AudioFlinger.cpp revision fce7a473248381cc83a01855f92581077d3c9ee2
1/* //device/include/server/AudioFlinger/AudioFlinger.cpp 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 39#include <media/AudioTrack.h> 40#include <media/AudioRecord.h> 41#include <media/IMediaPlayerService.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <hardware/audio.h> 47#include <hardware/audio_hal.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <media/EffectVisualizerApi.h> 54 55// ---------------------------------------------------------------------------- 56// the sim build doesn't have gettid 57 58#ifndef HAVE_GETTID 59# define gettid getpid 60#endif 61 62// ---------------------------------------------------------------------------- 63 64extern const char * const gEffectLibPath; 65 66namespace android { 67 68static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; 69static const char* kHardwareLockedString = "Hardware lock is taken\n"; 70 71//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 72static const float MAX_GAIN = 4096.0f; 73static const float MAX_GAIN_INT = 0x1000; 74 75// retry counts for buffer fill timeout 76// 50 * ~20msecs = 1 second 77static const int8_t kMaxTrackRetries = 50; 78static const int8_t kMaxTrackStartupRetries = 50; 79// allow less retry attempts on direct output thread. 80// direct outputs can be a scarce resource in audio hardware and should 81// be released as quickly as possible. 82static const int8_t kMaxTrackRetriesDirect = 2; 83 84static const int kDumpLockRetries = 50; 85static const int kDumpLockSleep = 20000; 86 87static const nsecs_t kWarningThrottle = seconds(5); 88 89 90#define AUDIOFLINGER_SECURITY_ENABLED 1 91 92// ---------------------------------------------------------------------------- 93 94static bool recordingAllowed() { 95#ifndef HAVE_ANDROID_OS 96 return true; 97#endif 98#if AUDIOFLINGER_SECURITY_ENABLED 99 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 100 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 101 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 102 return ok; 103#else 104 if (!checkCallingPermission(String16("android.permission.RECORD_AUDIO"))) 105 LOGW("WARNING: Need to add android.permission.RECORD_AUDIO to manifest"); 106 return true; 107#endif 108} 109 110static bool settingsAllowed() { 111#ifndef HAVE_ANDROID_OS 112 return true; 113#endif 114#if AUDIOFLINGER_SECURITY_ENABLED 115 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 116 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 117 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 118 return ok; 119#else 120 if (!checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS"))) 121 LOGW("WARNING: Need to add android.permission.MODIFY_AUDIO_SETTINGS to manifest"); 122 return true; 123#endif 124} 125 126// To collect the amplifier usage 127static void addBatteryData(uint32_t params) { 128 sp<IBinder> binder = 129 defaultServiceManager()->getService(String16("media.player")); 130 sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder); 131 if (service.get() == NULL) { 132 LOGW("Cannot connect to the MediaPlayerService for battery tracking"); 133 return; 134 } 135 136 service->addBatteryData(params); 137} 138 139// ---------------------------------------------------------------------------- 140 141AudioFlinger::AudioFlinger() 142 : BnAudioFlinger(), 143 mAudioHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1) 144{ 145 const hw_module_t *module; 146 int rc; 147 char mod_name[PATH_MAX]; 148 149 Mutex::Autolock _l(mLock); 150 151 mHardwareStatus = AUDIO_HW_IDLE; 152 153 /* get the audio hw module and create an audio_hw device */ 154 snprintf(mod_name, PATH_MAX, "%s.%s", AUDIO_HARDWARE_MODULE_ID, "primary"); 155 rc = hw_get_module(mod_name, &module); 156 if (rc) 157 return; 158 159 rc = audio_hw_device_open(module, &mAudioHardwareDev); 160 LOGE_IF(rc, "couldn't open audio hw device (%s)", strerror(-rc)); 161 if (rc) 162 return; 163 164 mHardwareStatus = AUDIO_HW_INIT; 165 166 rc = mAudioHardwareDev->init_check(mAudioHardwareDev); 167 if (rc == 0) { 168 AutoMutex lock(mHardwareLock); 169 mMode = AUDIO_MODE_NORMAL; 170 mHardwareStatus = AUDIO_HW_SET_MODE; 171 mAudioHardwareDev->set_mode(mAudioHardwareDev, mMode); 172 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 173 mAudioHardwareDev->set_master_volume(mAudioHardwareDev, 1.0f); 174 mHardwareStatus = AUDIO_HW_IDLE; 175 } else { 176 LOGE("Couldn't even initialize the stubbed audio hardware!"); 177 } 178} 179 180AudioFlinger::~AudioFlinger() 181{ 182 while (!mRecordThreads.isEmpty()) { 183 // closeInput() will remove first entry from mRecordThreads 184 closeInput(mRecordThreads.keyAt(0)); 185 } 186 while (!mPlaybackThreads.isEmpty()) { 187 // closeOutput() will remove first entry from mPlaybackThreads 188 closeOutput(mPlaybackThreads.keyAt(0)); 189 } 190 if (mAudioHardwareDev) { 191 audio_hw_device_close(mAudioHardwareDev); 192 } 193} 194 195 196 197status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 198{ 199 const size_t SIZE = 256; 200 char buffer[SIZE]; 201 String8 result; 202 203 result.append("Clients:\n"); 204 for (size_t i = 0; i < mClients.size(); ++i) { 205 wp<Client> wClient = mClients.valueAt(i); 206 if (wClient != 0) { 207 sp<Client> client = wClient.promote(); 208 if (client != 0) { 209 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 210 result.append(buffer); 211 } 212 } 213 } 214 write(fd, result.string(), result.size()); 215 return NO_ERROR; 216} 217 218 219status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 220{ 221 const size_t SIZE = 256; 222 char buffer[SIZE]; 223 String8 result; 224 int hardwareStatus = mHardwareStatus; 225 226 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 227 result.append(buffer); 228 write(fd, result.string(), result.size()); 229 return NO_ERROR; 230} 231 232status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 233{ 234 const size_t SIZE = 256; 235 char buffer[SIZE]; 236 String8 result; 237 snprintf(buffer, SIZE, "Permission Denial: " 238 "can't dump AudioFlinger from pid=%d, uid=%d\n", 239 IPCThreadState::self()->getCallingPid(), 240 IPCThreadState::self()->getCallingUid()); 241 result.append(buffer); 242 write(fd, result.string(), result.size()); 243 return NO_ERROR; 244} 245 246static bool tryLock(Mutex& mutex) 247{ 248 bool locked = false; 249 for (int i = 0; i < kDumpLockRetries; ++i) { 250 if (mutex.tryLock() == NO_ERROR) { 251 locked = true; 252 break; 253 } 254 usleep(kDumpLockSleep); 255 } 256 return locked; 257} 258 259status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 260{ 261 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 262 dumpPermissionDenial(fd, args); 263 } else { 264 // get state of hardware lock 265 bool hardwareLocked = tryLock(mHardwareLock); 266 if (!hardwareLocked) { 267 String8 result(kHardwareLockedString); 268 write(fd, result.string(), result.size()); 269 } else { 270 mHardwareLock.unlock(); 271 } 272 273 bool locked = tryLock(mLock); 274 275 // failed to lock - AudioFlinger is probably deadlocked 276 if (!locked) { 277 String8 result(kDeadlockedString); 278 write(fd, result.string(), result.size()); 279 } 280 281 dumpClients(fd, args); 282 dumpInternals(fd, args); 283 284 // dump playback threads 285 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 286 mPlaybackThreads.valueAt(i)->dump(fd, args); 287 } 288 289 // dump record threads 290 for (size_t i = 0; i < mRecordThreads.size(); i++) { 291 mRecordThreads.valueAt(i)->dump(fd, args); 292 } 293 294 if (mAudioHardwareDev) { 295 mAudioHardwareDev->dump(mAudioHardwareDev, fd); 296 } 297 if (locked) mLock.unlock(); 298 } 299 return NO_ERROR; 300} 301 302 303// IAudioFlinger interface 304 305 306sp<IAudioTrack> AudioFlinger::createTrack( 307 pid_t pid, 308 int streamType, 309 uint32_t sampleRate, 310 int format, 311 int channelCount, 312 int frameCount, 313 uint32_t flags, 314 const sp<IMemory>& sharedBuffer, 315 int output, 316 int *sessionId, 317 status_t *status) 318{ 319 sp<PlaybackThread::Track> track; 320 sp<TrackHandle> trackHandle; 321 sp<Client> client; 322 wp<Client> wclient; 323 status_t lStatus; 324 int lSessionId; 325 326 if (streamType >= AUDIO_STREAM_CNT) { 327 LOGE("invalid stream type"); 328 lStatus = BAD_VALUE; 329 goto Exit; 330 } 331 332 { 333 Mutex::Autolock _l(mLock); 334 PlaybackThread *thread = checkPlaybackThread_l(output); 335 PlaybackThread *effectThread = NULL; 336 if (thread == NULL) { 337 LOGE("unknown output thread"); 338 lStatus = BAD_VALUE; 339 goto Exit; 340 } 341 342 wclient = mClients.valueFor(pid); 343 344 if (wclient != NULL) { 345 client = wclient.promote(); 346 } else { 347 client = new Client(this, pid); 348 mClients.add(pid, client); 349 } 350 351 LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 352 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 353 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 354 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 355 if (mPlaybackThreads.keyAt(i) != output) { 356 // prevent same audio session on different output threads 357 uint32_t sessions = t->hasAudioSession(*sessionId); 358 if (sessions & PlaybackThread::TRACK_SESSION) { 359 lStatus = BAD_VALUE; 360 goto Exit; 361 } 362 // check if an effect with same session ID is waiting for a track to be created 363 if (sessions & PlaybackThread::EFFECT_SESSION) { 364 effectThread = t.get(); 365 } 366 } 367 } 368 lSessionId = *sessionId; 369 } else { 370 // if no audio session id is provided, create one here 371 lSessionId = nextUniqueId_l(); 372 if (sessionId != NULL) { 373 *sessionId = lSessionId; 374 } 375 } 376 LOGV("createTrack() lSessionId: %d", lSessionId); 377 378 track = thread->createTrack_l(client, streamType, sampleRate, format, 379 channelCount, frameCount, sharedBuffer, lSessionId, &lStatus); 380 381 // move effect chain to this output thread if an effect on same session was waiting 382 // for a track to be created 383 if (lStatus == NO_ERROR && effectThread != NULL) { 384 Mutex::Autolock _dl(thread->mLock); 385 Mutex::Autolock _sl(effectThread->mLock); 386 moveEffectChain_l(lSessionId, effectThread, thread, true); 387 } 388 } 389 if (lStatus == NO_ERROR) { 390 trackHandle = new TrackHandle(track); 391 } else { 392 // remove local strong reference to Client before deleting the Track so that the Client 393 // destructor is called by the TrackBase destructor with mLock held 394 client.clear(); 395 track.clear(); 396 } 397 398Exit: 399 if(status) { 400 *status = lStatus; 401 } 402 return trackHandle; 403} 404 405uint32_t AudioFlinger::sampleRate(int output) const 406{ 407 Mutex::Autolock _l(mLock); 408 PlaybackThread *thread = checkPlaybackThread_l(output); 409 if (thread == NULL) { 410 LOGW("sampleRate() unknown thread %d", output); 411 return 0; 412 } 413 return thread->sampleRate(); 414} 415 416int AudioFlinger::channelCount(int output) const 417{ 418 Mutex::Autolock _l(mLock); 419 PlaybackThread *thread = checkPlaybackThread_l(output); 420 if (thread == NULL) { 421 LOGW("channelCount() unknown thread %d", output); 422 return 0; 423 } 424 return thread->channelCount(); 425} 426 427int AudioFlinger::format(int output) const 428{ 429 Mutex::Autolock _l(mLock); 430 PlaybackThread *thread = checkPlaybackThread_l(output); 431 if (thread == NULL) { 432 LOGW("format() unknown thread %d", output); 433 return 0; 434 } 435 return thread->format(); 436} 437 438size_t AudioFlinger::frameCount(int output) const 439{ 440 Mutex::Autolock _l(mLock); 441 PlaybackThread *thread = checkPlaybackThread_l(output); 442 if (thread == NULL) { 443 LOGW("frameCount() unknown thread %d", output); 444 return 0; 445 } 446 return thread->frameCount(); 447} 448 449uint32_t AudioFlinger::latency(int output) const 450{ 451 Mutex::Autolock _l(mLock); 452 PlaybackThread *thread = checkPlaybackThread_l(output); 453 if (thread == NULL) { 454 LOGW("latency() unknown thread %d", output); 455 return 0; 456 } 457 return thread->latency(); 458} 459 460status_t AudioFlinger::setMasterVolume(float value) 461{ 462 // check calling permissions 463 if (!settingsAllowed()) { 464 return PERMISSION_DENIED; 465 } 466 467 // when hw supports master volume, don't scale in sw mixer 468 { // scope for the lock 469 AutoMutex lock(mHardwareLock); 470 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 471 if (mAudioHardwareDev->set_master_volume(mAudioHardwareDev, value) == NO_ERROR) { 472 value = 1.0f; 473 } 474 mHardwareStatus = AUDIO_HW_IDLE; 475 } 476 477 Mutex::Autolock _l(mLock); 478 mMasterVolume = value; 479 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 480 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 481 482 return NO_ERROR; 483} 484 485status_t AudioFlinger::setMode(int mode) 486{ 487 status_t ret; 488 489 // check calling permissions 490 if (!settingsAllowed()) { 491 return PERMISSION_DENIED; 492 } 493 if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) { 494 LOGW("Illegal value: setMode(%d)", mode); 495 return BAD_VALUE; 496 } 497 498 { // scope for the lock 499 AutoMutex lock(mHardwareLock); 500 mHardwareStatus = AUDIO_HW_SET_MODE; 501 ret = mAudioHardwareDev->set_mode(mAudioHardwareDev, mode); 502 mHardwareStatus = AUDIO_HW_IDLE; 503 } 504 505 if (NO_ERROR == ret) { 506 Mutex::Autolock _l(mLock); 507 mMode = mode; 508 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 509 mPlaybackThreads.valueAt(i)->setMode(mode); 510 } 511 512 return ret; 513} 514 515status_t AudioFlinger::setMicMute(bool state) 516{ 517 // check calling permissions 518 if (!settingsAllowed()) { 519 return PERMISSION_DENIED; 520 } 521 522 AutoMutex lock(mHardwareLock); 523 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 524 status_t ret = mAudioHardwareDev->set_mic_mute(mAudioHardwareDev, state); 525 mHardwareStatus = AUDIO_HW_IDLE; 526 return ret; 527} 528 529bool AudioFlinger::getMicMute() const 530{ 531 bool state = AUDIO_MODE_INVALID; 532 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 533 mAudioHardwareDev->get_mic_mute(mAudioHardwareDev, &state); 534 mHardwareStatus = AUDIO_HW_IDLE; 535 return state; 536} 537 538status_t AudioFlinger::setMasterMute(bool muted) 539{ 540 // check calling permissions 541 if (!settingsAllowed()) { 542 return PERMISSION_DENIED; 543 } 544 545 Mutex::Autolock _l(mLock); 546 mMasterMute = muted; 547 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 548 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 549 550 return NO_ERROR; 551} 552 553float AudioFlinger::masterVolume() const 554{ 555 return mMasterVolume; 556} 557 558bool AudioFlinger::masterMute() const 559{ 560 return mMasterMute; 561} 562 563status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 564{ 565 // check calling permissions 566 if (!settingsAllowed()) { 567 return PERMISSION_DENIED; 568 } 569 570 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 571 return BAD_VALUE; 572 } 573 574 AutoMutex lock(mLock); 575 PlaybackThread *thread = NULL; 576 if (output) { 577 thread = checkPlaybackThread_l(output); 578 if (thread == NULL) { 579 return BAD_VALUE; 580 } 581 } 582 583 mStreamTypes[stream].volume = value; 584 585 if (thread == NULL) { 586 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 587 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 588 } 589 } else { 590 thread->setStreamVolume(stream, value); 591 } 592 593 return NO_ERROR; 594} 595 596status_t AudioFlinger::setStreamMute(int stream, bool muted) 597{ 598 // check calling permissions 599 if (!settingsAllowed()) { 600 return PERMISSION_DENIED; 601 } 602 603 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || 604 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 605 return BAD_VALUE; 606 } 607 608 AutoMutex lock(mLock); 609 mStreamTypes[stream].mute = muted; 610 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 611 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 612 613 return NO_ERROR; 614} 615 616float AudioFlinger::streamVolume(int stream, int output) const 617{ 618 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 619 return 0.0f; 620 } 621 622 AutoMutex lock(mLock); 623 float volume; 624 if (output) { 625 PlaybackThread *thread = checkPlaybackThread_l(output); 626 if (thread == NULL) { 627 return 0.0f; 628 } 629 volume = thread->streamVolume(stream); 630 } else { 631 volume = mStreamTypes[stream].volume; 632 } 633 634 return volume; 635} 636 637bool AudioFlinger::streamMute(int stream) const 638{ 639 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 640 return true; 641 } 642 643 return mStreamTypes[stream].mute; 644} 645 646status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 647{ 648 status_t result; 649 650 LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 651 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 652 // check calling permissions 653 if (!settingsAllowed()) { 654 return PERMISSION_DENIED; 655 } 656 657 // ioHandle == 0 means the parameters are global to the audio hardware interface 658 if (ioHandle == 0) { 659 AutoMutex lock(mHardwareLock); 660 mHardwareStatus = AUDIO_SET_PARAMETER; 661 result = mAudioHardwareDev->set_parameters(mAudioHardwareDev, keyValuePairs.string()); 662 mHardwareStatus = AUDIO_HW_IDLE; 663 return result; 664 } 665 666 // hold a strong ref on thread in case closeOutput() or closeInput() is called 667 // and the thread is exited once the lock is released 668 sp<ThreadBase> thread; 669 { 670 Mutex::Autolock _l(mLock); 671 thread = checkPlaybackThread_l(ioHandle); 672 if (thread == NULL) { 673 thread = checkRecordThread_l(ioHandle); 674 } 675 } 676 if (thread != NULL) { 677 result = thread->setParameters(keyValuePairs); 678 return result; 679 } 680 return BAD_VALUE; 681} 682 683String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 684{ 685// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 686// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 687 688 if (ioHandle == 0) { 689 char *s; 690 String8 out_s8; 691 692 s = mAudioHardwareDev->get_parameters(mAudioHardwareDev, keys.string()); 693 out_s8 = String8(s); 694 free(s); 695 return out_s8; 696 } 697 698 Mutex::Autolock _l(mLock); 699 700 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 701 if (playbackThread != NULL) { 702 return playbackThread->getParameters(keys); 703 } 704 RecordThread *recordThread = checkRecordThread_l(ioHandle); 705 if (recordThread != NULL) { 706 return recordThread->getParameters(keys); 707 } 708 return String8(""); 709} 710 711size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 712{ 713 return mAudioHardwareDev->get_input_buffer_size(mAudioHardwareDev, sampleRate, format, channelCount); 714} 715 716unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 717{ 718 if (ioHandle == 0) { 719 return 0; 720 } 721 722 Mutex::Autolock _l(mLock); 723 724 RecordThread *recordThread = checkRecordThread_l(ioHandle); 725 if (recordThread != NULL) { 726 return recordThread->getInputFramesLost(); 727 } 728 return 0; 729} 730 731status_t AudioFlinger::setVoiceVolume(float value) 732{ 733 // check calling permissions 734 if (!settingsAllowed()) { 735 return PERMISSION_DENIED; 736 } 737 738 AutoMutex lock(mHardwareLock); 739 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 740 status_t ret = mAudioHardwareDev->set_voice_volume(mAudioHardwareDev, value); 741 mHardwareStatus = AUDIO_HW_IDLE; 742 743 return ret; 744} 745 746status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 747{ 748 status_t status; 749 750 Mutex::Autolock _l(mLock); 751 752 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 753 if (playbackThread != NULL) { 754 return playbackThread->getRenderPosition(halFrames, dspFrames); 755 } 756 757 return BAD_VALUE; 758} 759 760void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 761{ 762 763 Mutex::Autolock _l(mLock); 764 765 int pid = IPCThreadState::self()->getCallingPid(); 766 if (mNotificationClients.indexOfKey(pid) < 0) { 767 sp<NotificationClient> notificationClient = new NotificationClient(this, 768 client, 769 pid); 770 LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 771 772 mNotificationClients.add(pid, notificationClient); 773 774 sp<IBinder> binder = client->asBinder(); 775 binder->linkToDeath(notificationClient); 776 777 // the config change is always sent from playback or record threads to avoid deadlock 778 // with AudioSystem::gLock 779 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 780 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 781 } 782 783 for (size_t i = 0; i < mRecordThreads.size(); i++) { 784 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 785 } 786 } 787} 788 789void AudioFlinger::removeNotificationClient(pid_t pid) 790{ 791 Mutex::Autolock _l(mLock); 792 793 int index = mNotificationClients.indexOfKey(pid); 794 if (index >= 0) { 795 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 796 LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 797 mNotificationClients.removeItem(pid); 798 } 799} 800 801// audioConfigChanged_l() must be called with AudioFlinger::mLock held 802void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 803{ 804 size_t size = mNotificationClients.size(); 805 for (size_t i = 0; i < size; i++) { 806 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 807 } 808} 809 810// removeClient_l() must be called with AudioFlinger::mLock held 811void AudioFlinger::removeClient_l(pid_t pid) 812{ 813 LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 814 mClients.removeItem(pid); 815} 816 817 818// ---------------------------------------------------------------------------- 819 820AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id) 821 : Thread(false), 822 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 823 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false) 824{ 825} 826 827AudioFlinger::ThreadBase::~ThreadBase() 828{ 829 mParamCond.broadcast(); 830 mNewParameters.clear(); 831} 832 833void AudioFlinger::ThreadBase::exit() 834{ 835 // keep a strong ref on ourself so that we wont get 836 // destroyed in the middle of requestExitAndWait() 837 sp <ThreadBase> strongMe = this; 838 839 LOGV("ThreadBase::exit"); 840 { 841 AutoMutex lock(&mLock); 842 mExiting = true; 843 requestExit(); 844 mWaitWorkCV.signal(); 845 } 846 requestExitAndWait(); 847} 848 849uint32_t AudioFlinger::ThreadBase::sampleRate() const 850{ 851 return mSampleRate; 852} 853 854int AudioFlinger::ThreadBase::channelCount() const 855{ 856 return (int)mChannelCount; 857} 858 859int AudioFlinger::ThreadBase::format() const 860{ 861 return mFormat; 862} 863 864size_t AudioFlinger::ThreadBase::frameCount() const 865{ 866 return mFrameCount; 867} 868 869status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 870{ 