i = 0; i < mEffects.size(); i++) { 8372 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8373 effects.add(mEffects[i]); 8374 } 8375 } 8376} 8377 8378sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8379 const effect_uuid_t *type) 8380{ 8381 sp<EffectModule> effect = getEffectFromType_l(type); 8382 return effect != 0 && effect->isEnabled() ? effect : 0; 8383} 8384 8385void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8386 bool enabled) 8387{ 8388 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8389 if (enabled) { 8390 if (index < 0) { 8391 // if the effect is not suspend check if all effects are suspended 8392 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8393 if (index < 0) { 8394 return; 8395 } 8396 if (!isEffectEligibleForSuspend(effect->desc())) { 8397 return; 8398 } 8399 setEffectSuspended_l(&effect->desc().type, enabled); 8400 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8401 if (index < 0) { 8402 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8403 return; 8404 } 8405 } 8406 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8407 effect->desc().type.timeLow); 8408 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8409 // if effect is requested to suspended but was not yet enabled, supend it now. 8410 if (desc->mEffect == 0) { 8411 desc->mEffect = effect; 8412 effect->setEnabled(false); 8413 effect->setSuspended(true); 8414 } 8415 } else { 8416 if (index < 0) { 8417 return; 8418 } 8419 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8420 effect->desc().type.timeLow); 8421 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8422 desc->mEffect.clear(); 8423 effect->setSuspended(false); 8424 } 8425} 8426 8427#undef LOG_TAG 8428#define LOG_TAG "AudioFlinger" 8429 8430// ---------------------------------------------------------------------------- 8431 8432status_t AudioFlinger::onTransact( 8433 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8434{ 8435 return BnAudioFlinger::onTransact(code, data, reply, flags); 8436} 8437 8438}; // namespace android 8439