871 status_t status; 872 873 LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 874 Mutex::Autolock _l(mLock); 875 876 mNewParameters.add(keyValuePairs); 877 mWaitWorkCV.signal(); 878 // wait condition with timeout in case the thread loop has exited 879 // before the request could be processed 880 if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { 881 status = mParamStatus; 882 mWaitWorkCV.signal(); 883 } else { 884 status = TIMED_OUT; 885 } 886 return status; 887} 888 889void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 890{ 891 Mutex::Autolock _l(mLock); 892 sendConfigEvent_l(event, param); 893} 894 895// sendConfigEvent_l() must be called with ThreadBase::mLock held 896void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 897{ 898 ConfigEvent *configEvent = new ConfigEvent(); 899 configEvent->mEvent = event; 900 configEvent->mParam = param; 901 mConfigEvents.add(configEvent); 902 LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 903 mWaitWorkCV.signal(); 904} 905 906void AudioFlinger::ThreadBase::processConfigEvents() 907{ 908 mLock.lock(); 909 while(!mConfigEvents.isEmpty()) { 910 LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 911 ConfigEvent *configEvent = mConfigEvents[0]; 912 mConfigEvents.removeAt(0); 913 // release mLock before locking AudioFlinger mLock: lock order is always 914 // AudioFlinger then ThreadBase to avoid cross deadlock 915 mLock.unlock(); 916 mAudioFlinger->mLock.lock(); 917 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 918 mAudioFlinger->mLock.unlock(); 919 delete configEvent; 920 mLock.lock(); 921 } 922 mLock.unlock(); 923} 924 925status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 926{ 927 const size_t SIZE = 256; 928 char buffer[SIZE]; 929 String8 result; 930 931 bool locked = tryLock(mLock); 932 if (!locked) { 933 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 934 write(fd, buffer, strlen(buffer)); 935 } 936 937 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 938 result.append(buffer); 939 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 940 result.append(buffer); 941 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 942 result.append(buffer); 943 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 944 result.append(buffer); 945 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 946 result.append(buffer); 947 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 948 result.append(buffer); 949 950 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 951 result.append(buffer); 952 result.append(" Index Command"); 953 for (size_t i = 0; i < mNewParameters.size(); ++i) { 954 snprintf(buffer, SIZE, "\n %02d ", i); 955 result.append(buffer); 956 result.append(mNewParameters[i]); 957 } 958 959 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 960 result.append(buffer); 961 snprintf(buffer, SIZE, " Index event param\n"); 962 result.append(buffer); 963 for (size_t i = 0; i < mConfigEvents.size(); i++) { 964 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 965 result.append(buffer); 966 } 967 result.append("\n"); 968 969 write(fd, result.string(), result.size()); 970 971 if (locked) { 972 mLock.unlock(); 973 } 974 return NO_ERROR; 975} 976 977 978// ---------------------------------------------------------------------------- 979 980AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, struct audio_stream_out* output, int id, uint32_t device) 981 : ThreadBase(audioFlinger, id), 982 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 983 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 984 mDevice(device) 985{ 986 readOutputParameters(); 987 988 mMasterVolume = mAudioFlinger->masterVolume(); 989 mMasterMute = mAudioFlinger->masterMute(); 990 991 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 992 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 993 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 994 } 995} 996 997AudioFlinger::PlaybackThread::~PlaybackThread() 998{ 999 delete [] mMixBuffer; 1000} 1001 1002status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1003{ 1004 dumpInternals(fd, args); 1005 dumpTracks(fd, args); 1006 dumpEffectChains(fd, args); 1007 return NO_ERROR; 1008} 1009 1010status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1011{ 1012 const size_t SIZE = 256; 1013 char buffer[SIZE]; 1014 String8 result; 1015 1016 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1017 result.append(buffer); 1018 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1019 for (size_t i = 0; i < mTracks.size(); ++i) { 1020 sp<Track> track = mTracks[i]; 1021 if (track != 0) { 1022 track->dump(buffer, SIZE); 1023 result.append(buffer); 1024 } 1025 } 1026 1027 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1028 result.append(buffer); 1029 result.append(" Name Clien Typ Fmt Chn Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1030 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1031 wp<Track> wTrack = mActiveTracks[i]; 1032 if (wTrack != 0) { 1033 sp<Track> track = wTrack.promote(); 1034 if (track != 0) { 1035 track->dump(buffer, SIZE); 1036 result.append(buffer); 1037 } 1038 } 1039 } 1040 write(fd, result.string(), result.size()); 1041 return NO_ERROR; 1042} 1043 1044status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args) 1045{ 1046 const size_t SIZE = 256; 1047 char buffer[SIZE]; 1048 String8 result; 1049 1050 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1051 write(fd, buffer, strlen(buffer)); 1052 1053 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1054 sp<EffectChain> chain = mEffectChains[i]; 1055 if (chain != 0) { 1056 chain->dump(fd, args); 1057 } 1058 } 1059 return NO_ERROR; 1060} 1061 1062status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1063{ 1064 const size_t SIZE = 256; 1065 char buffer[SIZE]; 1066 String8 result; 1067 1068 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1069 result.append(buffer); 1070 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1071 result.append(buffer); 1072 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1073 result.append(buffer); 1074 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1075 result.append(buffer); 1076 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1077 result.append(buffer); 1078 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1079 result.append(buffer); 1080 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1081 result.append(buffer); 1082 write(fd, result.string(), result.size()); 1083 1084 dumpBase(fd, args); 1085 1086 return NO_ERROR; 1087} 1088 1089// Thread virtuals 1090status_t AudioFlinger::PlaybackThread::readyToRun() 1091{ 1092 if (mSampleRate == 0) { 1093 LOGE("No working audio driver found."); 1094 return NO_INIT; 1095 } 1096 LOGI("AudioFlinger's thread %p ready to run", this); 1097 return NO_ERROR; 1098} 1099 1100void AudioFlinger::PlaybackThread::onFirstRef() 1101{ 1102 const size_t SIZE = 256; 1103 char buffer[SIZE]; 1104 1105 snprintf(buffer, SIZE, "Playback Thread %p", this); 1106 1107 run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); 1108} 1109 1110// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1111sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1112 const sp<AudioFlinger::Client>& client, 1113 int streamType, 1114 uint32_t sampleRate, 1115 int format, 1116 int channelCount, 1117 int frameCount, 1118 const sp<IMemory>& sharedBuffer, 1119 int sessionId, 1120 status_t *status) 1121{ 1122 sp<Track> track; 1123 status_t lStatus; 1124 1125 if (mType == DIRECT) { 1126 if (sampleRate != mSampleRate || format != mFormat || channelCount != (int)mChannelCount) { 1127 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelCount %d for output %p", 1128 sampleRate, format, channelCount, mOutput); 1129 lStatus = BAD_VALUE; 1130 goto Exit; 1131 } 1132 } else { 1133 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1134 if (sampleRate > mSampleRate*2) { 1135 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1136 lStatus = BAD_VALUE; 1137 goto Exit; 1138 } 1139 } 1140 1141 if (mOutput == 0) { 1142 LOGE("Audio driver not initialized."); 1143 lStatus = NO_INIT; 1144 goto Exit; 1145 } 1146 1147 { // scope for mLock 1148 Mutex::Autolock _l(mLock); 1149 1150 // all tracks in same audio session must share the same routing strategy otherwise 1151 // conflicts will happen when tracks are moved from one output to another by audio policy 1152 // manager 1153 uint32_t strategy = 1154 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1155 for (size_t i = 0; i < mTracks.size(); ++i) { 1156 sp<Track> t = mTracks[i]; 1157 if (t != 0) { 1158 if (sessionId == t->sessionId() && 1159 strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) { 1160 lStatus = BAD_VALUE; 1161 goto Exit; 1162 } 1163 } 1164 } 1165 1166 track = new Track(this, client, streamType, sampleRate, format, 1167 channelCount, frameCount, sharedBuffer, sessionId); 1168 if (track->getCblk() == NULL || track->name() < 0) { 1169 lStatus = NO_MEMORY; 1170 goto Exit; 1171 } 1172 mTracks.add(track); 1173 1174 sp<EffectChain> chain = getEffectChain_l(sessionId); 1175 if (chain != 0) { 1176 LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1177 track->setMainBuffer(chain->inBuffer()); 1178 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1179 } 1180 } 1181 lStatus = NO_ERROR; 1182 1183Exit: 1184 if(status) { 1185 *status = lStatus; 1186 } 1187 return track; 1188} 1189 1190uint32_t AudioFlinger::PlaybackThread::latency() const 1191{ 1192 if (mOutput) { 1193 return mOutput->get_latency(mOutput); 1194 } 1195 else { 1196 return 0; 1197 } 1198} 1199 1200status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1201{ 1202 mMasterVolume = value; 1203 return NO_ERROR; 1204} 1205 1206status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1207{ 1208 mMasterMute = muted; 1209 return NO_ERROR; 1210} 1211 1212float AudioFlinger::PlaybackThread::masterVolume() const 1213{ 1214 return mMasterVolume; 1215} 1216 1217bool AudioFlinger::PlaybackThread::masterMute() const 1218{ 1219 return mMasterMute; 1220} 1221 1222status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1223{ 1224 mStreamTypes[stream].volume = value; 1225 return NO_ERROR; 1226} 1227 1228status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1229{ 1230 mStreamTypes[stream].mute = muted; 1231 return NO_ERROR; 1232} 1233 1234float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1235{ 1236 return mStreamTypes[stream].volume; 1237} 1238 1239bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1240{ 1241 return mStreamTypes[stream].mute; 1242} 1243 1244// addTrack_l() must be called with ThreadBase::mLock held 1245status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1246{ 1247 status_t status = ALREADY_EXISTS; 1248 1249 // set retry count for buffer fill 1250 track->mRetryCount = kMaxTrackStartupRetries; 1251 if (mActiveTracks.indexOf(track) < 0) { 1252 // the track is newly added, make sure it fills up all its 1253 // buffers before playing. This is to ensure the client will 1254 // effectively get the latency it requested. 1255 track->mFillingUpStatus = Track::FS_FILLING; 1256 track->mResetDone = false; 1257 mActiveTracks.add(track); 1258 if (track->mainBuffer() != mMixBuffer) { 1259 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1260 if (chain != 0) { 1261 LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1262 chain->startTrack(); 1263 } 1264 } 1265 1266 status = NO_ERROR; 1267 } 1268 1269 LOGV("mWaitWorkCV.broadcast"); 1270 mWaitWorkCV.broadcast(); 1271 1272 return status; 1273} 1274 1275// destroyTrack_l() must be called with ThreadBase::mLock held 1276void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1277{ 1278 track->mState = TrackBase::TERMINATED; 1279 if (mActiveTracks.indexOf(track) < 0) { 1280 mTracks.remove(track); 1281 deleteTrackName_l(track->name()); 1282 } 1283} 1284 1285String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1286{ 1287 String8 out_s8; 1288 char *s; 1289 1290 s = mOutput->common.get_parameters(&mOutput->common, keys.string()); 1291 out_s8 = String8(s); 1292 free(s); 1293 return out_s8; 1294} 1295 1296// destroyTrack_l() must be called with AudioFlinger::mLock held 1297void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1298 AudioSystem::OutputDescriptor desc; 1299 void *param2 = 0; 1300 1301 LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1302 1303 switch (event) { 1304 case AudioSystem::OUTPUT_OPENED: 1305 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1306 desc.channels = mChannels; 1307 desc.samplingRate = mSampleRate; 1308 desc.format = mFormat; 1309 desc.frameCount = mFrameCount; 1310 desc.latency = latency(); 1311 param2 = &desc; 1312 break; 1313 1314 case AudioSystem::STREAM_CONFIG_CHANGED: 1315 param2 = ¶m; 1316 case AudioSystem::OUTPUT_CLOSED: 1317 default: 1318 break; 1319 } 1320 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1321} 1322 1323void AudioFlinger::PlaybackThread::readOutputParameters() 1324{ 1325 mSampleRate = mOutput->common.get_sample_rate(&mOutput->common); 1326 mChannels = mOutput->common.get_channels(&mOutput->common); 1327 mChannelCount = (uint16_t)popcount(mChannels); 1328 mFormat = mOutput->common.get_format(&mOutput->common); 1329 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->common); 1330 mFrameCount = mOutput->common.get_buffer_size(&mOutput->common) / mFrameSize; 1331 1332 // FIXME - Current mixer implementation only supports stereo output: Always 1333 // Allocate a stereo buffer even if HW output is mono. 1334 if (mMixBuffer != NULL) delete[] mMixBuffer; 1335 mMixBuffer = new int16_t[mFrameCount * 2]; 1336 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1337 1338 // force reconfiguration of effect chains and engines to take new buffer size and audio 1339 // parameters into account 1340 // Note that mLock is not held when readOutputParameters() is called from the constructor 1341 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1342 // matter. 1343 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1344 Vector< sp<EffectChain> > effectChains = mEffectChains; 1345 for (size_t i = 0; i < effectChains.size(); i ++) { 1346 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1347 } 1348} 1349 1350status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1351{ 1352 if (halFrames == 0 || dspFrames == 0) { 1353 return BAD_VALUE; 1354 } 1355 if (mOutput == 0) { 1356 return INVALID_OPERATION; 1357 } 1358 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->common); 1359 1360 return mOutput->get_render_position(mOutput, dspFrames); 1361} 1362 1363uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1364{ 1365 Mutex::Autolock _l(mLock); 1366 uint32_t result = 0; 1367 if (getEffectChain_l(sessionId) != 0) { 1368 result = EFFECT_SESSION; 1369 } 1370 1371 for (size_t i = 0; i < mTracks.size(); ++i) { 1372 sp<Track> track = mTracks[i]; 1373 if (sessionId == track->sessionId() && 1374 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1375 result |= TRACK_SESSION; 1376 break; 1377 } 1378 } 1379 1380 return result; 1381} 1382 1383uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1384{ 1385 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1386 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1387 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1388 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1389 } 1390 for (size_t i = 0; i < mTracks.size(); i++) { 1391 sp<Track> track = mTracks[i]; 1392 if (sessionId == track->sessionId() && 1393 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1394 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1395 } 1396 } 1397 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1398} 1399 1400sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId) 1401{ 1402 Mutex::Autolock _l(mLock); 1403 return getEffectChain_l(sessionId); 1404} 1405 1406sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId) 1407{ 1408 sp<EffectChain> chain; 1409 1410 size_t size = mEffectChains.size(); 1411 for (size_t i = 0; i < size; i++) { 1412 if (mEffectChains[i]->sessionId() == sessionId) { 1413 chain = mEffectChains[i]; 1414 break; 1415 } 1416 } 1417 return chain; 1418} 1419 1420void AudioFlinger::PlaybackThread::setMode(uint32_t mode) 1421{ 1422 Mutex::Autolock _l(mLock); 1423 size_t size = mEffectChains.size(); 1424 for (size_t i = 0; i < size; i++) { 1425 mEffectChains[i]->setMode_l(mode); 1426 } 1427} 1428 1429// ---------------------------------------------------------------------------- 1430 1431AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, struct audio_stream_out* output, int id, uint32_t device) 1432 : PlaybackThread(audioFlinger, output, id, device), 1433 mAudioMixer(0) 1434{ 1435 mType = PlaybackThread::MIXER; 1436 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1437 1438 // FIXME - Current mixer implementation only supports stereo output 1439 if (mChannelCount == 1) { 1440 LOGE("Invalid audio hardware channel count"); 1441 } 1442} 1443 1444AudioFlinger::MixerThread::~MixerThread() 1445{ 1446 delete mAudioMixer; 1447} 1448 1449bool AudioFlinger::MixerThread::threadLoop() 1450{ 1451 Vector< sp<Track> > tracksToRemove; 1452 uint32_t mixerStatus = MIXER_IDLE; 1453 nsecs_t standbyTime = systemTime(); 1454 size_t mixBufferSize = mFrameCount * mFrameSize; 1455 // FIXME: Relaxed timing because of a certain device that can't meet latency 1456 // Should be reduced to 2x after the vendor fixes the driver issue 1457 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1458 nsecs_t lastWarning = 0; 1459 bool longStandbyExit = false; 1460 uint32_t activeSleepTime = activeSleepTimeUs(); 1461 uint32_t idleSleepTime = idleSleepTimeUs(); 1462 uint32_t sleepTime = idleSleepTime; 1463 Vector< sp<EffectChain> > effectChains; 1464 1465 while (!exitPending()) 1466 { 1467 processConfigEvents(); 1468 1469 mixerStatus = MIXER_IDLE; 1470 { // scope for mLock 1471 1472 Mutex::Autolock _l(mLock); 1473 1474 if (checkForNewParameters_l()) { 1475 mixBufferSize = mFrameCount * mFrameSize; 1476 // FIXME: Relaxed timing because of a certain device that can't meet latency 1477 // Should be reduced to 2x after the vendor fixes the driver issue 1478 maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1479 activeSleepTime = activeSleepTimeUs(); 1480 idleSleepTime = idleSleepTimeUs(); 1481 } 1482 1483 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1484 1485 // put audio hardware into standby after short delay 1486 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1487 mSuspended) { 1488 if (!mStandby) { 1489 LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1490 mOutput->common.standby(&mOutput->common); 1491 mStandby = true; 1492 mBytesWritten = 0; 1493 } 1494 1495 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1496 // we're about to wait, flush the binder command buffer 1497 IPCThreadState::self()->flushCommands(); 1498 1499 if (exitPending()) break; 1500 1501 // wait until we have something to do... 1502 LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1503 mWaitWorkCV.wait(mLock); 1504 LOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1505 1506 if (mMasterMute == false) { 1507 char value[PROPERTY_VALUE_MAX]; 1508 property_get("ro.audio.silent", value, "0"); 1509 if (atoi(value)) { 1510 LOGD("Silence is golden"); 1511 setMasterMute(true); 1512 } 1513 } 1514 1515 standbyTime = systemTime() + kStandbyTimeInNsecs; 1516 sleepTime = idleSleepTime; 1517 continue; 1518 } 1519 } 1520 1521 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1522 1523 // prevent any changes in effect chain list and in each effect chain 1524 // during mixing and effect process as the audio buffers could be deleted 1525 // or modified if an effect is created or deleted 1526 lockEffectChains_l(effectChains); 1527 } 1528 1529 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1530 // mix buffers... 1531 mAudioMixer->process(); 1532 sleepTime = 0; 1533 standbyTime = systemTime() + kStandbyTimeInNsecs; 1534 //TODO: delay standby when effects have a tail 1535 } else { 1536 // If no tracks are ready, sleep once for the duration of an output 1537 // buffer size, then write 0s to the output 1538 if (sleepTime == 0) { 1539 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1540 sleepTime = activeSleepTime; 1541 } else { 1542 sleepTime = idleSleepTime; 1543 } 1544 } else if (mBytesWritten != 0 || 1545 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 1546 memset (mMixBuffer, 0, mixBufferSize); 1547 sleepTime = 0; 1548 LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 1549 } 1550 // TODO add standby time extension fct of effect tail 1551 } 1552 1553 if (mSuspended) { 1554 sleepTime = suspendSleepTimeUs(); 1555 } 1556 // sleepTime == 0 means we must write to audio hardware 1557 if (sleepTime == 0) { 1558 for (size_t i = 0; i < effectChains.size(); i ++) { 1559 effectChains[i]->process_l(); 1560 } 1561 // enable changes in effect chain 1562 unlockEffectChains(effectChains); 1563 mLastWriteTime = systemTime(); 1564 mInWrite = true; 1565 mBytesWritten += mixBufferSize; 1566 1567 int bytesWritten = (int)mOutput->write(mOutput, mMixBuffer, mixBufferSize); 1568 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 1569 mNumWrites++; 1570 mInWrite = false; 1571 nsecs_t now = systemTime(); 1572 nsecs_t delta = now - mLastWriteTime; 1573 if (delta > maxPeriod) { 1574 mNumDelayedWrites++; 1575 if ((now - lastWarning) > kWarningThrottle) { 1576 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 1577 ns2ms(delta), mNumDelayedWrites, this); 1578 lastWarning = now; 1579 } 1580 if (mStandby) { 1581 longStandbyExit = true; 1582 } 1583 } 1584 mStandby = false; 1585 } else { 1586 // enable changes in effect chain 1587 unlockEffectChains(effectChains); 1588 usleep(sleepTime); 1589 } 1590 1591 // finally let go of all our tracks, without the lock held 1592 // since we can't guarantee the destructors won't acquire that 1593 // same lock. 1594 tracksToRemove.clear(); 1595 1596 // Effect chains will be actually deleted here if they were removed from 1597 // mEffectChains list during mixing or effects processing 1598 effectChains.clear(); 1599 } 1600 1601 if (!mStandby) { 1602 mOutput->common.standby(&mOutput->common); 1603 } 1604 1605 LOGV("MixerThread %p exiting", this); 1606 return false; 1607} 1608 1609// prepareTracks_l() must be called with ThreadBase::mLock held 1610uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 1611{ 1612 1613 uint32_t mixerStatus = MIXER_IDLE; 1614 // find out which tracks need to be processed 1615 size_t count = activeTracks.size(); 1616 size_t mixedTracks = 0; 1617 size_t tracksWithEffect = 0; 1618 1619 float masterVolume = mMasterVolume; 1620 bool masterMute = mMasterMute; 1621 1622 if (masterMute) { 1623 masterVolume = 0; 1624 } 1625 // Delegate master volume control to effect in output mix effect chain if needed 1626 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 1627 if (chain != 0) { 1628 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 1629 chain->setVolume_l(&v, &v); 1630 masterVolume = (float)((v + (1 << 23)) >> 24); 1631 chain.clear(); 1632 } 1633 1634 for (size_t i=0 ; i<count ; i++) { 1635 sp<Track> t = activeTracks[i].promote(); 1636 if (t == 0) continue; 1637 1638 Track* const track = t.get(); 1639 audio_track_cblk_t* cblk = track->cblk(); 1640 1641 // The first time a track is added we wait 1642 // for all its buffers to be filled before processing it 1643 mAudioMixer->setActiveTrack(track->name()); 1644 if (cblk->framesReady() && track->isReady() && 1645 !track->isPaused() && !track->isTerminated()) 1646 { 1647 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 1648 1649 mixedTracks++; 1650 1651 // track->mainBuffer() != mMixBuffer means there is an effect chain 1652 // connected to the track 1653 chain.clear(); 1654 if (track->mainBuffer() != mMixBuffer) { 1655 chain = getEffectChain_l(track->sessionId()); 1656 // Delegate volume control to effect in track effect chain if needed 1657 if (chain != 0) { 1658 tracksWithEffect++; 1659 } else { 1660 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 1661 track->name(), track->sessionId()); 1662 } 1663 } 1664 1665 1666 int param = AudioMixer::VOLUME; 1667 if (track->mFillingUpStatus == Track::FS_FILLED) { 1668 // no ramp for the first volume setting 1669 track->mFillingUpStatus = Track::FS_ACTIVE; 1670 if (track->mState == TrackBase::RESUMING) { 1671 track->mState = TrackBase::ACTIVE; 1672 param = AudioMixer::RAMP_VOLUME; 1673 } 1674 mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 1675 } else if (cblk->server != 0) { 1676 // If the track is stopped before the first frame was mixed, 1677 // do not apply ramp 1678 param = AudioMixer::RAMP_VOLUME; 1679 } 1680 1681 // compute volume for this track 1682 uint32_t vl, vr, va; 1683 if (track->isMuted() || track->isPausing() || 1684 mStreamTypes[track->type()].mute) { 1685 vl = vr = va = 0; 1686 if (track->isPausing()) { 1687 track->setPaused(); 1688 } 1689 } else { 1690 1691 // read original volumes with volume control 1692 float typeVolume = mStreamTypes[track->type()].volume; 1693 float v = masterVolume * typeVolume; 1694 vl = (uint32_t)(v * cblk->volume[0]) << 12; 1695 vr = (uint32_t)(v * cblk->volume[1]) << 12; 1696 1697 va = (uint32_t)(v * cblk->sendLevel); 1698 } 1699 // Delegate volume control to effect in track effect chain if needed 1700 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1701 // Do not ramp volume if volume is controlled by effect 1702 param = AudioMixer::VOLUME; 1703 track->mHasVolumeController = true; 1704 } else { 1705 // force no volume ramp when volume controller was just disabled or removed 1706 // from effect chain to avoid volume spike 1707 if (track->mHasVolumeController) { 1708 param = AudioMixer::VOLUME; 1709 } 1710 track->mHasVolumeController = false; 1711 } 1712 1713 // Convert volumes from 8.24 to 4.12 format 1714 int16_t left, right, aux; 1715 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1716 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1717 left = int16_t(v_clamped); 1718 v_clamped = (vr + (1 << 11)) >> 12; 1719 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1720 right = int16_t(v_clamped); 1721 1722 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 1723 aux = int16_t(va); 1724 1725 // XXX: these things DON'T need to be done each time 1726 mAudioMixer->setBufferProvider(track); 1727 mAudioMixer->enable(AudioMixer::MIXING); 1728 1729 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1730 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1731 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1732 mAudioMixer->setParameter( 1733 AudioMixer::TRACK, 1734 AudioMixer::FORMAT, (void *)track->format()); 1735 mAudioMixer->setParameter( 1736 AudioMixer::TRACK, 1737 AudioMixer::CHANNEL_COUNT, (void *)track->channelCount()); 1738 mAudioMixer->setParameter( 1739 AudioMixer::RESAMPLE, 1740 AudioMixer::SAMPLE_RATE, 1741 (void *)(cblk->sampleRate)); 1742 mAudioMixer->setParameter( 1743 AudioMixer::TRACK, 1744 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1745 mAudioMixer->setParameter( 1746 AudioMixer::TRACK, 1747 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1748 1749 // reset retry count 1750 track->mRetryCount = kMaxTrackRetries; 1751 mixerStatus = MIXER_TRACKS_READY; 1752 } else { 1753 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1754 if (track->isStopped()) { 1755 track->reset(); 1756 } 1757 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1758 // We have consumed all the buffers of this track. 1759 // Remove it from the list of active tracks. 1760 tracksToRemove->add(track); 1761 } else { 1762 // No buffers for this track. Give it a few chances to 1763 // fill a buffer, then remove it from active list. 1764 if (--(track->mRetryCount) <= 0) { 1765 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1766 tracksToRemove->add(track); 1767 // indicate to client process that the track was disabled because of underrun 1768 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 1769 } else if (mixerStatus != MIXER_TRACKS_READY) { 1770 mixerStatus = MIXER_TRACKS_ENABLED; 1771 } 1772 } 1773 mAudioMixer->disable(AudioMixer::MIXING); 1774 } 1775 } 1776 1777 // remove all the tracks that need to be... 1778 count = tracksToRemove->size(); 1779 if (UNLIKELY(count)) { 1780 for (size_t i=0 ; i<count ; i++) { 1781 const sp<Track>& track = tracksToRemove->itemAt(i); 1782 mActiveTracks.remove(track); 1783 if (track->mainBuffer() != mMixBuffer) { 1784 chain = getEffectChain_l(track->sessionId()); 1785 if (chain != 0) { 1786 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1787 chain->stopTrack(); 1788 } 1789 } 1790 if (track->isTerminated()) { 1791 mTracks.remove(track); 1792 deleteTrackName_l(track->mName); 1793 } 1794 } 1795 } 1796 1797 // mix buffer must be cleared if all tracks are connected to an 1798 // effect chain as in this case the mixer will not write to 1799 // mix buffer and track effects will accumulate into it 1800 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1801 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1802 } 1803 1804 return mixerStatus; 1805} 1806 1807void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1808{ 1809 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1810 this, streamType, mTracks.size()); 1811 Mutex::Autolock _l(mLock); 1812 1813 size_t size = mTracks.size(); 1814 for (size_t i = 0; i < size; i++) { 1815 sp<Track> t = mTracks[i]; 1816 if (t->type() == streamType) { 1817 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 1818 t->mCblk->cv.signal(); 1819 } 1820 } 1821} 1822 1823 1824// getTrackName_l() must be called with ThreadBase::mLock held 1825int AudioFlinger::MixerThread::getTrackName_l() 1826{ 1827 return mAudioMixer->getTrackName(); 1828} 1829 1830// deleteTrackName_l() must be called with ThreadBase::mLock held 1831void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1832{ 1833 LOGV("remove track (%d) and delete from mixer", name); 1834 mAudioMixer->deleteTrackName(name); 1835} 1836 1837// checkForNewParameters_l() must be called with ThreadBase::mLock held 1838bool AudioFlinger::MixerThread::checkForNewParameters_l() 1839{ 1840 bool reconfig = false; 1841 1842 while (!mNewParameters.isEmpty()) { 1843 status_t status = NO_ERROR; 1844 String8 keyValuePair = mNewParameters[0]; 1845 AudioParameter param = AudioParameter(keyValuePair); 1846 int value; 1847 1848 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1849 reconfig = true; 1850 } 1851 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1852 if (value != AUDIO_FORMAT_PCM_16_BIT) { 1853 status = BAD_VALUE; 1854 } else { 1855 reconfig = true; 1856 } 1857 } 1858 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1859 if (value != AUDIO_CHANNEL_OUT_STEREO) { 1860 status = BAD_VALUE; 1861 } else { 1862 reconfig = true; 1863 } 1864 } 1865 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1866 // do not accept frame count changes if tracks are open as the track buffer 1867 // size depends on frame count and correct behavior would not be garantied 1868 // if frame count is changed after track creation 1869 if (!mTracks.isEmpty()) { 1870 status = INVALID_OPERATION; 1871 } else { 1872 reconfig = true; 1873 } 1874 } 1875 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1876 // when changing the audio output device, call addBatteryData to notify 1877 // the change 1878 if (mDevice != value) { 1879 uint32_t params = 0; 1880 // check whether speaker is on 1881 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 1882 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 1883 } 1884 1885 int deviceWithoutSpeaker 1886 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 1887 // check if any other device (except speaker) is on 1888 if (value & deviceWithoutSpeaker ) { 1889 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 1890 } 1891 1892 if (params != 0) { 1893 addBatteryData(params); 1894 } 1895 } 1896 1897 // forward device change to effects that have requested to be 1898 // aware of attached audio device. 1899 mDevice = (uint32_t)value; 1900 for (size_t i = 0; i < mEffectChains.size(); i++) { 1901 mEffectChains[i]->setDevice_l(mDevice); 1902 } 1903 } 1904 1905 if (status == NO_ERROR) { 1906 status = mOutput->common.set_parameters(&mOutput->common, 1907 keyValuePair.string()); 1908 if (!mStandby && status == INVALID_OPERATION) { 1909 mOutput->common.standby(&mOutput->common); 1910 mStandby = true; 1911 mBytesWritten = 0; 1912 status = mOutput->common.set_parameters(&mOutput->common, 1913 keyValuePair.string()); 1914 } 1915 if (status == NO_ERROR && reconfig) { 1916 delete mAudioMixer; 1917 readOutputParameters(); 1918 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1919 for (size_t i = 0; i < mTracks.size() ; i++) { 1920 int name = getTrackName_l(); 1921 if (name < 0) break; 1922 mTracks[i]->mName = name; 1923 // limit track sample rate to 2 x new output sample rate 1924 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 1925 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 1926 } 1927 } 1928 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 1929 } 1930 } 1931 1932 mNewParameters.removeAt(0); 1933 1934 mParamStatus = status; 1935 mParamCond.signal(); 1936 mWaitWorkCV.wait(mLock); 1937 } 1938 return reconfig; 1939} 1940 1941status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 1942{ 1943 const size_t SIZE = 256; 1944 char buffer[SIZE]; 1945 String8 result; 1946 1947 PlaybackThread::dumpInternals(fd, args); 1948 1949 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 1950 result.append(buffer); 1951 write(fd, result.string(), result.size()); 1952 return NO_ERROR; 1953} 1954 1955uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 1956{ 1957 return (uint32_t)(mOutput->get_latency(mOutput) * 1000) / 2; 1958} 1959 1960uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 1961{ 1962 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 1963} 1964 1965uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 1966{ 1967 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 1968} 1969 1970// ---------------------------------------------------------------------------- 1971AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, struct audio_stream_out* output, int id, uint32_t device) 1972 : PlaybackThread(audioFlinger, output, id, device) 1973{ 1974 mType = PlaybackThread::DIRECT; 1975} 1976 1977AudioFlinger::DirectOutputThread::~DirectOutputThread() 1978{ 1979} 1980 1981 1982static inline int16_t clamp16(int32_t sample) 1983{ 1984 if ((sample>>15) ^ (sample>>31)) 1985 sample = 0x7FFF ^ (sample>>31); 1986 return sample; 1987} 1988 1989static inline 1990int32_t mul(int16_t in, int16_t v) 1991{ 1992#if defined(__arm__) && !defined(__thumb__) 1993 int32_t out; 1994 asm( "smulbb %[out], %[in], %[v] \n" 1995 : [out]"=r"(out) 1996 : [in]"%r"(in), [v]"r"(v) 1997 : ); 1998 return out; 1999#else 2000 return in * int32_t(v); 2001#endif 2002} 2003 2004void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2005{ 2006 // Do not apply volume on compressed audio 2007 if (!audio_is_linear_pcm(mFormat)) { 2008 return; 2009 } 2010 2011 // convert to signed 16 bit before volume calculation 2012 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2013 size_t count = mFrameCount * mChannelCount; 2014 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2015 int16_t *dst = mMixBuffer + count-1; 2016 while(count--) { 2017 *dst-- = (int16_t)(*src--^0x80) << 8; 2018 } 2019 } 2020 2021 size_t frameCount = mFrameCount; 2022 int16_t *out = mMixBuffer; 2023 if (ramp) { 2024 if (mChannelCount == 1) { 2025 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2026 int32_t vlInc = d / (int32_t)frameCount; 2027 int32_t vl = ((int32_t)mLeftVolShort << 16); 2028 do { 2029 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2030 out++; 2031 vl += vlInc; 2032 } while (--frameCount); 2033 2034 } else { 2035 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2036 int32_t vlInc = d / (int32_t)frameCount; 2037 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2038 int32_t vrInc = d / (int32_t)frameCount; 2039 int32_t vl = ((int32_t)mLeftVolShort << 16); 2040 int32_t vr = ((int32_t)mRightVolShort << 16); 2041 do { 2042 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2043 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2044 out += 2; 2045 vl += vlInc; 2046 vr += vrInc; 2047 } while (--frameCount); 2048 } 2049 } else { 2050 if (mChannelCount == 1) { 2051 do { 2052 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2053 out++; 2054 } while (--frameCount); 2055 } else { 2056 do { 2057 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2058 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2059 out += 2; 2060 } while (--frameCount); 2061 } 2062 } 2063 2064 // convert back to unsigned 8 bit after volume calculation 2065 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2066 size_t count = mFrameCount * mChannelCount; 2067 int16_t *src = mMixBuffer; 2068 uint8_t *dst = (uint8_t *)mMixBuffer; 2069 while(count--) { 2070 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2071 } 2072 } 2073 2074 mLeftVolShort = leftVol; 2075 mRightVolShort = rightVol; 2076} 2077 2078bool AudioFlinger::DirectOutputThread::threadLoop() 2079{ 2080 uint32_t mixerStatus = MIXER_IDLE; 2081 sp<Track> trackToRemove; 2082 sp<Track> activeTrack; 2083 nsecs_t standbyTime = systemTime(); 2084 int8_t *curBuf; 2085 size_t mixBufferSize = mFrameCount*mFrameSize; 2086 uint32_t activeSleepTime = activeSleepTimeUs(); 2087 uint32_t idleSleepTime = idleSleepTimeUs(); 2088 uint32_t sleepTime = idleSleepTime; 2089 // use shorter standby delay as on normal output to release 2090 // hardware resources as soon as possible 2091 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2092 2093 while (!exitPending()) 2094 { 2095 bool rampVolume; 2096 uint16_t leftVol; 2097 uint16_t rightVol; 2098 Vector< sp<EffectChain> > effectChains; 2099 2100 processConfigEvents(); 2101 2102 mixerStatus = MIXER_IDLE; 2103 2104 { // scope for the mLock 2105 2106 Mutex::Autolock _l(mLock); 2107 2108 if (checkForNewParameters_l()) { 2109 mixBufferSize = mFrameCount*mFrameSize; 2110 activeSleepTime = activeSleepTimeUs(); 2111 idleSleepTime = idleSleepTimeUs(); 2112 standbyDelay = microseconds(activeSleepTime*2); 2113 } 2114 2115 // put audio hardware into standby after short delay 2116 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2117 mSuspended) { 2118 // wait until we have something to do... 2119 if (!mStandby) { 2120 LOGV("Audio hardware entering standby, mixer %p\n", this); 2121 mOutput->common.standby(&mOutput->common); 2122 mStandby = true; 2123 mBytesWritten = 0; 2124 } 2125 2126 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2127 // we're about to wait, flush the binder command buffer 2128 IPCThreadState::self()->flushCommands(); 2129 2130 if (exitPending()) break; 2131 2132 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2133 mWaitWorkCV.wait(mLock); 2134 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2135 2136 if (mMasterMute == false) { 2137 char value[PROPERTY_VALUE_MAX]; 2138 property_get("ro.audio.silent", value, "0"); 2139 if (atoi(value)) { 2140 LOGD("Silence is golden"); 2141 setMasterMute(true); 2142 } 2143 } 2144 2145 standbyTime = systemTime() + standbyDelay; 2146 sleepTime = idleSleepTime; 2147 continue; 2148 } 2149 } 2150 2151 effectChains = mEffectChains; 2152 2153 // find out which tracks need to be processed 2154 if (mActiveTracks.size() != 0) { 2155 sp<Track> t = mActiveTracks[0].promote(); 2156 if (t == 0) continue; 2157 2158 Track* const track = t.get(); 2159 audio_track_cblk_t* cblk = track->cblk(); 2160 2161 // The first time a track is added we wait 2162 // for all its buffers to be filled before processing it 2163 if (cblk->framesReady() && track->isReady() && 2164 !track->isPaused() && !track->isTerminated()) 2165 { 2166 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2167 2168 if (track->mFillingUpStatus == Track::FS_FILLED) { 2169 track->mFillingUpStatus = Track::FS_ACTIVE; 2170 mLeftVolFloat = mRightVolFloat = 0; 2171 mLeftVolShort = mRightVolShort = 0; 2172 if (track->mState == TrackBase::RESUMING) { 2173 track->mState = TrackBase::ACTIVE; 2174 rampVolume = true; 2175 } 2176 } else if (cblk->server != 0) { 2177 // If the track is stopped before the first frame was mixed, 2178 // do not apply ramp 2179 rampVolume = true; 2180 } 2181 // compute volume for this track 2182 float left, right; 2183 if (track->isMuted() || mMasterMute || track->isPausing() || 2184 mStreamTypes[track->type()].mute) { 2185 left = right = 0; 2186 if (track->isPausing()) { 2187 track->setPaused(); 2188 } 2189 } else { 2190 float typeVolume = mStreamTypes[track->type()].volume; 2191 float v = mMasterVolume * typeVolume; 2192 float v_clamped = v * cblk->volume[0]; 2193 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2194 left = v_clamped/MAX_GAIN; 2195 v_clamped = v * cblk->volume[1]; 2196 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2197 right = v_clamped/MAX_GAIN; 2198 } 2199 2200 if (left != mLeftVolFloat || right != mRightVolFloat) { 2201 mLeftVolFloat = left; 2202 mRightVolFloat = right; 2203 2204 // If audio HAL implements volume control, 2205 // force software volume to nominal value 2206 if (mOutput->set_volume(mOutput, left, right) == NO_ERROR) { 2207 left = 1.0f; 2208 right = 1.0f; 2209 } 2210 2211 // Convert volumes from float to 8.24 2212 uint32_t vl = (uint32_t)(left * (1 << 24)); 2213 uint32_t vr = (uint32_t)(right * (1 << 24)); 2214 2215 // Delegate volume control to effect in track effect chain if needed 2216 // only one effect chain can be present on DirectOutputThread, so if 2217 // there is one, the track is connected to it 2218 if (!effectChains.isEmpty()) { 2219 // Do not ramp volume if volume is controlled by effect 2220 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2221 rampVolume = false; 2222 } 2223 } 2224 2225 // Convert volumes from 8.24 to 4.12 format 2226 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2227 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2228 leftVol = (uint16_t)v_clamped; 2229 v_clamped = (vr + (1 << 11)) >> 12; 2230 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2231 rightVol = (uint16_t)v_clamped; 2232 } else { 2233 leftVol = mLeftVolShort; 2234 rightVol = mRightVolShort; 2235 rampVolume = false; 2236 } 2237 2238 // reset retry count 2239 track->mRetryCount = kMaxTrackRetriesDirect; 2240 activeTrack = t; 2241 mixerStatus = MIXER_TRACKS_READY; 2242 } else { 2243 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2244 if (track->isStopped()) { 2245 track->reset(); 2246 } 2247 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2248 // We have consumed all the buffers of this track. 2249 // Remove it from the list of active tracks. 2250 trackToRemove = track; 2251 } else { 2252 // No buffers for this track. Give it a few chances to 2253 // fill a buffer, then remove it from active list. 2254 if (--(track->mRetryCount) <= 0) { 2255 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2256 trackToRemove = track; 2257 } else { 2258 mixerStatus = MIXER_TRACKS_ENABLED; 2259 } 2260 } 2261 } 2262 } 2263 2264 // remove all the tracks that need to be... 2265 if (UNLIKELY(trackToRemove != 0)) { 2266 mActiveTracks.remove(trackToRemove); 2267 if (!effectChains.isEmpty()) { 2268 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2269 trackToRemove->sessionId()); 2270 effectChains[0]->stopTrack(); 2271 } 2272 if (trackToRemove->isTerminated()) { 2273 mTracks.remove(trackToRemove); 2274 deleteTrackName_l(trackToRemove->mName); 2275 } 2276 } 2277 2278 lockEffectChains_l(effectChains); 2279 } 2280 2281 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2282 AudioBufferProvider::Buffer buffer; 2283 size_t frameCount = mFrameCount; 2284 curBuf = (int8_t *)mMixBuffer; 2285 // output audio to hardware 2286 while (frameCount) { 2287 buffer.frameCount = frameCount; 2288 activeTrack->getNextBuffer(&buffer); 2289 if (UNLIKELY(buffer.raw == 0)) { 2290 memset(curBuf, 0, frameCount * mFrameSize); 2291 break; 2292 } 2293 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2294 frameCount -= buffer.frameCount; 2295 curBuf += buffer.frameCount * mFrameSize; 2296 activeTrack->releaseBuffer(&buffer); 2297 } 2298 sleepTime = 0; 2299 standbyTime = systemTime() + standbyDelay; 2300 } else { 2301 if (sleepTime == 0) { 2302 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2303 sleepTime = activeSleepTime; 2304 } else { 2305 sleepTime = idleSleepTime; 2306 } 2307 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2308 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2309 sleepTime = 0; 2310 } 2311 } 2312 2313 if (mSuspended) { 2314 sleepTime = suspendSleepTimeUs(); 2315 } 2316 // sleepTime == 0 means we must write to audio hardware 2317 if (sleepTime == 0) { 2318 if (mixerStatus == MIXER_TRACKS_READY) { 2319 applyVolume(leftVol, rightVol, rampVolume); 2320 } 2321 for (size_t i = 0; i < effectChains.size(); i ++) { 2322 effectChains[i]->process_l(); 2323 } 2324 unlockEffectChains(effectChains); 2325 2326 mLastWriteTime = systemTime(); 2327 mInWrite = true; 2328 mBytesWritten += mixBufferSize; 2329 int bytesWritten = (int)mOutput->write(mOutput, mMixBuffer, mixBufferSize); 2330 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2331 mNumWrites++; 2332 mInWrite = false; 2333 mStandby = false; 2334 } else { 2335 unlockEffectChains(effectChains); 2336 usleep(sleepTime); 2337 } 2338 2339 // finally let go of removed track, without the lock held 2340 // since we can't guarantee the destructors won't acquire that 2341 // same lock. 2342 trackToRemove.clear(); 2343 activeTrack.clear(); 2344 2345 // Effect chains will be actually deleted here if they were removed from 2346 // mEffectChains list during mixing or effects processing 2347 effectChains.clear(); 2348 } 2349 2350 if (!mStandby) { 2351 mOutput->common.standby(&mOutput->common); 2352 } 2353 2354 LOGV("DirectOutputThread %p exiting", this); 2355 return false; 2356} 2357 2358// getTrackName_l() must be called with ThreadBase::mLock held 2359int AudioFlinger::DirectOutputThread::getTrackName_l() 2360{ 2361 return 0; 2362} 2363 2364// deleteTrackName_l() must be called with ThreadBase::mLock held 2365void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2366{ 2367} 2368 2369// checkForNewParameters_l() must be called with ThreadBase::mLock held 2370bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2371{ 2372 bool reconfig = false; 2373 2374 while (!mNewParameters.isEmpty()) { 2375 status_t status = NO_ERROR; 2376 String8 keyValuePair = mNewParameters[0]; 2377 AudioParameter param = AudioParameter(keyValuePair); 2378 int value; 2379 2380 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2381 // do not accept frame count changes if tracks are open as the track buffer 2382 // size depends on frame count and correct behavior would not be garantied 2383 // if frame count is changed after track creation 2384 if (!mTracks.isEmpty()) { 2385 status = INVALID_OPERATION; 2386 } else { 2387 reconfig = true; 2388 } 2389 } 2390 if (status == NO_ERROR) { 2391 status = mOutput->common.set_parameters(&mOutput->common, 2392 keyValuePair.string()); 2393 if (!mStandby && status == INVALID_OPERATION) { 2394 mOutput->common.standby(&mOutput->common); 2395 mStandby = true; 2396 mBytesWritten = 0; 2397 status = mOutput->common.set_parameters(&mOutput->common, 2398 keyValuePair.string()); 2399 } 2400 if (status == NO_ERROR && reconfig) { 2401 readOutputParameters(); 2402 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2403 } 2404 } 2405 2406 mNewParameters.removeAt(0); 2407 2408 mParamStatus = status; 2409 mParamCond.signal(); 2410 mWaitWorkCV.wait(mLock); 2411 } 2412 return reconfig; 2413} 2414 2415uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2416{ 2417 uint32_t time; 2418 if (audio_is_linear_pcm(mFormat)) { 2419 time = (uint32_t)(mOutput->get_latency(mOutput) * 1000) / 2; 2420 } else { 2421 time = 10000; 2422 } 2423 return time; 2424} 2425 2426uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2427{ 2428 uint32_t time; 2429 if (audio_is_linear_pcm(mFormat)) { 2430 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2431 } else { 2432 time = 10000; 2433 } 2434 return time; 2435} 2436 2437uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2438{ 2439 uint32_t time; 2440 if (audio_is_linear_pcm(mFormat)) { 2441 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2442 } else { 2443 time = 10000; 2444 } 2445 return time; 2446} 2447 2448 2449// ---------------------------------------------------------------------------- 2450 2451AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2452 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2453{ 2454 mType = PlaybackThread::DUPLICATING; 2455 addOutputTrack(mainThread); 2456} 2457 2458AudioFlinger::DuplicatingThread::~DuplicatingThread() 2459{ 2460 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2461 mOutputTracks[i]->destroy(); 2462 } 2463 mOutputTracks.clear(); 2464} 2465 2466bool AudioFlinger::DuplicatingThread::threadLoop() 2467{ 2468 Vector< sp<Track> > tracksToRemove; 2469 uint32_t mixerStatus = MIXER_IDLE; 2470 nsecs_t standbyTime = systemTime(); 2471 size_t mixBufferSize = mFrameCount*mFrameSize; 2472 SortedVector< sp<OutputTrack> > outputTracks; 2473 uint32_t writeFrames = 0; 2474 uint32_t activeSleepTime = activeSleepTimeUs(); 2475 uint32_t idleSleepTime = idleSleepTimeUs(); 2476 uint32_t sleepTime = idleSleepTime; 2477 Vector< sp<EffectChain> > effectChains; 2478 2479 while (!exitPending()) 2480 { 2481 processConfigEvents(); 2482 2483 mixerStatus = MIXER_IDLE; 2484 { // scope for the mLock 2485 2486 Mutex::Autolock _l(mLock); 2487 2488 if (checkForNewParameters_l()) { 2489 mixBufferSize = mFrameCount*mFrameSize; 2490 updateWaitTime(); 2491 activeSleepTime = activeSleepTimeUs(); 2492 idleSleepTime = idleSleepTimeUs(); 2493 } 2494 2495 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2496 2497 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2498 outputTracks.add(mOutputTracks[i]); 2499 } 2500 2501 // put audio hardware into standby after short delay 2502 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2503 mSuspended) { 2504 if (!mStandby) { 2505 for (size_t i = 0; i < outputTracks.size(); i++) { 2506 outputTracks[i]->stop(); 2507 } 2508 mStandby = true; 2509 mBytesWritten = 0; 2510 } 2511 2512 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2513 // we're about to wait, flush the binder command buffer 2514 IPCThreadState::self()->flushCommands(); 2515 outputTracks.clear(); 2516 2517 if (exitPending()) break; 2518 2519 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2520 mWaitWorkCV.wait(mLock); 2521 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2522 if (mMasterMute == false) { 2523 char value[PROPERTY_VALUE_MAX]; 2524 property_get("ro.audio.silent", value, "0"); 2525 if (atoi(value)) { 2526 LOGD("Silence is golden"); 2527 setMasterMute(true); 2528 } 2529 } 2530 2531 standbyTime = systemTime() + kStandbyTimeInNsecs; 2532 sleepTime = idleSleepTime; 2533 continue; 2534 } 2535 } 2536 2537 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2538 2539 // prevent any changes in effect chain list and in each effect chain 2540 // during mixing and effect process as the audio buffers could be deleted 2541 // or modified if an effect is created or deleted 2542 lockEffectChains_l(effectChains); 2543 } 2544 2545 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2546 // mix buffers... 2547 if (outputsReady(outputTracks)) { 2548 mAudioMixer->process(); 2549 } else { 2550 memset(mMixBuffer, 0, mixBufferSize); 2551 } 2552 sleepTime = 0; 2553 writeFrames = mFrameCount; 2554 } else { 2555 if (sleepTime == 0) { 2556 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2557 sleepTime = activeSleepTime; 2558 } else { 2559 sleepTime = idleSleepTime; 2560 } 2561 } else if (mBytesWritten != 0) { 2562 // flush remaining overflow buffers in output tracks 2563 for (size_t i = 0; i < outputTracks.size(); i++) { 2564 if (outputTracks[i]->isActive()) { 2565 sleepTime = 0; 2566 writeFrames = 0; 2567 memset(mMixBuffer, 0, mixBufferSize); 2568 break; 2569 } 2570 } 2571 } 2572 } 2573 2574 if (mSuspended) { 2575 sleepTime = suspendSleepTimeUs(); 2576 } 2577 // sleepTime == 0 means we must write to audio hardware 2578 if (sleepTime == 0) { 2579 for (size_t i = 0; i < effectChains.size(); i ++) { 2580 effectChains[i]->process_l(); 2581 } 2582 // enable changes in effect chain 2583 unlockEffectChains(effectChains); 2584 2585 standbyTime = systemTime() + kStandbyTimeInNsecs; 2586 for (size_t i = 0; i < outputTracks.size(); i++) { 2587 outputTracks[i]->write(mMixBuffer, writeFrames); 2588 } 2589 mStandby = false; 2590 mBytesWritten += mixBufferSize; 2591 } else { 2592 // enable changes in effect chain 2593 unlockEffectChains(effectChains); 2594 usleep(sleepTime); 2595 } 2596 2597 // finally let go of all our tracks, without the lock held 2598 // since we can't guarantee the destructors won't acquire that 2599 // same lock. 2600 tracksToRemove.clear(); 2601 outputTracks.clear(); 2602 2603 // Effect chains will be actually deleted here if they were removed from 2604 // mEffectChains list during mixing or effects processing 2605 effectChains.clear(); 2606 } 2607 2608 return false; 2609} 2610 2611void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2612{ 2613 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2614 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2615 this, 2616 mSampleRate, 2617 mFormat, 2618 mChannelCount, 2619 frameCount); 2620 if (outputTrack->cblk() != NULL) { 2621 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 2622 mOutputTracks.add(outputTrack); 2623 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2624 updateWaitTime(); 2625 } 2626} 2627 2628void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2629{ 2630 Mutex::Autolock _l(mLock); 2631 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2632 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2633 mOutputTracks[i]->destroy(); 2634 mOutputTracks.removeAt(i); 2635 updateWaitTime(); 2636 return; 2637 } 2638 } 2639 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2640} 2641 2642void AudioFlinger::DuplicatingThread::updateWaitTime() 2643{ 2644 mWaitTimeMs = UINT_MAX; 2645 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2646 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2647 if (strong != NULL) { 2648 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2649 if (waitTimeMs < mWaitTimeMs) { 2650 mWaitTimeMs = waitTimeMs; 2651 } 2652 } 2653 } 2654} 2655 2656 2657bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2658{ 2659 for (size_t i = 0; i < outputTracks.size(); i++) { 2660 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2661 if (thread == 0) { 2662 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2663 return false; 2664 } 2665 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2666 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2667 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2668 return false; 2669 } 2670 } 2671 return true; 2672} 2673 2674uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2675{ 2676 return (mWaitTimeMs * 1000) / 2; 2677} 2678 2679// ---------------------------------------------------------------------------- 2680 2681// TrackBase constructor must be called with AudioFlinger::mLock held 2682AudioFlinger::ThreadBase::TrackBase::TrackBase( 2683 const wp<ThreadBase>& thread, 2684 const sp<Client>& client, 2685 uint32_t sampleRate, 2686 int format, 2687 int channelCount, 2688 int frameCount, 2689 uint32_t flags, 2690 const sp<IMemory>& sharedBuffer, 2691 int sessionId) 2692 : RefBase(), 2693 mThread(thread), 2694 mClient(client), 2695 mCblk(0), 2696 mFrameCount(0), 2697 mState(IDLE), 2698 mClientTid(-1), 2699 mFormat(format), 2700 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2701 mSessionId(sessionId) 2702{ 2703 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2704 2705 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2706 size_t size = sizeof(audio_track_cblk_t); 2707 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2708 if (sharedBuffer == 0) { 2709 size += bufferSize; 2710 } 2711 2712 if (client != NULL) { 2713 mCblkMemory = client->heap()->allocate(size); 2714 if (mCblkMemory != 0) { 2715 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2716 if (mCblk) { // construct the shared structure in-place. 2717 new(mCblk) audio_track_cblk_t(); 2718 // clear all buffers 2719 mCblk->frameCount = frameCount; 2720 mCblk->sampleRate = sampleRate; 2721 mCblk->channelCount = (uint8_t)channelCount; 2722 if (sharedBuffer == 0) { 2723 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2724 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2725 // Force underrun condition to avoid false underrun callback until first data is 2726 // written to buffer (other flags are cleared) 2727 mCblk->flags = CBLK_UNDERRUN_ON; 2728 } else { 2729 mBuffer = sharedBuffer->pointer(); 2730 } 2731 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2732 } 2733 } else { 2734 LOGE("not enough memory for AudioTrack size=%u", size); 2735 client->heap()->dump("AudioTrack"); 2736 return; 2737 } 2738 } else { 2739 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2740 if (mCblk) { // construct the shared structure in-place. 2741 new(mCblk) audio_track_cblk_t(); 2742 // clear all buffers 2743 mCblk->frameCount = frameCount; 2744 mCblk->sampleRate = sampleRate; 2745 mCblk->channelCount = (uint8_t)channelCount; 2746 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2747 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2748 // Force underrun condition to avoid false underrun callback until first data is 2749 // written to buffer (other flags are cleared) 2750 mCblk->flags = CBLK_UNDERRUN_ON; 2751 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2752 } 2753 } 2754} 2755 2756AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2757{ 2758 if (mCblk) { 2759 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2760 if (mClient == NULL) { 2761 delete mCblk; 2762 } 2763 } 2764 mCblkMemory.clear(); // and free the shared memory 2765 if (mClient != NULL) { 2766 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2767 mClient.clear(); 2768 } 2769} 2770 2771void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2772{ 2773 buffer->raw = 0; 2774 mFrameCount = buffer->frameCount; 2775 step(); 2776 buffer->frameCount = 0; 2777} 2778 2779bool AudioFlinger::ThreadBase::TrackBase::step() { 2780 bool result; 2781 audio_track_cblk_t* cblk = this->cblk(); 2782 2783 result = cblk->stepServer(mFrameCount); 2784 if (!result) { 2785 LOGV("stepServer failed acquiring cblk mutex"); 2786 mFlags |= STEPSERVER_FAILED; 2787 } 2788 return result; 2789} 2790 2791void AudioFlinger::ThreadBase::TrackBase::reset() { 2792 audio_track_cblk_t* cblk = this->cblk(); 2793 2794 cblk->user = 0; 2795 cblk->server = 0; 2796 cblk->userBase = 0; 2797 cblk->serverBase = 0; 2798 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2799 LOGV("TrackBase::reset"); 2800} 2801 2802sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2803{ 2804 return mCblkMemory; 2805} 2806 2807int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2808 return (int)mCblk->sampleRate; 2809} 2810 2811int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2812 return (int)mCblk->channelCount; 2813} 2814 2815void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2816 audio_track_cblk_t* cblk = this->cblk(); 2817 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2818 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2819 2820 // Check validity of returned pointer in case the track control block would have been corrupted. 2821 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2822 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2823 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2824 server %d, serverBase %d, user %d, userBase %d, channelCount %d", 2825 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2826 cblk->server, cblk->serverBase, cblk->user, cblk->userBase, cblk->channelCount); 2827 return 0; 2828 } 2829 2830 return bufferStart; 2831} 2832 2833// ---------------------------------------------------------------------------- 2834 2835// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2836AudioFlinger::PlaybackThread::Track::Track( 2837 const wp<ThreadBase>& thread, 2838 const sp<Client>& client, 2839 int streamType, 2840 uint32_t sampleRate, 2841 int format, 2842 int channelCount, 2843 int frameCount, 2844 const sp<IMemory>& sharedBuffer, 2845 int sessionId) 2846 : TrackBase(thread, client, sampleRate, format, channelCount, frameCount, 0, sharedBuffer, sessionId), 2847 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 2848 mAuxEffectId(0), mHasVolumeController(false) 2849{ 2850 if (mCblk != NULL) { 2851 sp<ThreadBase> baseThread = thread.promote(); 2852 if (baseThread != 0) { 2853 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2854 mName = playbackThread->getTrackName_l(); 2855 mMainBuffer = playbackThread->mixBuffer(); 2856 } 2857 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2858 if (mName < 0) { 2859 LOGE("no more track names available"); 2860 } 2861 mVolume[0] = 1.0f; 2862 mVolume[1] = 1.0f; 2863 mStreamType = streamType; 2864 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2865 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2866 mCblk->frameSize = audio_is_linear_pcm(format) ? channelCount * sizeof(int16_t) : sizeof(int8_t); 2867 } 2868} 2869 2870AudioFlinger::PlaybackThread::Track::~Track() 2871{ 2872 LOGV("PlaybackThread::Track destructor"); 2873 sp<ThreadBase> thread = mThread.promote(); 2874 if (thread != 0) { 2875 Mutex::Autolock _l(thread->mLock); 2876 mState = TERMINATED; 2877 } 2878} 2879 2880void AudioFlinger::PlaybackThread::Track::destroy() 2881{ 2882 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2883 // by removing it from mTracks vector, so there is a risk that this Tracks's 2884 // desctructor is called. As the destructor needs to lock mLock, 2885 // we must acquire a strong reference on this Track before locking mLock 2886 // here so that the destructor is called only when exiting this function. 2887 // On the other hand, as long as Track::destroy() is only called by 2888 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2889 // this Track with its member mTrack. 2890 sp<Track> keep(this); 2891 { // scope for mLock 2892 sp<ThreadBase> thread = mThread.promote(); 2893 if (thread != 0) { 2894 if (!isOutputTrack()) { 2895 if (mState == ACTIVE || mState == RESUMING) { 2896 AudioSystem::stopOutput(thread->id(), 2897 (audio_stream_type_t)mStreamType, 2898 mSessionId); 2899 2900 // to track the speaker usage 2901 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2902 } 2903 AudioSystem::releaseOutput(thread->id()); 2904 } 2905 Mutex::Autolock _l(thread->mLock); 2906 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2907 playbackThread->destroyTrack_l(this); 2908 } 2909 } 2910} 2911 2912void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 2913{ 2914 snprintf(buffer, size, " %05d %05d %03u %03u %03u %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 2915 mName - AudioMixer::TRACK0, 2916 (mClient == NULL) ? getpid() : mClient->pid(), 2917 mStreamType, 2918 mFormat, 2919 mCblk->channelCount, 2920 mSessionId, 2921 mFrameCount, 2922 mState, 2923 mMute, 2924 mFillingUpStatus, 2925 mCblk->sampleRate, 2926 mCblk->volume[0], 2927 mCblk->volume[1], 2928 mCblk->server, 2929 mCblk->user, 2930 (int)mMainBuffer, 2931 (int)mAuxBuffer); 2932} 2933 2934status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 2935{ 2936 audio_track_cblk_t* cblk = this->cblk(); 2937 uint32_t framesReady; 2938 uint32_t framesReq = buffer->frameCount; 2939 2940 // Check if last stepServer failed, try to step now 2941 if (mFlags & TrackBase::STEPSERVER_FAILED) { 2942 if (!step()) goto getNextBuffer_exit; 2943 LOGV("stepServer recovered"); 2944 mFlags &= ~TrackBase::STEPSERVER_FAILED; 2945 } 2946 2947 framesReady = cblk->framesReady(); 2948 2949 if (LIKELY(framesReady)) { 2950 uint32_t s = cblk->server; 2951 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 2952 2953 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 2954 if (framesReq > framesReady) { 2955 framesReq = framesReady; 2956 } 2957 if (s + framesReq > bufferEnd) { 2958 framesReq = bufferEnd - s; 2959 } 2960 2961 buffer->raw = getBuffer(s, framesReq); 2962 if (buffer->raw == 0) goto getNextBuffer_exit; 2963 2964 buffer->frameCount = framesReq; 2965 return NO_ERROR; 2966 } 2967 2968getNextBuffer_exit: 2969 buffer->raw = 0; 2970 buffer->frameCount = 0; 2971 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 2972 return NOT_ENOUGH_DATA; 2973} 2974 2975bool AudioFlinger::PlaybackThread::Track::isReady() const { 2976 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 2977 2978 if (mCblk->framesReady() >= mCblk->frameCount || 2979 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 2980 mFillingUpStatus = FS_FILLED; 2981 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 2982 return true; 2983 } 2984 return false; 2985} 2986 2987status_t AudioFlinger::PlaybackThread::Track::start() 2988{ 2989 status_t status = NO_ERROR; 2990 LOGV("start(%d), calling thread %d session %d", 2991 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 2992 sp<ThreadBase> thread = mThread.promote(); 2993 if (thread != 0) { 2994 Mutex::Autolock _l(thread->mLock); 2995 int state = mState; 2996 // here the track could be either new, or restarted 2997 // in both cases "unstop" the track 2998 if (mState == PAUSED) { 2999 mState = TrackBase::RESUMING; 3000 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3001 } else { 3002 mState = TrackBase::ACTIVE; 3003 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3004 } 3005 3006 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3007 thread->mLock.unlock(); 3008 status = AudioSystem::startOutput(thread->id(), 3009 (audio_stream_type_t)mStreamType, 3010 mSessionId); 3011 thread->mLock.lock(); 3012 3013 // to track the speaker usage 3014 if (status == NO_ERROR) { 3015 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3016 } 3017 } 3018 if (status == NO_ERROR) { 3019 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3020 playbackThread->addTrack_l(this); 3021 } else { 3022 mState = state; 3023 } 3024 } else { 3025 status = BAD_VALUE; 3026 } 3027 return status; 3028} 3029 3030void AudioFlinger::PlaybackThread::Track::stop() 3031{ 3032 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3033 sp<ThreadBase> thread = mThread.promote(); 3034 if (thread != 0) { 3035 Mutex::Autolock _l(thread->mLock); 3036 int state = mState; 3037 if (mState > STOPPED) { 3038 mState = STOPPED; 3039 // If the track is not active (PAUSED and buffers full), flush buffers 3040 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3041 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3042 reset(); 3043 } 3044 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3045 } 3046 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3047 thread->mLock.unlock(); 3048 AudioSystem::stopOutput(thread->id(), 3049 (audio_stream_type_t)mStreamType, 3050 mSessionId); 3051 thread->mLock.lock(); 3052 3053 // to track the speaker usage 3054 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3055 } 3056 } 3057} 3058 3059void AudioFlinger::PlaybackThread::Track::pause() 3060{ 3061 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3062 sp<ThreadBase> thread = mThread.promote(); 3063 if (thread != 0) { 3064 Mutex::Autolock _l(thread->mLock); 3065 if (mState == ACTIVE || mState == RESUMING) { 3066 mState = PAUSING; 3067 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3068 if (!isOutputTrack()) { 3069 thread->mLock.unlock(); 3070 AudioSystem::stopOutput(thread->id(), 3071 (audio_stream_type_t)mStreamType, 3072 mSessionId); 3073 thread->mLock.lock(); 3074 3075 // to track the speaker usage 3076 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3077 } 3078 } 3079 } 3080} 3081 3082void AudioFlinger::PlaybackThread::Track::flush() 3083{ 3084 LOGV("flush(%d)", mName); 3085 sp<ThreadBase> thread = mThread.promote(); 3086 if (thread != 0) { 3087 Mutex::Autolock _l(thread->mLock); 3088 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3089 return; 3090 } 3091 // No point remaining in PAUSED state after a flush => go to 3092 // STOPPED state 3093 mState = STOPPED; 3094 3095 // do not reset the track if it is still in the process of being stopped or paused. 3096 // this will be done by prepareTracks_l() when the track is stopped. 3097 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3098 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3099 reset(); 3100 } 3101 } 3102} 3103 3104void AudioFlinger::PlaybackThread::Track::reset() 3105{ 3106 // Do not reset twice to avoid discarding data written just after a flush and before 3107 // the audioflinger thread detects the track is stopped. 3108 if (!mResetDone) { 3109 TrackBase::reset(); 3110 // Force underrun condition to avoid false underrun callback until first data is 3111 // written to buffer 3112 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3113 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3114 mFillingUpStatus = FS_FILLING; 3115 mResetDone = true; 3116 } 3117} 3118 3119void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3120{ 3121 mMute = muted; 3122} 3123 3124void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3125{ 3126 mVolume[0] = left; 3127 mVolume[1] = right; 3128} 3129 3130status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3131{ 3132 status_t status = DEAD_OBJECT; 3133 sp<ThreadBase> thread = mThread.promote(); 3134 if (thread != 0) { 3135 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3136 status = playbackThread->attachAuxEffect(this, EffectId); 3137 } 3138 return status; 3139} 3140 3141void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3142{ 3143 mAuxEffectId = EffectId; 3144 mAuxBuffer = buffer; 3145} 3146 3147// ---------------------------------------------------------------------------- 3148 3149// RecordTrack constructor must be called with AudioFlinger::mLock held 3150AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3151 const wp<ThreadBase>& thread, 3152 const sp<Client>& client, 3153 uint32_t sampleRate, 3154 int format, 3155 int channelCount, 3156 int frameCount, 3157 uint32_t flags, 3158 int sessionId) 3159 : TrackBase(thread, client, sampleRate, format, 3160 channelCount, frameCount, flags, 0, sessionId), 3161 mOverflow(false) 3162{ 3163 if (mCblk != NULL) { 3164 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3165 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3166 mCblk->frameSize = channelCount * sizeof(int16_t); 3167 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3168 mCblk->frameSize = channelCount * sizeof(int8_t); 3169 } else { 3170 mCblk->frameSize = sizeof(int8_t); 3171 } 3172 } 3173} 3174 3175AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3176{ 3177 sp<ThreadBase> thread = mThread.promote(); 3178 if (thread != 0) { 3179 AudioSystem::releaseInput(thread->id()); 3180 } 3181} 3182 3183status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3184{ 3185 audio_track_cblk_t* cblk = this->cblk(); 3186 uint32_t framesAvail; 3187 uint32_t framesReq = buffer->frameCount; 3188 3189 // Check if last stepServer failed, try to step now 3190 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3191 if (!step()) goto getNextBuffer_exit; 3192 LOGV("stepServer recovered"); 3193 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3194 } 3195 3196 framesAvail = cblk->framesAvailable_l(); 3197 3198 if (LIKELY(framesAvail)) { 3199 uint32_t s = cblk->server; 3200 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3201 3202 if (framesReq > framesAvail) { 3203 framesReq = framesAvail; 3204 } 3205 if (s + framesReq > bufferEnd) { 3206 framesReq = bufferEnd - s; 3207 } 3208 3209 buffer->raw = getBuffer(s, framesReq); 3210 if (buffer->raw == 0) goto getNextBuffer_exit; 3211 3212 buffer->frameCount = framesReq; 3213 return NO_ERROR; 3214 } 3215 3216getNextBuffer_exit: 3217 buffer->raw = 0; 3218 buffer->frameCount = 0; 3219 return NOT_ENOUGH_DATA; 3220} 3221 3222status_t AudioFlinger::RecordThread::RecordTrack::start() 3223{ 3224 sp<ThreadBase> thread = mThread.promote(); 3225 if (thread != 0) { 3226 RecordThread *recordThread = (RecordThread *)thread.get(); 3227 return recordThread->start(this); 3228 } else { 3229 return BAD_VALUE; 3230 } 3231} 3232 3233void AudioFlinger::RecordThread::RecordTrack::stop() 3234{ 3235 sp<ThreadBase> thread = mThread.promote(); 3236 if (thread != 0) { 3237 RecordThread *recordThread = (RecordThread *)thread.get(); 3238 recordThread->stop(this); 3239 TrackBase::reset(); 3240 // Force overerrun condition to avoid false overrun callback until first data is 3241 // read from buffer 3242 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3243 } 3244} 3245 3246void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3247{ 3248 snprintf(buffer, size, " %05d %03u %03u %05d %04u %01d %05u %08x %08x\n", 3249 (mClient == NULL) ? getpid() : mClient->pid(), 3250 mFormat, 3251 mCblk->channelCount, 3252 mSessionId, 3253 mFrameCount, 3254 mState, 3255 mCblk->sampleRate, 3256 mCblk->server, 3257 mCblk->user); 3258} 3259 3260 3261// ---------------------------------------------------------------------------- 3262 3263AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3264 const wp<ThreadBase>& thread, 3265 DuplicatingThread *sourceThread, 3266 uint32_t sampleRate, 3267 int format, 3268 int channelCount, 3269 int frameCount) 3270 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelCount, frameCount, NULL, 0), 3271 mActive(false), mSourceThread(sourceThread) 3272{ 3273 3274 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3275 if (mCblk != NULL) { 3276 mCblk->flags |= CBLK_DIRECTION_OUT; 3277 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3278 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3279 mOutBuffer.frameCount = 0; 3280 playbackThread->mTracks.add(this); 3281 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, mCblk->frameCount %d, mCblk->sampleRate %d, mCblk->channelCount %d mBufferEnd %p", 3282 mCblk, mBuffer, mCblk->buffers, mCblk->frameCount, mCblk->sampleRate, mCblk->channelCount, mBufferEnd); 3283 } else { 3284 LOGW("Error creating output track on thread %p", playbackThread); 3285 } 3286} 3287 3288AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3289{ 3290 clearBufferQueue(); 3291} 3292 3293status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3294{ 3295 status_t status = Track::start(); 3296 if (status != NO_ERROR) { 3297 return status; 3298 } 3299 3300 mActive = true; 3301 mRetryCount = 127; 3302 return status; 3303} 3304 3305void AudioFlinger::PlaybackThread::OutputTrack::stop() 3306{ 3307 Track::stop(); 3308 clearBufferQueue(); 3309 mOutBuffer.frameCount = 0; 3310 mActive = false; 3311} 3312 3313bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3314{ 3315 Buffer *pInBuffer; 3316 Buffer inBuffer; 3317 uint32_t channelCount = mCblk->channelCount; 3318 bool outputBufferFull = false; 3319 inBuffer.frameCount = frames; 3320 inBuffer.i16 = data; 3321 3322 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3323 3324 if (!mActive && frames != 0) { 3325 start(); 3326 sp<ThreadBase> thread = mThread.promote(); 3327 if (thread != 0) { 3328 MixerThread *mixerThread = (MixerThread *)thread.get(); 3329 if (mCblk->frameCount > frames){ 3330 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3331 uint32_t startFrames = (mCblk->frameCount - frames); 3332 pInBuffer = new Buffer; 3333 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3334 pInBuffer->frameCount = startFrames; 3335 pInBuffer->i16 = pInBuffer->mBuffer; 3336 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3337 mBufferQueue.add(pInBuffer); 3338 } else { 3339 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3340 } 3341 } 3342 } 3343 } 3344 3345 while (waitTimeLeftMs) { 3346 // First write pending buffers, then new data 3347 if (mBufferQueue.size()) { 3348 pInBuffer = mBufferQueue.itemAt(0); 3349 } else { 3350 pInBuffer = &inBuffer; 3351 } 3352 3353 if (pInBuffer->frameCount == 0) { 3354 break; 3355 } 3356 3357 if (mOutBuffer.frameCount == 0) { 3358 mOutBuffer.frameCount = pInBuffer->frameCount; 3359 nsecs_t startTime = systemTime(); 3360 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3361 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3362 outputBufferFull = true; 3363 break; 3364 } 3365 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3366 if (waitTimeLeftMs >= waitTimeMs) { 3367 waitTimeLeftMs -= waitTimeMs; 3368 } else { 3369 waitTimeLeftMs = 0; 3370 } 3371 } 3372 3373 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3374 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3375 mCblk->stepUser(outFrames); 3376 pInBuffer->frameCount -= outFrames; 3377 pInBuffer->i16 += outFrames * channelCount; 3378 mOutBuffer.frameCount -= outFrames; 3379 mOutBuffer.i16 += outFrames * channelCount; 3380 3381 if (pInBuffer->frameCount == 0) { 3382 if (mBufferQueue.size()) { 3383 mBufferQueue.removeAt(0); 3384 delete [] pInBuffer->mBuffer; 3385 delete pInBuffer; 3386 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3387 } else { 3388 break; 3389 } 3390 } 3391 } 3392 3393 // If we could not write all frames, allocate a buffer and queue it for next time. 3394 if (inBuffer.frameCount) { 3395 sp<ThreadBase> thread = mThread.promote(); 3396 if (thread != 0 && !thread->standby()) { 3397 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3398 pInBuffer = new Buffer; 3399 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3400 pInBuffer->frameCount = inBuffer.frameCount; 3401 pInBuffer->i16 = pInBuffer->mBuffer; 3402 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3403 mBufferQueue.add(pInBuffer); 3404 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3405 } else { 3406 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3407 } 3408 } 3409 } 3410 3411 // Calling write() with a 0 length buffer, means that no more data will be written: 3412 // If no more buffers are pending, fill output track buffer to make sure it is started 3413 // by output mixer. 3414 if (frames == 0 && mBufferQueue.size() == 0) { 3415 if (mCblk->user < mCblk->frameCount) { 3416 frames = mCblk->frameCount - mCblk->user; 3417 pInBuffer = new Buffer; 3418 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3419 pInBuffer->frameCount = frames; 3420 pInBuffer->i16 = pInBuffer->mBuffer; 3421 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3422 mBufferQueue.add(pInBuffer); 3423 } else if (mActive) { 3424 stop(); 3425 } 3426 } 3427 3428 return outputBufferFull; 3429} 3430 3431status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3432{ 3433 int active; 3434 status_t result; 3435 audio_track_cblk_t* cblk = mCblk; 3436 uint32_t framesReq = buffer->frameCount; 3437 3438// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3439 buffer->frameCount = 0; 3440 3441 uint32_t framesAvail = cblk->framesAvailable(); 3442 3443 3444 if (framesAvail == 0) { 3445 Mutex::Autolock _l(cblk->lock); 3446 goto start_loop_here; 3447 while (framesAvail == 0) { 3448 active = mActive; 3449 if (UNLIKELY(!active)) { 3450 LOGV("Not active and NO_MORE_BUFFERS"); 3451 return AudioTrack::NO_MORE_BUFFERS; 3452 } 3453 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3454 if (result != NO_ERROR) { 3455 return AudioTrack::NO_MORE_BUFFERS; 3456 } 3457 // read the server count again 3458 start_loop_here: 3459 framesAvail = cblk->framesAvailable_l(); 3460 } 3461 } 3462 3463// if (framesAvail < framesReq) { 3464// return AudioTrack::NO_MORE_BUFFERS; 3465// } 3466 3467 if (framesReq > framesAvail) { 3468 framesReq = framesAvail; 3469 } 3470 3471 uint32_t u = cblk->user; 3472 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3473 3474 if (u + framesReq > bufferEnd) { 3475 framesReq = bufferEnd - u; 3476 } 3477 3478 buffer->frameCount = framesReq; 3479 buffer->raw = (void *)cblk->buffer(u); 3480 return NO_ERROR; 3481} 3482 3483 3484void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3485{ 3486 size_t size = mBufferQueue.size(); 3487 Buffer *pBuffer; 3488 3489 for (size_t i = 0; i < size; i++) { 3490 pBuffer = mBufferQueue.itemAt(i); 3491 delete [] pBuffer->mBuffer; 3492 delete pBuffer; 3493 } 3494 mBufferQueue.clear(); 3495} 3496 3497// ---------------------------------------------------------------------------- 3498 3499AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3500 : RefBase(), 3501 mAudioFlinger(audioFlinger), 3502 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3503 mPid(pid) 3504{ 3505 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3506} 3507 3508// Client destructor must be called with AudioFlinger::mLock held 3509AudioFlinger::Client::~Client() 3510{ 3511 mAudioFlinger->removeClient_l(mPid); 3512} 3513 3514const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3515{ 3516 return mMemoryDealer; 3517} 3518 3519// ---------------------------------------------------------------------------- 3520 3521AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3522 const sp<IAudioFlingerClient>& client, 3523 pid_t pid) 3524 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3525{ 3526} 3527 3528AudioFlinger::NotificationClient::~NotificationClient() 3529{ 3530 mClient.clear(); 3531} 3532 3533void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3534{ 3535 sp<NotificationClient> keep(this); 3536 { 3537 mAudioFlinger->removeNotificationClient(mPid); 3538 } 3539} 3540 3541// ---------------------------------------------------------------------------- 3542 3543AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3544 : BnAudioTrack(), 3545 mTrack(track) 3546{ 3547} 3548 3549AudioFlinger::TrackHandle::~TrackHandle() { 3550 // just stop the track on deletion, associated resources 3551 // will be freed from the main thread once all pending buffers have 3552 // been played. Unless it's not in the active track list, in which 3553 // case we free everything now... 3554 mTrack->destroy(); 3555} 3556 3557status_t AudioFlinger::TrackHandle::start() { 3558 return mTrack->start(); 3559} 3560 3561void AudioFlinger::TrackHandle::stop() { 3562 mTrack->stop(); 3563} 3564 3565void AudioFlinger::TrackHandle::flush() { 3566 mTrack->flush(); 3567} 3568 3569void AudioFlinger::TrackHandle::mute(bool e) { 3570 mTrack->mute(e); 3571} 3572 3573void AudioFlinger::TrackHandle::pause() { 3574 mTrack->pause(); 3575} 3576 3577void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3578 mTrack->setVolume(left, right); 3579} 3580 3581sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3582 return mTrack->getCblk(); 3583} 3584 3585status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3586{ 3587 return mTrack->attachAuxEffect(EffectId); 3588} 3589 3590status_t AudioFlinger::TrackHandle::onTransact( 3591 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3592{ 3593 return BnAudioTrack::onTransact(code, data, reply, flags); 3594} 3595 3596// ---------------------------------------------------------------------------- 3597 3598sp<IAudioRecord> AudioFlinger::openRecord( 3599 pid_t pid, 3600 int input, 3601 uint32_t sampleRate, 3602 int format, 3603 int channelCount, 3604 int frameCount, 3605 uint32_t flags, 3606 int *sessionId, 3607 status_t *status) 3608{ 3609 sp<RecordThread::RecordTrack> recordTrack; 3610 sp<RecordHandle> recordHandle; 3611 sp<Client> client; 3612 wp<Client> wclient; 3613 status_t lStatus; 3614 RecordThread *thread; 3615 size_t inFrameCount; 3616 int lSessionId; 3617 3618 // check calling permissions 3619 if (!recordingAllowed()) { 3620 lStatus = PERMISSION_DENIED; 3621 goto Exit; 3622 } 3623 3624 // add client to list 3625 { // scope for mLock 3626 Mutex::Autolock _l(mLock); 3627 thread = checkRecordThread_l(input); 3628 if (thread == NULL) { 3629 lStatus = BAD_VALUE; 3630 goto Exit; 3631 } 3632 3633 wclient = mClients.valueFor(pid); 3634 if (wclient != NULL) { 3635 client = wclient.promote(); 3636 } else { 3637 client = new Client(this, pid); 3638 mClients.add(pid, client); 3639 } 3640 3641 // If no audio session id is provided, create one here 3642 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 3643 lSessionId = *sessionId; 3644 } else { 3645 lSessionId = nextUniqueId_l(); 3646 if (sessionId != NULL) { 3647 *sessionId = lSessionId; 3648 } 3649 } 3650 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3651 recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, 3652 format, channelCount, frameCount, flags, lSessionId); 3653 } 3654 if (recordTrack->getCblk() == NULL) { 3655 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3656 // destructor is called by the TrackBase destructor with mLock held 3657 client.clear(); 3658 recordTrack.clear(); 3659 lStatus = NO_MEMORY; 3660 goto Exit; 3661 } 3662 3663 // return to handle to client 3664 recordHandle = new RecordHandle(recordTrack); 3665 lStatus = NO_ERROR; 3666 3667Exit: 3668 if (status) { 3669 *status = lStatus; 3670 } 3671 return recordHandle; 3672} 3673 3674// ---------------------------------------------------------------------------- 3675 3676AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3677 : BnAudioRecord(), 3678 mRecordTrack(recordTrack) 3679{ 3680} 3681 3682AudioFlinger::RecordHandle::~RecordHandle() { 3683 stop(); 3684} 3685 3686status_t AudioFlinger::RecordHandle::start() { 3687 LOGV("RecordHandle::start()"); 3688 return mRecordTrack->start(); 3689} 3690 3691void AudioFlinger::RecordHandle::stop() { 3692 LOGV("RecordHandle::stop()"); 3693 mRecordTrack->stop(); 3694} 3695 3696sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3697 return mRecordTrack->getCblk(); 3698} 3699 3700status_t AudioFlinger::RecordHandle::onTransact( 3701 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3702{ 3703 return BnAudioRecord::onTransact(code, data, reply, flags); 3704} 3705 3706// ---------------------------------------------------------------------------- 3707 3708AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, struct audio_stream_in *input, uint32_t sampleRate, uint32_t channels, int id) : 3709 ThreadBase(audioFlinger, id), 3710 mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3711{ 3712 mReqChannelCount = popcount(channels); 3713 mReqSampleRate = sampleRate; 3714 readInputParameters(); 3715} 3716 3717 3718AudioFlinger::RecordThread::~RecordThread() 3719{ 3720 delete[] mRsmpInBuffer; 3721 if (mResampler != 0) { 3722 delete mResampler; 3723 delete[] mRsmpOutBuffer; 3724 } 3725} 3726 3727void AudioFlinger::RecordThread::onFirstRef() 3728{ 3729 const size_t SIZE = 256; 3730 char buffer[SIZE]; 3731 3732 snprintf(buffer, SIZE, "Record Thread %p", this); 3733 3734 run(buffer, PRIORITY_URGENT_AUDIO); 3735} 3736 3737bool AudioFlinger::RecordThread::threadLoop() 3738{ 3739 AudioBufferProvider::Buffer buffer; 3740 sp<RecordTrack> activeTrack; 3741 3742 nsecs_t lastWarning = 0; 3743 3744 // start recording 3745 while (!exitPending()) { 3746 3747 processConfigEvents(); 3748 3749 { // scope for mLock 3750 Mutex::Autolock _l(mLock); 3751 checkForNewParameters_l(); 3752 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3753 if (!mStandby) { 3754 mInput->common.standby(&mInput->common); 3755 mStandby = true; 3756 } 3757 3758 if (exitPending()) break; 3759 3760 LOGV("RecordThread: loop stopping"); 3761 // go to sleep 3762 mWaitWorkCV.wait(mLock); 3763 LOGV("RecordThread: loop starting"); 3764 continue; 3765 } 3766 if (mActiveTrack != 0) { 3767 if (mActiveTrack->mState == TrackBase::PAUSING) { 3768 if (!mStandby) { 3769 mInput->common.standby(&mInput->common); 3770 mStandby = true; 3771 } 3772 mActiveTrack.clear(); 3773 mStartStopCond.broadcast(); 3774 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3775 if (mReqChannelCount != mActiveTrack->channelCount()) { 3776 mActiveTrack.clear(); 3777 mStartStopCond.broadcast(); 3778 } else if (mBytesRead != 0) { 3779 // record start succeeds only if first read from audio input 3780 // succeeds 3781 if (mBytesRead > 0) { 3782 mActiveTrack->mState = TrackBase::ACTIVE; 3783 } else { 3784 mActiveTrack.clear(); 3785 } 3786 mStartStopCond.broadcast(); 3787 } 3788 mStandby = false; 3789 } 3790 } 3791 } 3792 3793 if (mActiveTrack != 0) { 3794 if (mActiveTrack->mState != TrackBase::ACTIVE && 3795 mActiveTrack->mState != TrackBase::RESUMING) { 3796 usleep(5000); 3797 continue; 3798 } 3799 buffer.frameCount = mFrameCount; 3800 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3801 size_t framesOut = buffer.frameCount; 3802 if (mResampler == 0) { 3803 // no resampling 3804 while (framesOut) { 3805 size_t framesIn = mFrameCount - mRsmpInIndex; 3806 if (framesIn) { 3807 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3808 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3809 if (framesIn > framesOut) 3810 framesIn = framesOut; 3811 mRsmpInIndex += framesIn; 3812 framesOut -= framesIn; 3813 if ((int)mChannelCount == mReqChannelCount || 3814 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3815 memcpy(dst, src, framesIn * mFrameSize); 3816 } else { 3817 int16_t *src16 = (int16_t *)src; 3818 int16_t *dst16 = (int16_t *)dst; 3819 if (mChannelCount == 1) { 3820 while (framesIn--) { 3821 *dst16++ = *src16; 3822 *dst16++ = *src16++; 3823 } 3824 } else { 3825 while (framesIn--) { 3826 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3827 src16 += 2; 3828 } 3829 } 3830 } 3831 } 3832 if (framesOut && mFrameCount == mRsmpInIndex) { 3833 if (framesOut == mFrameCount && 3834 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3835 mBytesRead = mInput->read(mInput, buffer.raw, mInputBytes); 3836 framesOut = 0; 3837 } else { 3838 mBytesRead = mInput->read(mInput, mRsmpInBuffer, mInputBytes); 3839 mRsmpInIndex = 0; 3840 } 3841 if (mBytesRead < 0) { 3842 LOGE("Error reading audio input"); 3843 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3844 // Force input into standby so that it tries to 3845 // recover at next read attempt 3846 mInput->common.standby(&mInput->common); 3847 usleep(5000); 3848 } 3849 mRsmpInIndex = mFrameCount; 3850 framesOut = 0; 3851 buffer.frameCount = 0; 3852 } 3853 } 3854 } 3855 } else { 3856 // resampling 3857 3858 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3859 // alter output frame count as if we were expecting stereo samples 3860 if (mChannelCount == 1 && mReqChannelCount == 1) { 3861 framesOut >>= 1; 3862 } 3863 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3864 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3865 // are 32 bit aligned which should be always true. 3866 if (mChannelCount == 2 && mReqChannelCount == 1) { 3867 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3868 // the resampler always outputs stereo samples: do post stereo to mono conversion 3869 int16_t *src = (int16_t *)mRsmpOutBuffer; 3870 int16_t *dst = buffer.i16; 3871 while (framesOut--) { 3872 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3873 src += 2; 3874 } 3875 } else { 3876 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3877 } 3878 3879 } 3880 mActiveTrack->releaseBuffer(&buffer); 3881 mActiveTrack->overflow(); 3882 } 3883 // client isn't retrieving buffers fast enough 3884 else { 3885 if (!mActiveTrack->setOverflow()) { 3886 nsecs_t now = systemTime(); 3887 if ((now - lastWarning) > kWarningThrottle) { 3888 LOGW("RecordThread: buffer overflow"); 3889 lastWarning = now; 3890 } 3891 } 3892 // Release the processor for a while before asking for a new buffer. 3893 // This will give the application more chance to read from the buffer and 3894 // clear the overflow. 3895 usleep(5000); 3896 } 3897 } 3898 } 3899 3900 if (!mStandby) { 3901 mInput->common.standby(&mInput->common); 3902 } 3903 mActiveTrack.clear(); 3904 3905 mStartStopCond.broadcast(); 3906 3907 LOGV("RecordThread %p exiting", this); 3908 return false; 3909} 3910 3911status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 3912{ 3913 LOGV("RecordThread::start"); 3914 sp <ThreadBase> strongMe = this; 3915 status_t status = NO_ERROR; 3916 { 3917 AutoMutex lock(&mLock); 3918 if (mActiveTrack != 0) { 3919 if (recordTrack != mActiveTrack.get()) { 3920 status = -EBUSY; 3921 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 3922 mActiveTrack->mState = TrackBase::ACTIVE; 3923 } 3924 return status; 3925 } 3926 3927 recordTrack->mState = TrackBase::IDLE; 3928 mActiveTrack = recordTrack; 3929 mLock.unlock(); 3930 status_t status = AudioSystem::startInput(mId); 3931 mLock.lock(); 3932 if (status != NO_ERROR) { 3933 mActiveTrack.clear(); 3934 return status; 3935 } 3936 mRsmpInIndex = mFrameCount; 3937 mBytesRead = 0; 3938 if (mResampler != NULL) { 3939 mResampler->reset(); 3940 } 3941 mActiveTrack->mState = TrackBase::RESUMING; 3942 // signal thread to start 3943 LOGV("Signal record thread"); 3944 mWaitWorkCV.signal(); 3945 // do not wait for mStartStopCond if exiting 3946 if (mExiting) { 3947 mActiveTrack.clear(); 3948 status = INVALID_OPERATION; 3949 goto startError; 3950 } 3951 mStartStopCond.wait(mLock); 3952 if (mActiveTrack == 0) { 3953 LOGV("Record failed to start"); 3954 status = BAD_VALUE; 3955 goto startError; 3956 } 3957 LOGV("Record started OK"); 3958 return status; 3959 } 3960startError: 3961 AudioSystem::stopInput(mId); 3962 return status; 3963} 3964 3965void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 3966 LOGV("RecordThread::stop"); 3967 sp <ThreadBase> strongMe = this; 3968 { 3969 AutoMutex lock(&mLock); 3970 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 3971 mActiveTrack->mState = TrackBase::PAUSING; 3972 // do not wait for mStartStopCond if exiting 3973 if (mExiting) { 3974 return; 3975 } 3976 mStartStopCond.wait(mLock); 3977 // if we have been restarted, recordTrack == mActiveTrack.get() here 3978 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 3979 mLock.unlock(); 3980 AudioSystem::stopInput(mId); 3981 mLock.lock(); 3982 LOGV("Record stopped OK"); 3983 } 3984 } 3985 } 3986} 3987 3988status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 3989{ 3990 const size_t SIZE = 256; 3991 char buffer[SIZE]; 3992 String8 result; 3993 pid_t pid = 0; 3994 3995 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 3996 result.append(buffer); 3997 3998 if (mActiveTrack != 0) { 3999 result.append("Active Track:\n"); 4000 result.append(" Clien Fmt Chn Session Buf S SRate Serv User\n"); 4001 mActiveTrack->dump(buffer, SIZE); 4002 result.append(buffer); 4003 4004 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4005 result.append(buffer); 4006 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4007 result.append(buffer); 4008 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4009 result.append(buffer); 4010 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4011 result.append(buffer); 4012 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4013 result.append(buffer); 4014 4015 4016 } else { 4017 result.append("No record client\n"); 4018 } 4019 write(fd, result.string(), result.size()); 4020 4021 dumpBase(fd, args); 4022 4023 return NO_ERROR; 4024} 4025 4026status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4027{ 4028 size_t framesReq = buffer->frameCount; 4029 size_t framesReady = mFrameCount - mRsmpInIndex; 4030 int channelCount; 4031 4032 if (framesReady == 0) { 4033 mBytesRead = mInput->read(mInput, mRsmpInBuffer, mInputBytes); 4034 if (mBytesRead < 0) { 4035 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4036 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4037 // Force input into standby so that it tries to 4038 // recover at next read attempt 4039 mInput->common.standby(&mInput->common); 4040 usleep(5000); 4041 } 4042 buffer->raw = 0; 4043 buffer->frameCount = 0; 4044 return NOT_ENOUGH_DATA; 4045 } 4046 mRsmpInIndex = 0; 4047 framesReady = mFrameCount; 4048 } 4049 4050 if (framesReq > framesReady) { 4051 framesReq = framesReady; 4052 } 4053 4054 if (mChannelCount == 1 && mReqChannelCount == 2) { 4055 channelCount = 1; 4056 } else { 4057 channelCount = 2; 4058 } 4059 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4060 buffer->frameCount = framesReq; 4061 return NO_ERROR; 4062} 4063 4064void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4065{ 4066 mRsmpInIndex += buffer->frameCount; 4067 buffer->frameCount = 0; 4068} 4069 4070bool AudioFlinger::RecordThread::checkForNewParameters_l() 4071{ 4072 bool reconfig = false; 4073 4074 while (!mNewParameters.isEmpty()) { 4075 status_t status = NO_ERROR; 4076 String8 keyValuePair = mNewParameters[0]; 4077 AudioParameter param = AudioParameter(keyValuePair); 4078 int value; 4079 int reqFormat = mFormat; 4080 int reqSamplingRate = mReqSampleRate; 4081 int reqChannelCount = mReqChannelCount; 4082 4083 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4084 reqSamplingRate = value; 4085 reconfig = true; 4086 } 4087 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4088 reqFormat = value; 4089 reconfig = true; 4090 } 4091 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4092 reqChannelCount = popcount(value); 4093 reconfig = true; 4094 } 4095 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4096 // do not accept frame count changes if tracks are open as the track buffer 4097 // size depends on frame count and correct behavior would not be garantied 4098 // if frame count is changed after track creation 4099 if (mActiveTrack != 0) { 4100 status = INVALID_OPERATION; 4101 } else { 4102 reconfig = true; 4103 } 4104 } 4105 if (status == NO_ERROR) { 4106 status = mInput->common.set_parameters(&mInput->common, keyValuePair.string()); 4107 if (status == INVALID_OPERATION) { 4108 mInput->common.standby(&mInput->common); 4109 status = mInput->common.set_parameters(&mInput->common, keyValuePair.string()); 4110 } 4111 if (reconfig) { 4112 if (status == BAD_VALUE && 4113 reqFormat == mInput->common.get_format(&mInput->common) && 4114 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4115 ((int)mInput->common.get_sample_rate(&mInput->common) <= (2 * reqSamplingRate)) && 4116 (popcount(mInput->common.get_channels(&mInput->common)) < 3) && 4117 (reqChannelCount < 3)) { 4118 status = NO_ERROR; 4119 } 4120 if (status == NO_ERROR) { 4121 readInputParameters(); 4122 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4123 } 4124 } 4125 } 4126 4127 mNewParameters.removeAt(0); 4128 4129 mParamStatus = status; 4130 mParamCond.signal(); 4131 mWaitWorkCV.wait(mLock); 4132 } 4133 return reconfig; 4134} 4135 4136String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4137{ 4138 char *s; 4139 String8 out_s8; 4140 4141 s = mInput->common.get_parameters(&mInput->common, keys.string()); 4142 out_s8 = String8(s); 4143 free(s); 4144 return out_s8; 4145} 4146 4147void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4148 AudioSystem::OutputDescriptor desc; 4149 void *param2 = 0; 4150 4151 switch (event) { 4152 case AudioSystem::INPUT_OPENED: 4153 case AudioSystem::INPUT_CONFIG_CHANGED: 4154 desc.channels = mChannels; 4155 desc.samplingRate = mSampleRate; 4156 desc.format = mFormat; 4157 desc.frameCount = mFrameCount; 4158 desc.latency = 0; 4159 param2 = &desc; 4160 break; 4161 4162 case AudioSystem::INPUT_CLOSED: 4163 default: 4164 break; 4165 } 4166 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4167} 4168 4169void AudioFlinger::RecordThread::readInputParameters() 4170{ 4171 if (mRsmpInBuffer) delete mRsmpInBuffer; 4172 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4173 if (mResampler) delete mResampler; 4174 mResampler = 0; 4175 4176 mSampleRate = mInput->common.get_sample_rate(&mInput->common); 4177 mChannels = mInput->common.get_channels(&mInput->common); 4178 mChannelCount = (uint16_t)popcount(mChannels); 4179 mFormat = mInput->common.get_format(&mInput->common); 4180 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->common); 4181 mInputBytes = mInput->common.get_buffer_size(&mInput->common); 4182 mFrameCount = mInputBytes / mFrameSize; 4183 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4184 4185 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4186 { 4187 int channelCount; 4188 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4189 // stereo to mono post process as the resampler always outputs stereo. 4190 if (mChannelCount == 1 && mReqChannelCount == 2) { 4191 channelCount = 1; 4192 } else { 4193 channelCount = 2; 4194 } 4195 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4196 mResampler->setSampleRate(mSampleRate); 4197 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4198 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4199 4200 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4201 if (mChannelCount == 1 && mReqChannelCount == 1) { 4202 mFrameCount >>= 1; 4203 } 4204 4205 } 4206 mRsmpInIndex = mFrameCount; 4207} 4208 4209unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4210{ 4211 return mInput->get_input_frames_lost(mInput); 4212} 4213 4214// ---------------------------------------------------------------------------- 4215 4216int AudioFlinger::openOutput(uint32_t *pDevices, 4217 uint32_t *pSamplingRate, 4218 uint32_t *pFormat, 4219 uint32_t *pChannels, 4220 uint32_t *pLatencyMs, 4221 uint32_t flags) 4222{ 4223 status_t status; 4224 PlaybackThread *thread = NULL; 4225 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4226 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4227 uint32_t format = pFormat ? *pFormat : 0; 4228 uint32_t channels = pChannels ? *pChannels : 0; 4229 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4230 struct audio_stream_out *output; 4231 4232 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4233 pDevices ? *pDevices : 0, 4234 samplingRate, 4235 format, 4236 channels, 4237 flags); 4238 4239 if (pDevices == NULL || *pDevices == 0) { 4240 return 0; 4241 } 4242 Mutex::Autolock _l(mLock); 4243 4244 status = mAudioHardwareDev->open_output_stream(mAudioHardwareDev, *pDevices, 4245 (int *)&format, 4246 &channels, 4247 &samplingRate, 4248 &output); 4249 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4250 output, 4251 samplingRate, 4252 format, 4253 channels, 4254 status); 4255 4256 mHardwareStatus = AUDIO_HW_IDLE; 4257 if (output != 0) { 4258 int id = nextUniqueId_l(); 4259 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4260 (format != AUDIO_FORMAT_PCM_16_BIT) || 4261 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4262 thread = new DirectOutputThread(this, output, id, *pDevices); 4263 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4264 } else { 4265 thread = new MixerThread(this, output, id, *pDevices); 4266 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4267 } 4268 mPlaybackThreads.add(id, thread); 4269 4270 if (pSamplingRate) *pSamplingRate = samplingRate; 4271 if (pFormat) *pFormat = format; 4272 if (pChannels) *pChannels = channels; 4273 if (pLatencyMs) *pLatencyMs = thread->latency(); 4274 4275 // notify client processes of the new output creation 4276 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4277 return id; 4278 } 4279 4280 return 0; 4281} 4282 4283int AudioFlinger::openDuplicateOutput(int output1, int output2) 4284{ 4285 Mutex::Autolock _l(mLock); 4286 MixerThread *thread1 = checkMixerThread_l(output1); 4287 MixerThread *thread2 = checkMixerThread_l(output2); 4288 4289 if (thread1 == NULL || thread2 == NULL) { 4290 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4291 return 0; 4292 } 4293 4294 int id = nextUniqueId_l(); 4295 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4296 thread->addOutputTrack(thread2); 4297 mPlaybackThreads.add(id, thread); 4298 // notify client processes of the new output creation 4299 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4300 return id; 4301} 4302 4303status_t AudioFlinger::closeOutput(int output) 4304{ 4305 // keep strong reference on the playback thread so that 4306 // it is not destroyed while exit() is executed 4307 sp <PlaybackThread> thread; 4308 { 4309 Mutex::Autolock _l(mLock); 4310 thread = checkPlaybackThread_l(output); 4311 if (thread == NULL) { 4312 return BAD_VALUE; 4313 } 4314 4315 LOGV("closeOutput() %d", output); 4316 4317 if (thread->type() == PlaybackThread::MIXER) { 4318 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4319 if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { 4320 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4321 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4322 } 4323 } 4324 } 4325 void *param2 = 0; 4326 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4327 mPlaybackThreads.removeItem(output); 4328 } 4329 thread->exit(); 4330 4331 if (thread->type() != PlaybackThread::DUPLICATING) { 4332 mAudioHardwareDev->close_output_stream(mAudioHardwareDev, thread->getOutput()); 4333 } 4334 return NO_ERROR; 4335} 4336 4337status_t AudioFlinger::suspendOutput(int output) 4338{ 4339 Mutex::Autolock _l(mLock); 4340 PlaybackThread *thread = checkPlaybackThread_l(output); 4341 4342 if (thread == NULL) { 4343 return BAD_VALUE; 4344 } 4345 4346 LOGV("suspendOutput() %d", output); 4347 thread->suspend(); 4348 4349 return NO_ERROR; 4350} 4351 4352status_t AudioFlinger::restoreOutput(int output) 4353{ 4354 Mutex::Autolock _l(mLock); 4355 PlaybackThread *thread = checkPlaybackThread_l(output); 4356 4357 if (thread == NULL) { 4358 return BAD_VALUE; 4359 } 4360 4361 LOGV("restoreOutput() %d", output); 4362 4363 thread->restore(); 4364 4365 return NO_ERROR; 4366} 4367 4368int AudioFlinger::openInput(uint32_t *pDevices, 4369 uint32_t *pSamplingRate, 4370 uint32_t *pFormat, 4371 uint32_t *pChannels, 4372 uint32_t acoustics) 4373{ 4374 status_t status; 4375 RecordThread *thread = NULL; 4376 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4377 uint32_t format = pFormat ? *pFormat : 0; 4378 uint32_t channels = pChannels ? *pChannels : 0; 4379 uint32_t reqSamplingRate = samplingRate; 4380 uint32_t reqFormat = format; 4381 uint32_t reqChannels = channels; 4382 struct audio_stream_in *input; 4383 4384 if (pDevices == NULL || *pDevices == 0) { 4385 return 0; 4386 } 4387 Mutex::Autolock _l(mLock); 4388 4389 status = mAudioHardwareDev->open_input_stream(mAudioHardwareDev, 4390 *pDevices, 4391 (int *)&format, 4392 &channels, 4393 &samplingRate, 4394 (audio_in_acoustics_t)acoustics, 4395 &input); 4396 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4397 input, 4398 samplingRate, 4399 format, 4400 channels, 4401 acoustics, 4402 status); 4403 4404 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4405 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4406 // or stereo to mono conversions on 16 bit PCM inputs. 4407 if (input == 0 && status == BAD_VALUE && 4408 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 4409 (samplingRate <= 2 * reqSamplingRate) && 4410 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 4411 LOGV("openInput() reopening with proposed sampling rate and channels"); 4412 status = mAudioHardwareDev->open_input_stream(mAudioHardwareDev, 4413 *pDevices, 4414 (int *)&format, 4415 &channels, 4416 &samplingRate, 4417 (audio_in_acoustics_t)acoustics, 4418 &input); 4419 } 4420 4421 if (input != 0) { 4422 int id = nextUniqueId_l(); 4423 // Start record thread 4424 thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); 4425 mRecordThreads.add(id, thread); 4426 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4427 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4428 if (pFormat) *pFormat = format; 4429 if (pChannels) *pChannels = reqChannels; 4430 4431 input->common.standby(&input->common); 4432 4433 // notify client processes of the new input creation 4434 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4435 return id; 4436 } 4437 4438 return 0; 4439} 4440 4441status_t AudioFlinger::closeInput(int input) 4442{ 4443 // keep strong reference on the record thread so that 4444 // it is not destroyed while exit() is executed 4445 sp <RecordThread> thread; 4446 { 4447 Mutex::Autolock _l(mLock); 4448 thread = checkRecordThread_l(input); 4449 if (thread == NULL) { 4450 return BAD_VALUE; 4451 } 4452 4453 LOGV("closeInput() %d", input); 4454 void *param2 = 0; 4455 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4456 mRecordThreads.removeItem(input); 4457 } 4458 thread->exit(); 4459 4460 mAudioHardwareDev->close_input_stream(mAudioHardwareDev, thread->getInput()); 4461 4462 return NO_ERROR; 4463} 4464 4465status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4466{ 4467 Mutex::Autolock _l(mLock); 4468 MixerThread *dstThread = checkMixerThread_l(output); 4469 if (dstThread == NULL) { 4470 LOGW("setStreamOutput() bad output id %d", output); 4471 return BAD_VALUE; 4472 } 4473 4474 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4475 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4476 4477 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4478 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4479 if (thread != dstThread && 4480 thread->type() != PlaybackThread::DIRECT) { 4481 MixerThread *srcThread = (MixerThread *)thread; 4482 srcThread->invalidateTracks(stream); 4483 } 4484 } 4485 4486 return NO_ERROR; 4487} 4488 4489 4490int AudioFlinger::newAudioSessionId() 4491{ 4492 AutoMutex _l(mLock); 4493 return nextUniqueId_l(); 4494} 4495 4496// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4497AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4498{ 4499 PlaybackThread *thread = NULL; 4500 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4501 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4502 } 4503 return thread; 4504} 4505 4506// checkMixerThread_l() must be called with AudioFlinger::mLock held 4507AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4508{ 4509 PlaybackThread *thread = checkPlaybackThread_l(output); 4510 if (thread != NULL) { 4511 if (thread->type() == PlaybackThread::DIRECT) { 4512 thread = NULL; 4513 } 4514 } 4515 return (MixerThread *)thread; 4516} 4517 4518// checkRecordThread_l() must be called with AudioFlinger::mLock held 4519AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4520{ 4521 RecordThread *thread = NULL; 4522 if (mRecordThreads.indexOfKey(input) >= 0) { 4523 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4524 } 4525 return thread; 4526} 4527 4528// nextUniqueId_l() must be called with AudioFlinger::mLock held 4529int AudioFlinger::nextUniqueId_l() 4530{ 4531 return mNextUniqueId++; 4532} 4533 4534// ---------------------------------------------------------------------------- 4535// Effect management 4536// ---------------------------------------------------------------------------- 4537 4538 4539status_t AudioFlinger::loadEffectLibrary(const char *libPath, int *handle) 4540{ 4541 // check calling permissions 4542 if (!settingsAllowed()) { 4543 return PERMISSION_DENIED; 4544 } 4545 // only allow libraries loaded from /system/lib/soundfx for now 4546 if (strncmp(gEffectLibPath, libPath, strlen(gEffectLibPath)) != 0) { 4547 return PERMISSION_DENIED; 4548 } 4549 4550 Mutex::Autolock _l(mLock); 4551 return EffectLoadLibrary(libPath, handle); 4552} 4553 4554status_t AudioFlinger::unloadEffectLibrary(int handle) 4555{ 4556 // check calling permissions 4557 if (!settingsAllowed()) { 4558 return PERMISSION_DENIED; 4559 } 4560 4561 Mutex::Autolock _l(mLock); 4562 return EffectUnloadLibrary(handle); 4563} 4564 4565status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4566{ 4567 Mutex::Autolock _l(mLock); 4568 return EffectQueryNumberEffects(numEffects); 4569} 4570 4571status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4572{ 4573 Mutex::Autolock _l(mLock); 4574 return EffectQueryEffect(index, descriptor); 4575} 4576 4577status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4578{ 4579 Mutex::Autolock _l(mLock); 4580 return EffectGetDescriptor(pUuid, descriptor); 4581} 4582 4583 4584// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4585static const effect_uuid_t VISUALIZATION_UUID_ = 4586 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4587 4588sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4589 effect_descriptor_t *pDesc, 4590 const sp<IEffectClient>& effectClient, 4591 int32_t priority, 4592 int output, 4593 int sessionId, 4594 status_t *status, 4595 int *id, 4596 int *enabled) 4597{ 4598 status_t lStatus = NO_ERROR; 4599 sp<EffectHandle> handle; 4600 effect_interface_t itfe; 4601 effect_descriptor_t desc; 4602 sp<Client> client; 4603 wp<Client> wclient; 4604 4605 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", 4606 pid, effectClient.get(), priority, sessionId, output); 4607 4608 if (pDesc == NULL) { 4609 lStatus = BAD_VALUE; 4610 goto Exit; 4611 } 4612 4613 // check audio settings permission for global effects 4614 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 4615 lStatus = PERMISSION_DENIED; 4616 goto Exit; 4617 } 4618 4619 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 4620 // that can only be created by audio policy manager (running in same process) 4621 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 4622 lStatus = PERMISSION_DENIED; 4623 goto Exit; 4624 } 4625 4626 // check recording permission for visualizer 4627 if ((memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4628 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) && 4629 !recordingAllowed()) { 4630 lStatus = PERMISSION_DENIED; 4631 goto Exit; 4632 } 4633 4634 if (output == 0) { 4635 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 4636 // output must be specified by AudioPolicyManager when using session 4637 // AUDIO_SESSION_OUTPUT_STAGE 4638 lStatus = BAD_VALUE; 4639 goto Exit; 4640 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 4641 // if the output returned by getOutputForEffect() is removed before we lock the 4642 // mutex below, the call to checkPlaybackThread_l(output) below will detect it 4643 // and we will exit safely 4644 output = AudioSystem::getOutputForEffect(&desc); 4645 } 4646 } 4647 4648 { 4649 Mutex::Autolock _l(mLock); 4650 4651 4652 if (!EffectIsNullUuid(&pDesc->uuid)) { 4653 // if uuid is specified, request effect descriptor 4654 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4655 if (lStatus < 0) { 4656 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4657 goto Exit; 4658 } 4659 } else { 4660 // if uuid is not specified, look for an available implementation 4661 // of the required type in effect factory 4662 if (EffectIsNullUuid(&pDesc->type)) { 4663 LOGW("createEffect() no effect type"); 4664 lStatus = BAD_VALUE; 4665 goto Exit; 4666 } 4667 uint32_t numEffects = 0; 4668 effect_descriptor_t d; 4669 bool found = false; 4670 4671 lStatus = EffectQueryNumberEffects(&numEffects); 4672 if (lStatus < 0) { 4673 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4674 goto Exit; 4675 } 4676 for (uint32_t i = 0; i < numEffects; i++) { 4677 lStatus = EffectQueryEffect(i, &desc); 4678 if (lStatus < 0) { 4679 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4680 continue; 4681 } 4682 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4683 // If matching type found save effect descriptor. If the session is 4684 // 0 and the effect is not auxiliary, continue enumeration in case 4685 // an auxiliary version of this effect type is available 4686 found = true; 4687 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4688 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 4689 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4690 break; 4691 } 4692 } 4693 } 4694 if (!found) { 4695 lStatus = BAD_VALUE; 4696 LOGW("createEffect() effect not found"); 4697 goto Exit; 4698 } 4699 // For same effect type, chose auxiliary version over insert version if 4700 // connect to output mix (Compliance to OpenSL ES) 4701 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 4702 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4703 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4704 } 4705 } 4706 4707 // Do not allow auxiliary effects on a session different from 0 (output mix) 4708 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 4709 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4710 lStatus = INVALID_OPERATION; 4711 goto Exit; 4712 } 4713 4714 // return effect descriptor 4715 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4716 4717 // If output is not specified try to find a matching audio session ID in one of the 4718 // output threads. 4719 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 4720 // because of code checking output when entering the function. 4721 if (output == 0) { 4722 // look for the thread where the specified audio session is present 4723 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4724 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4725 output = mPlaybackThreads.keyAt(i); 4726 break; 4727 } 4728 } 4729 // If no output thread contains the requested session ID, default to 4730 // first output. The effect chain will be moved to the correct output 4731 // thread when a track with the same session ID is created 4732 if (output == 0 && mPlaybackThreads.size()) { 4733 output = mPlaybackThreads.keyAt(0); 4734 } 4735 } 4736 LOGV("createEffect() got output %d for effect %s", output, desc.name); 4737 PlaybackThread *thread = checkPlaybackThread_l(output); 4738 if (thread == NULL) { 4739 LOGE("createEffect() unknown output thread"); 4740 lStatus = BAD_VALUE; 4741 goto Exit; 4742 } 4743 4744 // TODO: allow attachment of effect to inputs 4745 4746 wclient = mClients.valueFor(pid); 4747 4748 if (wclient != NULL) { 4749 client = wclient.promote(); 4750 } else { 4751 client = new Client(this, pid); 4752 mClients.add(pid, client); 4753 } 4754 4755 // create effect on selected output trhead 4756 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4757 &desc, enabled, &lStatus); 4758 if (handle != 0 && id != NULL) { 4759 *id = handle->id(); 4760 } 4761 } 4762 4763Exit: 4764 if(status) { 4765 *status = lStatus; 4766 } 4767 return handle; 4768} 4769 4770status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4771{ 4772 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4773 session, srcOutput, dstOutput); 4774 Mutex::Autolock _l(mLock); 4775 if (srcOutput == dstOutput) { 4776 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4777 return NO_ERROR; 4778 } 4779 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4780 if (srcThread == NULL) { 4781 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4782 return BAD_VALUE; 4783 } 4784 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 4785 if (dstThread == NULL) { 4786 LOGW("moveEffects() bad dstOutput %d", dstOutput); 4787 return BAD_VALUE; 4788 } 4789 4790 Mutex::Autolock _dl(dstThread->mLock); 4791 Mutex::Autolock _sl(srcThread->mLock); 4792 moveEffectChain_l(session, srcThread, dstThread, false); 4793 4794 return NO_ERROR; 4795} 4796 4797// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 4798status_t AudioFlinger::moveEffectChain_l(int session, 4799 AudioFlinger::PlaybackThread *srcThread, 4800 AudioFlinger::PlaybackThread *dstThread, 4801 bool reRegister) 4802{ 4803 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 4804 session, srcThread, dstThread); 4805 4806 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 4807 if (chain == 0) { 4808 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 4809 session, srcThread); 4810 return INVALID_OPERATION; 4811 } 4812 4813 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 4814 // so that a new chain is created with correct parameters when first effect is added. This is 4815 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 4816 // removed. 4817 srcThread->removeEffectChain_l(chain); 4818 4819 // transfer all effects one by one so that new effect chain is created on new thread with 4820 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 4821 int dstOutput = dstThread->id(); 4822 sp<EffectChain> dstChain; 4823 uint32_t strategy; 4824 sp<EffectModule> effect = chain->getEffectFromId_l(0); 4825 while (effect != 0) { 4826 srcThread->removeEffect_l(effect); 4827 dstThread->addEffect_l(effect); 4828 // if the move request is not received from audio policy manager, the effect must be 4829 // re-registered with the new strategy and output 4830 if (dstChain == 0) { 4831 dstChain = effect->chain().promote(); 4832 if (dstChain == 0) { 4833 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 4834 srcThread->addEffect_l(effect); 4835 return NO_INIT; 4836 } 4837 strategy = dstChain->strategy(); 4838 } 4839 if (reRegister) { 4840 AudioSystem::unregisterEffect(effect->id()); 4841 AudioSystem::registerEffect(&effect->desc(), 4842 dstOutput, 4843 strategy, 4844 session, 4845 effect->id()); 4846 } 4847 effect = chain->getEffectFromId_l(0); 4848 } 4849 4850 return NO_ERROR; 4851} 4852 4853// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 4854sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( 4855 const sp<AudioFlinger::Client>& client, 4856 const sp<IEffectClient>& effectClient, 4857 int32_t priority, 4858 int sessionId, 4859 effect_descriptor_t *desc, 4860 int *enabled, 4861 status_t *status 4862 ) 4863{ 4864 sp<EffectModule> effect; 4865 sp<EffectHandle> handle; 4866 status_t lStatus; 4867 sp<Track> track; 4868 sp<EffectChain> chain; 4869 bool chainCreated = false; 4870 bool effectCreated = false; 4871 bool effectRegistered = false; 4872 4873 if (mOutput == 0) { 4874 LOGW("createEffect_l() Audio driver not initialized."); 4875 lStatus = NO_INIT; 4876 goto Exit; 4877 } 4878 4879 // Do not allow auxiliary effect on session other than 0 4880 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && 4881 sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4882 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4883 desc->name, sessionId); 4884 lStatus = BAD_VALUE; 4885 goto Exit; 4886 } 4887 4888 // Do not allow effects with session ID 0 on direct output or duplicating threads 4889 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 4890 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 4891 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4892 desc->name, sessionId); 4893 lStatus = BAD_VALUE; 4894 goto Exit; 4895 } 4896 4897 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 4898 4899 { // scope for mLock 4900 Mutex::Autolock _l(mLock); 4901 4902 // check for existing effect chain with the requested audio session 4903 chain = getEffectChain_l(sessionId); 4904 if (chain == 0) { 4905 // create a new chain for this session 4906 LOGV("createEffect_l() new effect chain for session %d", sessionId); 4907 chain = new EffectChain(this, sessionId); 4908 addEffectChain_l(chain); 4909 chain->setStrategy(getStrategyForSession_l(sessionId)); 4910 chainCreated = true; 4911 } else { 4912 effect = chain->getEffectFromDesc_l(desc); 4913 } 4914 4915 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 4916 4917 if (effect == 0) { 4918 int id = mAudioFlinger->nextUniqueId_l(); 4919 // Check CPU and memory usage 4920 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 4921 if (lStatus != NO_ERROR) { 4922 goto Exit; 4923 } 4924 effectRegistered = true; 4925 // create a new effect module if none present in the chain 4926 effect = new EffectModule(this, chain, desc, id, sessionId); 4927 lStatus = effect->status(); 4928 if (lStatus != NO_ERROR) { 4929 goto Exit; 4930 } 4931 lStatus = chain->addEffect_l(effect); 4932 if (lStatus != NO_ERROR) { 4933 goto Exit; 4934 } 4935 effectCreated = true; 4936 4937 effect->setDevice(mDevice); 4938 effect->setMode(mAudioFlinger->getMode()); 4939 } 4940 // create effect handle and connect it to effect module 4941 handle = new EffectHandle(effect, client, effectClient, priority); 4942 lStatus = effect->addHandle(handle); 4943 if (enabled) { 4944 *enabled = (int)effect->isEnabled(); 4945 } 4946 } 4947 4948Exit: 4949 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 4950 Mutex::Autolock _l(mLock); 4951 if (effectCreated) { 4952 chain->removeEffect_l(effect); 4953 } 4954 if (effectRegistered) { 4955 AudioSystem::unregisterEffect(effect->id()); 4956 } 4957 if (chainCreated) { 4958 removeEffectChain_l(chain); 4959 } 4960 handle.clear(); 4961 } 4962 4963 if(status) { 4964 *status = lStatus; 4965 } 4966 return handle; 4967} 4968 4969// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 4970// PlaybackThread::mLock held 4971status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect) 4972{ 4973 // check for existing effect chain with the requested audio session 4974 int sessionId = effect->sessionId(); 4975 sp<EffectChain> chain = getEffectChain_l(sessionId); 4976 bool chainCreated = false; 4977 4978 if (chain == 0) { 4979 // create a new chain for this session 4980 LOGV("addEffect_l() new effect chain for session %d", sessionId); 4981 chain = new EffectChain(this, sessionId); 4982 addEffectChain_l(chain); 4983 chain->setStrategy(getStrategyForSession_l(sessionId)); 4984 chainCreated = true; 4985 } 4986 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 4987 4988 if (chain->getEffectFromId_l(effect->id()) != 0) { 4989 LOGW("addEffect_l() %p effect %s already present in chain %p", 4990 this, effect->desc().name, chain.get()); 4991 return BAD_VALUE; 4992 } 4993 4994 status_t status = chain->addEffect_l(effect); 4995 if (status != NO_ERROR) { 4996 if (chainCreated) { 4997 removeEffectChain_l(chain); 4998 } 4999 return status; 5000 } 5001 5002 effect->setDevice(mDevice); 5003 effect->setMode(mAudioFlinger->getMode()); 5004 return NO_ERROR; 5005} 5006 5007void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) { 5008 5009 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5010 effect_descriptor_t desc = effect->desc(); 5011 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5012 detachAuxEffect_l(effect->id()); 5013 } 5014 5015 sp<EffectChain> chain = effect->chain().promote(); 5016 if (chain != 0) { 5017 // remove effect chain if removing last effect 5018 if (chain->removeEffect_l(effect) == 0) { 5019 removeEffectChain_l(chain); 5020 } 5021 } else { 5022 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5023 } 5024} 5025 5026void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect, 5027 const wp<EffectHandle>& handle) { 5028 Mutex::Autolock _l(mLock); 5029 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5030 // delete the effect module if removing last handle on it 5031 if (effect->removeHandle(handle) == 0) { 5032 removeEffect_l(effect); 5033 AudioSystem::unregisterEffect(effect->id()); 5034 } 5035} 5036 5037status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5038{ 5039 int session = chain->sessionId(); 5040 int16_t *buffer = mMixBuffer; 5041 bool ownsBuffer = false; 5042 5043 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5044 if (session > 0) { 5045 // Only one effect chain can be present in direct output thread and it uses 5046 // the mix buffer as input 5047 if (mType != DIRECT) { 5048 size_t numSamples = mFrameCount * mChannelCount; 5049 buffer = new int16_t[numSamples]; 5050 memset(buffer, 0, numSamples * sizeof(int16_t)); 5051 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5052 ownsBuffer = true; 5053 } 5054 5055 // Attach all tracks with same session ID to this chain. 5056 for (size_t i = 0; i < mTracks.size(); ++i) { 5057 sp<Track> track = mTracks[i]; 5058 if (session == track->sessionId()) { 5059 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5060 track->setMainBuffer(buffer); 5061 } 5062 } 5063 5064 // indicate all active tracks in the chain 5065 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5066 sp<Track> track = mActiveTracks[i].promote(); 5067 if (track == 0) continue; 5068 if (session == track->sessionId()) { 5069 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5070 chain->startTrack(); 5071 } 5072 } 5073 } 5074 5075 chain->setInBuffer(buffer, ownsBuffer); 5076 chain->setOutBuffer(mMixBuffer); 5077 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5078 // chains list in order to be processed last as it contains output stage effects 5079 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5080 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5081 // after track specific effects and before output stage 5082 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5083 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5084 // Effect chain for other sessions are inserted at beginning of effect 5085 // chains list to be processed before output mix effects. Relative order between other 5086 // sessions is not important 5087 size_t size = mEffectChains.size(); 5088 size_t i = 0; 5089 for (i = 0; i < size; i++) { 5090 if (mEffectChains[i]->sessionId() < session) break; 5091 } 5092 mEffectChains.insertAt(chain, i); 5093 5094 return NO_ERROR; 5095} 5096 5097size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5098{ 5099 int session = chain->sessionId(); 5100 5101 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5102 5103 for (size_t i = 0; i < mEffectChains.size(); i++) { 5104 if (chain == mEffectChains[i]) { 5105 mEffectChains.removeAt(i); 5106 // detach all tracks with same session ID from this chain 5107 for (size_t i = 0; i < mTracks.size(); ++i) { 5108 sp<Track> track = mTracks[i]; 5109 if (session == track->sessionId()) { 5110 track->setMainBuffer(mMixBuffer); 5111 } 5112 } 5113 break; 5114 } 5115 } 5116 return mEffectChains.size(); 5117} 5118 5119void AudioFlinger::PlaybackThread::lockEffectChains_l( 5120 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5121{ 5122 effectChains = mEffectChains; 5123 for (size_t i = 0; i < mEffectChains.size(); i++) { 5124 mEffectChains[i]->lock(); 5125 } 5126} 5127 5128void AudioFlinger::PlaybackThread::unlockEffectChains( 5129 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5130{ 5131 for (size_t i = 0; i < effectChains.size(); i++) { 5132 effectChains[i]->unlock(); 5133 } 5134} 5135 5136 5137sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) 5138{ 5139 sp<EffectModule> effect; 5140 5141 sp<EffectChain> chain = getEffectChain_l(sessionId); 5142 if (chain != 0) { 5143 effect = chain->getEffectFromId_l(effectId); 5144 } 5145 return effect; 5146} 5147 5148status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5149 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5150{ 5151 Mutex::Autolock _l(mLock); 5152 return attachAuxEffect_l(track, EffectId); 5153} 5154 5155status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5156 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5157{ 5158 status_t status = NO_ERROR; 5159 5160 if (EffectId == 0) { 5161 track->setAuxBuffer(0, NULL); 5162 } else { 5163 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 5164 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 5165 if (effect != 0) { 5166 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5167 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5168 } else { 5169 status = INVALID_OPERATION; 5170 } 5171 } else { 5172 status = BAD_VALUE; 5173 } 5174 } 5175 return status; 5176} 5177 5178void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5179{ 5180 for (size_t i = 0; i < mTracks.size(); ++i) { 5181 sp<Track> track = mTracks[i]; 5182 if (track->auxEffectId() == effectId) { 5183 attachAuxEffect_l(track, 0); 5184 } 5185 } 5186} 5187 5188// ---------------------------------------------------------------------------- 5189// EffectModule implementation 5190// ---------------------------------------------------------------------------- 5191 5192#undef LOG_TAG 5193#define LOG_TAG "AudioFlinger::EffectModule" 5194 5195AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5196 const wp<AudioFlinger::EffectChain>& chain, 5197 effect_descriptor_t *desc, 5198 int id, 5199 int sessionId) 5200 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5201 mStatus(NO_INIT), mState(IDLE) 5202{ 5203 LOGV("Constructor %p", this); 5204 int lStatus; 5205 sp<ThreadBase> thread = mThread.promote(); 5206 if (thread == 0) { 5207 return; 5208 } 5209 PlaybackThread *p = (PlaybackThread *)thread.get(); 5210 5211 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5212 5213 // create effect engine from effect factory 5214 mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); 5215 5216 if (mStatus != NO_ERROR) { 5217 return; 5218 } 5219 lStatus = init(); 5220 if (lStatus < 0) { 5221 mStatus = lStatus; 5222 goto Error; 5223 } 5224 5225 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5226 return; 5227Error: 5228 EffectRelease(mEffectInterface); 5229 mEffectInterface = NULL; 5230 LOGV("Constructor Error %d", mStatus); 5231} 5232 5233AudioFlinger::EffectModule::~EffectModule() 5234{ 5235 LOGV("Destructor %p", this); 5236 if (mEffectInterface != NULL) { 5237 // release effect engine 5238 EffectRelease(mEffectInterface); 5239 } 5240} 5241 5242status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5243{ 5244 status_t status; 5245 5246 Mutex::Autolock _l(mLock); 5247 // First handle in mHandles has highest priority and controls the effect module 5248 int priority = handle->priority(); 5249 size_t size = mHandles.size(); 5250 sp<EffectHandle> h; 5251 size_t i; 5252 for (i = 0; i < size; i++) { 5253 h = mHandles[i].promote(); 5254 if (h == 0) continue; 5255 if (h->priority() <= priority) break; 5256 } 5257 // if inserted in first place, move effect control from previous owner to this handle 5258 if (i == 0) { 5259 if (h != 0) { 5260 h->setControl(false, true); 5261 } 5262 handle->setControl(true, false); 5263 status = NO_ERROR; 5264 } else { 5265 status = ALREADY_EXISTS; 5266 } 5267 mHandles.insertAt(handle, i); 5268 return status; 5269} 5270 5271size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5272{ 5273 Mutex::Autolock _l(mLock); 5274 size_t size = mHandles.size(); 5275 size_t i; 5276 for (i = 0; i < size; i++) { 5277 if (mHandles[i] == handle) break; 5278 } 5279 if (i == size) { 5280 return size; 5281 } 5282 mHandles.removeAt(i); 5283 size = mHandles.size(); 5284 // if removed from first place, move effect control from this handle to next in line 5285 if (i == 0 && size != 0) { 5286 sp<EffectHandle> h = mHandles[0].promote(); 5287 if (h != 0) { 5288 h->setControl(true, true); 5289 } 5290 } 5291 5292 // Release effect engine here so that it is done immediately. Otherwise it will be released 5293 // by the destructor when the last strong reference on the this object is released which can 5294 // happen after next process is called on this effect. 5295 if (size == 0 && mEffectInterface != NULL) { 5296 // release effect engine 5297 EffectRelease(mEffectInterface); 5298 mEffectInterface = NULL; 5299 } 5300 5301 return size; 5302} 5303 5304void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5305{ 5306 // keep a strong reference on this EffectModule to avoid calling the 5307 // destructor before we exit 5308 sp<EffectModule> keep(this); 5309 { 5310 sp<ThreadBase> thread = mThread.promote(); 5311 if (thread != 0) { 5312 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5313 playbackThread->disconnectEffect(keep, handle); 5314 } 5315 } 5316} 5317 5318void AudioFlinger::EffectModule::updateState() { 5319 Mutex::Autolock _l(mLock); 5320 5321 switch (mState) { 5322 case RESTART: 5323 reset_l(); 5324 // FALL THROUGH 5325 5326 case STARTING: 5327 // clear auxiliary effect input buffer for next accumulation 5328 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5329 memset(mConfig.inputCfg.buffer.raw, 5330 0, 5331 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5332 } 5333 start_l(); 5334 mState = ACTIVE; 5335 break; 5336 case STOPPING: 5337 stop_l(); 5338 mDisableWaitCnt = mMaxDisableWaitCnt; 5339 mState = STOPPED; 5340 break; 5341 case STOPPED: 5342 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5343 // turn off sequence. 5344 if (--mDisableWaitCnt == 0) { 5345 reset_l(); 5346 mState = IDLE; 5347 } 5348 break; 5349 default: //IDLE , ACTIVE 5350 break; 5351 } 5352} 5353 5354void AudioFlinger::EffectModule::process() 5355{ 5356 Mutex::Autolock _l(mLock); 5357 5358 if (mEffectInterface == NULL || 5359 mConfig.inputCfg.buffer.raw == NULL || 5360 mConfig.outputCfg.buffer.raw == NULL) { 5361 return; 5362 } 5363 5364 if (isProcessEnabled()) { 5365 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5366 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5367 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5368 mConfig.inputCfg.buffer.s32, 5369 mConfig.inputCfg.buffer.frameCount/2); 5370 } 5371 5372 // do the actual processing in the effect engine 5373 int ret = (*mEffectInterface)->process(mEffectInterface, 5374 &mConfig.inputCfg.buffer, 5375 &mConfig.outputCfg.buffer); 5376 5377 // force transition to IDLE state when engine is ready 5378 if (mState == STOPPED && ret == -ENODATA) { 5379 mDisableWaitCnt = 1; 5380 } 5381 5382 // clear auxiliary effect input buffer for next accumulation 5383 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5384 memset(mConfig.inputCfg.buffer.raw, 0, 5385 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5386 } 5387 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5388 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5389 // If an insert effect is idle and input buffer is different from output buffer, 5390 // accumulate input onto output 5391 sp<EffectChain> chain = mChain.promote(); 5392 if (chain != 0 && chain->activeTracks() != 0) { 5393 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 5394 int16_t *in = mConfig.inputCfg.buffer.s16; 5395 int16_t *out = mConfig.outputCfg.buffer.s16; 5396 for (size_t i = 0; i < frameCnt; i++) { 5397 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 5398 } 5399 } 5400 } 5401} 5402 5403void AudioFlinger::EffectModule::reset_l() 5404{ 5405 if (mEffectInterface == NULL) { 5406 return; 5407 } 5408 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5409} 5410 5411status_t AudioFlinger::EffectModule::configure() 5412{ 5413 uint32_t channels; 5414 if (mEffectInterface == NULL) { 5415 return NO_INIT; 5416 } 5417 5418 sp<ThreadBase> thread = mThread.promote(); 5419 if (thread == 0) { 5420 return DEAD_OBJECT; 5421 } 5422 5423 // TODO: handle configuration of effects replacing track process 5424 if (thread->channelCount() == 1) { 5425 channels = CHANNEL_MONO; 5426 } else { 5427 channels = CHANNEL_STEREO; 5428 } 5429 5430 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5431 mConfig.inputCfg.channels = CHANNEL_MONO; 5432 } else { 5433 mConfig.inputCfg.channels = channels; 5434 } 5435 mConfig.outputCfg.channels = channels; 5436 mConfig.inputCfg.format = SAMPLE_FORMAT_PCM_S15; 5437 mConfig.outputCfg.format = SAMPLE_FORMAT_PCM_S15; 5438 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5439 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5440 mConfig.inputCfg.bufferProvider.cookie = NULL; 5441 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5442 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5443 mConfig.outputCfg.bufferProvider.cookie = NULL; 5444 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5445 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5446 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5447 // Insert effect: 5448 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 5449 // always overwrites output buffer: input buffer == output buffer 5450 // - in other sessions: 5451 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5452 // other effect: overwrites output buffer: input buffer == output buffer 5453 // Auxiliary effect: 5454 // accumulates in output buffer: input buffer != output buffer 5455 // Therefore: accumulate <=> input buffer != output buffer 5456 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5457 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5458 } else { 5459 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5460 } 5461 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5462 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5463 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5464 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5465 5466 LOGV("configure() %p thread %p buffer %p framecount %d", 5467 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5468 5469 status_t cmdStatus; 5470 uint32_t size = sizeof(int); 5471 status_t status = (*mEffectInterface)->command(mEffectInterface, 5472 EFFECT_CMD_CONFIGURE, 5473 sizeof(effect_config_t), 5474 &mConfig, 5475 &size, 5476 &cmdStatus); 5477 if (status == 0) { 5478 status = cmdStatus; 5479 } 5480 5481 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5482 (1000 * mConfig.outputCfg.buffer.frameCount); 5483 5484 return status; 5485} 5486 5487status_t AudioFlinger::EffectModule::init() 5488{ 5489 Mutex::Autolock _l(mLock); 5490 if (mEffectInterface == NULL) { 5491 return NO_INIT; 5492 } 5493 status_t cmdStatus; 5494 uint32_t size = sizeof(status_t); 5495 status_t status = (*mEffectInterface)->command(mEffectInterface, 5496 EFFECT_CMD_INIT, 5497 0, 5498 NULL, 5499 &size, 5500 &cmdStatus); 5501 if (status == 0) { 5502 status = cmdStatus; 5503 } 5504 return status; 5505} 5506 5507status_t AudioFlinger::EffectModule::start_l() 5508{ 5509 if (mEffectInterface == NULL) { 5510 return NO_INIT; 5511 } 5512 status_t cmdStatus; 5513 uint32_t size = sizeof(status_t); 5514 status_t status = (*mEffectInterface)->command(mEffectInterface, 5515 EFFECT_CMD_ENABLE, 5516 0, 5517 NULL, 5518 &size, 5519 &cmdStatus); 5520 if (status == 0) { 5521 status = cmdStatus; 5522 } 5523 return status; 5524} 5525 5526status_t AudioFlinger::EffectModule::stop_l() 5527{ 5528 if (mEffectInterface == NULL) { 5529 return NO_INIT; 5530 } 5531 status_t cmdStatus; 5532 uint32_t size = sizeof(status_t); 5533 status_t status = (*mEffectInterface)->command(mEffectInterface, 5534 EFFECT_CMD_DISABLE, 5535 0, 5536 NULL, 5537 &size, 5538 &cmdStatus); 5539 if (status == 0) { 5540 status = cmdStatus; 5541 } 5542 return status; 5543} 5544 5545status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5546 uint32_t cmdSize, 5547 void *pCmdData, 5548 uint32_t *replySize, 5549 void *pReplyData) 5550{ 5551 Mutex::Autolock _l(mLock); 5552// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5553 5554 if (mEffectInterface == NULL) { 5555 return NO_INIT; 5556 } 5557 status_t status = (*mEffectInterface)->command(mEffectInterface, 5558 cmdCode, 5559 cmdSize, 5560 pCmdData, 5561 replySize, 5562 pReplyData); 5563 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5564 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5565 for (size_t i = 1; i < mHandles.size(); i++) { 5566 sp<EffectHandle> h = mHandles[i].promote(); 5567 if (h != 0) { 5568 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5569 } 5570 } 5571 } 5572 return status; 5573} 5574 5575status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5576{ 5577 Mutex::Autolock _l(mLock); 5578 LOGV("setEnabled %p enabled %d", this, enabled); 5579 5580 if (enabled != isEnabled()) { 5581 switch (mState) { 5582 // going from disabled to enabled 5583 case IDLE: 5584 mState = STARTING; 5585 break; 5586 case STOPPED: 5587 mState = RESTART; 5588 break; 5589 case STOPPING: 5590 mState = ACTIVE; 5591 break; 5592 5593 // going from enabled to disabled 5594 case RESTART: 5595 mState = STOPPED; 5596 break; 5597 case STARTING: 5598 mState = IDLE; 5599 break; 5600 case ACTIVE: 5601 mState = STOPPING; 5602 break; 5603 } 5604 for (size_t i = 1; i < mHandles.size(); i++) { 5605 sp<EffectHandle> h = mHandles[i].promote(); 5606 if (h != 0) { 5607 h->setEnabled(enabled); 5608 } 5609 } 5610 } 5611 return NO_ERROR; 5612} 5613 5614bool AudioFlinger::EffectModule::isEnabled() 5615{ 5616 switch (mState) { 5617 case RESTART: 5618 case STARTING: 5619 case ACTIVE: 5620 return true; 5621 case IDLE: 5622 case STOPPING: 5623 case STOPPED: 5624 default: 5625 return false; 5626 } 5627} 5628 5629bool AudioFlinger::EffectModule::isProcessEnabled() 5630{ 5631 switch (mState) { 5632 case RESTART: 5633 case ACTIVE: 5634 case STOPPING: 5635 case STOPPED: 5636 return true; 5637 case IDLE: 5638 case STARTING: 5639 default: 5640 return false; 5641 } 5642} 5643 5644status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5645{ 5646 Mutex::Autolock _l(mLock); 5647 status_t status = NO_ERROR; 5648 5649 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5650 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5651 if (isProcessEnabled() && 5652 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5653 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5654 status_t cmdStatus; 5655 uint32_t volume[2]; 5656 uint32_t *pVolume = NULL; 5657 uint32_t size = sizeof(volume); 5658 volume[0] = *left; 5659 volume[1] = *right; 5660 if (controller) { 5661 pVolume = volume; 5662 } 5663 status = (*mEffectInterface)->command(mEffectInterface, 5664 EFFECT_CMD_SET_VOLUME, 5665 size, 5666 volume, 5667 &size, 5668 pVolume); 5669 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5670 *left = volume[0]; 5671 *right = volume[1]; 5672 } 5673 } 5674 return status; 5675} 5676 5677status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5678{ 5679 Mutex::Autolock _l(mLock); 5680 status_t status = NO_ERROR; 5681 if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5682 // convert device bit field from AudioSystem to EffectApi format. 5683 device = deviceAudioSystemToEffectApi(device); 5684 if (device == 0) { 5685 return BAD_VALUE; 5686 } 5687 status_t cmdStatus; 5688 uint32_t size = sizeof(status_t); 5689 status = (*mEffectInterface)->command(mEffectInterface, 5690 EFFECT_CMD_SET_DEVICE, 5691 sizeof(uint32_t), 5692 &device, 5693 &size, 5694 &cmdStatus); 5695 if (status == NO_ERROR) { 5696 status = cmdStatus; 5697 } 5698 } 5699 return status; 5700} 5701 5702status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 5703{ 5704 Mutex::Autolock _l(mLock); 5705 status_t status = NO_ERROR; 5706 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 5707 // convert audio mode from AudioSystem to EffectApi format. 5708 int effectMode = modeAudioSystemToEffectApi(mode); 5709 if (effectMode < 0) { 5710 return BAD_VALUE; 5711 } 5712 status_t cmdStatus; 5713 uint32_t size = sizeof(status_t); 5714 status = (*mEffectInterface)->command(mEffectInterface, 5715 EFFECT_CMD_SET_AUDIO_MODE, 5716 sizeof(int), 5717 &effectMode, 5718 &size, 5719 &cmdStatus); 5720 if (status == NO_ERROR) { 5721 status = cmdStatus; 5722 } 5723 } 5724 return status; 5725} 5726 5727// update this table when AudioSystem::audio_devices or audio_device_e (in EffectApi.h) are modified 5728const uint32_t AudioFlinger::EffectModule::sDeviceConvTable[] = { 5729 DEVICE_EARPIECE, // AUDIO_DEVICE_OUT_EARPIECE 5730 DEVICE_SPEAKER, // AUDIO_DEVICE_OUT_SPEAKER 5731 DEVICE_WIRED_HEADSET, // case AUDIO_DEVICE_OUT_WIRED_HEADSET 5732 DEVICE_WIRED_HEADPHONE, // AUDIO_DEVICE_OUT_WIRED_HEADPHONE 5733 DEVICE_BLUETOOTH_SCO, // AUDIO_DEVICE_OUT_BLUETOOTH_SCO 5734 DEVICE_BLUETOOTH_SCO_HEADSET, // AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET 5735 DEVICE_BLUETOOTH_SCO_CARKIT, // AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT 5736 DEVICE_BLUETOOTH_A2DP, // AUDIO_DEVICE_OUT_BLUETOOTH_A2DP 5737 DEVICE_BLUETOOTH_A2DP_HEADPHONES, // AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES 5738 DEVICE_BLUETOOTH_A2DP_SPEAKER, // AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER 5739 DEVICE_AUX_DIGITAL // AUDIO_DEVICE_OUT_AUX_DIGITAL 5740}; 5741 5742uint32_t AudioFlinger::EffectModule::deviceAudioSystemToEffectApi(uint32_t device) 5743{ 5744 uint32_t deviceOut = 0; 5745 while (device) { 5746 const uint32_t i = 31 - __builtin_clz(device); 5747 device &= ~(1 << i); 5748 if (i >= sizeof(sDeviceConvTable)/sizeof(uint32_t)) { 5749 LOGE("device conversion error for AudioSystem device 0x%08x", device); 5750 return 0; 5751 } 5752 deviceOut |= (uint32_t)sDeviceConvTable[i]; 5753 } 5754 return deviceOut; 5755} 5756 5757// update this table when AudioSystem::audio_mode or audio_mode_e (in EffectApi.h) are modified 5758const uint32_t AudioFlinger::EffectModule::sModeConvTable[] = { 5759 AUDIO_EFFECT_MODE_NORMAL, // AUDIO_MODE_NORMAL 5760 AUDIO_EFFECT_MODE_RINGTONE, // AUDIO_MODE_RINGTONE 5761 AUDIO_EFFECT_MODE_IN_CALL, // AUDIO_MODE_IN_CALL 5762 AUDIO_EFFECT_MODE_IN_CALL // AUDIO_MODE_IN_COMMUNICATION, same conversion as for AUDIO_MODE_IN_CALL 5763}; 5764 5765int AudioFlinger::EffectModule::modeAudioSystemToEffectApi(uint32_t mode) 5766{ 5767 int modeOut = -1; 5768 if (mode < sizeof(sModeConvTable) / sizeof(uint32_t)) { 5769 modeOut = (int)sModeConvTable[mode]; 5770 } 5771 return modeOut; 5772} 5773 5774status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 5775{ 5776 const size_t SIZE = 256; 5777 char buffer[SIZE]; 5778 String8 result; 5779 5780 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 5781 result.append(buffer); 5782 5783 bool locked = tryLock(mLock); 5784 // failed to lock - AudioFlinger is probably deadlocked 5785 if (!locked) { 5786 result.append("\t\tCould not lock Fx mutex:\n"); 5787 } 5788 5789 result.append("\t\tSession Status State Engine:\n"); 5790 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 5791 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 5792 result.append(buffer); 5793 5794 result.append("\t\tDescriptor:\n"); 5795 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5796 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 5797 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 5798 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 5799 result.append(buffer); 5800 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5801 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 5802 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 5803 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 5804 result.append(buffer); 5805 snprintf(buffer, SIZE, "\t\t- apiVersion: %04X\n\t\t- flags: %08X\n", 5806 mDescriptor.apiVersion, 5807 mDescriptor.flags); 5808 result.append(buffer); 5809 snprintf(buffer, SIZE, "\t\t- name: %s\n", 5810 mDescriptor.name); 5811 result.append(buffer); 5812 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 5813 mDescriptor.implementor); 5814 result.append(buffer); 5815 5816 result.append("\t\t- Input configuration:\n"); 5817 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5818 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5819 (uint32_t)mConfig.inputCfg.buffer.raw, 5820 mConfig.inputCfg.buffer.frameCount, 5821 mConfig.inputCfg.samplingRate, 5822 mConfig.inputCfg.channels, 5823 mConfig.inputCfg.format); 5824 result.append(buffer); 5825 5826 result.append("\t\t- Output configuration:\n"); 5827 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5828 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5829 (uint32_t)mConfig.outputCfg.buffer.raw, 5830 mConfig.outputCfg.buffer.frameCount, 5831 mConfig.outputCfg.samplingRate, 5832 mConfig.outputCfg.channels, 5833 mConfig.outputCfg.format); 5834 result.append(buffer); 5835 5836 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 5837 result.append(buffer); 5838 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 5839 for (size_t i = 0; i < mHandles.size(); ++i) { 5840 sp<EffectHandle> handle = mHandles[i].promote(); 5841 if (handle != 0) { 5842 handle->dump(buffer, SIZE); 5843 result.append(buffer); 5844 } 5845 } 5846 5847 result.append("\n"); 5848 5849 write(fd, result.string(), result.length()); 5850 5851 if (locked) { 5852 mLock.unlock(); 5853 } 5854 5855 return NO_ERROR; 5856} 5857 5858// ---------------------------------------------------------------------------- 5859// EffectHandle implementation 5860// ---------------------------------------------------------------------------- 5861 5862#undef LOG_TAG 5863#define LOG_TAG "AudioFlinger::EffectHandle" 5864 5865AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 5866 const sp<AudioFlinger::Client>& client, 5867 const sp<IEffectClient>& effectClient, 5868 int32_t priority) 5869 : BnEffect(), 5870 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 5871{ 5872 LOGV("constructor %p", this); 5873 5874 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 5875 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 5876 if (mCblkMemory != 0) { 5877 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 5878 5879 if (mCblk) { 5880 new(mCblk) effect_param_cblk_t(); 5881 mBuffer = (uint8_t *)mCblk + bufOffset; 5882 } 5883 } else { 5884 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 5885 return; 5886 } 5887} 5888 5889AudioFlinger::EffectHandle::~EffectHandle() 5890{ 5891 LOGV("Destructor %p", this); 5892 disconnect(); 5893} 5894 5895status_t AudioFlinger::EffectHandle::enable() 5896{ 5897 if (!mHasControl) return INVALID_OPERATION; 5898 if (mEffect == 0) return DEAD_OBJECT; 5899 5900 return mEffect->setEnabled(true); 5901} 5902 5903status_t AudioFlinger::EffectHandle::disable() 5904{ 5905 if (!mHasControl) return INVALID_OPERATION; 5906 if (mEffect == NULL) return DEAD_OBJECT; 5907 5908 return mEffect->setEnabled(false); 5909} 5910 5911void AudioFlinger::EffectHandle::disconnect() 5912{ 5913 if (mEffect == 0) { 5914 return; 5915 } 5916 mEffect->disconnect(this); 5917 // release sp on module => module destructor can be called now 5918 mEffect.clear(); 5919 if (mCblk) { 5920 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 5921 } 5922 mCblkMemory.clear(); // and free the shared memory 5923 if (mClient != 0) { 5924 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 5925 mClient.clear(); 5926 } 5927} 5928 5929status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 5930 uint32_t cmdSize, 5931 void *pCmdData, 5932 uint32_t *replySize, 5933 void *pReplyData) 5934{ 5935// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 5936// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 5937 5938 // only get parameter command is permitted for applications not controlling the effect 5939 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 5940 return INVALID_OPERATION; 5941 } 5942 if (mEffect == 0) return DEAD_OBJECT; 5943 5944 // handle commands that are not forwarded transparently to effect engine 5945 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 5946 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 5947 // no risk to block the whole media server process or mixer threads is we are stuck here 5948 Mutex::Autolock _l(mCblk->lock); 5949 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 5950 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 5951 mCblk->serverIndex = 0; 5952 mCblk->clientIndex = 0; 5953 return BAD_VALUE; 5954 } 5955 status_t status = NO_ERROR; 5956 while (mCblk->serverIndex < mCblk->clientIndex) { 5957 int reply; 5958 uint32_t rsize = sizeof(int); 5959 int *p = (int *)(mBuffer + mCblk->serverIndex); 5960 int size = *p++; 5961 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 5962 LOGW("command(): invalid parameter block size"); 5963 break; 5964 } 5965 effect_param_t *param = (effect_param_t *)p; 5966 if (param->psize == 0 || param->vsize == 0) { 5967 LOGW("command(): null parameter or value size"); 5968 mCblk->serverIndex += size; 5969 continue; 5970 } 5971 uint32_t psize = sizeof(effect_param_t) + 5972 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 5973 param->vsize; 5974 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 5975 psize, 5976 p, 5977 &rsize, 5978 &reply); 5979 // stop at first error encountered 5980 if (ret != NO_ERROR) { 5981 status = ret; 5982 *(int *)pReplyData = reply; 5983 break; 5984 } else if (reply != NO_ERROR) { 5985 *(int *)pReplyData = reply; 5986 break; 5987 } 5988 mCblk->serverIndex += size; 5989 } 5990 mCblk->serverIndex = 0; 5991 mCblk->clientIndex = 0; 5992 return status; 5993 } else if (cmdCode == EFFECT_CMD_ENABLE) { 5994 *(int *)pReplyData = NO_ERROR; 5995 return enable(); 5996 } else if (cmdCode == EFFECT_CMD_DISABLE) { 5997 *(int *)pReplyData = NO_ERROR; 5998 return disable(); 5999 } 6000 6001 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6002} 6003 6004sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 6005 return mCblkMemory; 6006} 6007 6008void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 6009{ 6010 LOGV("setControl %p control %d", this, hasControl); 6011 6012 mHasControl = hasControl; 6013 if (signal && mEffectClient != 0) { 6014 mEffectClient->controlStatusChanged(hasControl); 6015 } 6016} 6017 6018void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6019 uint32_t cmdSize, 6020 void *pCmdData, 6021 uint32_t replySize, 6022 void *pReplyData) 6023{ 6024 if (mEffectClient != 0) { 6025 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6026 } 6027} 6028 6029 6030 6031void AudioFlinger::EffectHandle::setEnabled(bool enabled) 6032{ 6033 if (mEffectClient != 0) { 6034 mEffectClient->enableStatusChanged(enabled); 6035 } 6036} 6037 6038status_t AudioFlinger::EffectHandle::onTransact( 6039 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6040{ 6041 return BnEffect::onTransact(code, data, reply, flags); 6042} 6043 6044 6045void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 6046{ 6047 bool locked = tryLock(mCblk->lock); 6048 6049 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 6050 (mClient == NULL) ? getpid() : mClient->pid(), 6051 mPriority, 6052 mHasControl, 6053 !locked, 6054 mCblk->clientIndex, 6055 mCblk->serverIndex 6056 ); 6057 6058 if (locked) { 6059 mCblk->lock.unlock(); 6060 } 6061} 6062 6063#undef LOG_TAG 6064#define LOG_TAG "AudioFlinger::EffectChain" 6065 6066AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 6067 int sessionId) 6068 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mOwnInBuffer(false), 6069 mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 6070 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 6071{ 6072 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 6073} 6074 6075AudioFlinger::EffectChain::~EffectChain() 6076{ 6077 if (mOwnInBuffer) { 6078 delete mInBuffer; 6079 } 6080 6081} 6082 6083// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6084sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6085{ 6086 sp<EffectModule> effect; 6087 size_t size = mEffects.size(); 6088 6089 for (size_t i = 0; i < size; i++) { 6090 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6091 effect = mEffects[i]; 6092 break; 6093 } 6094 } 6095 return effect; 6096} 6097 6098// getEffectFromId_l() must be called with PlaybackThread::mLock held 6099sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6100{ 6101 sp<EffectModule> effect; 6102 size_t size = mEffects.size(); 6103 6104 for (size_t i = 0; i < size; i++) { 6105 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6106 if (id == 0 || mEffects[i]->id() == id) { 6107 effect = mEffects[i]; 6108 break; 6109 } 6110 } 6111 return effect; 6112} 6113 6114// Must be called with EffectChain::mLock locked 6115void AudioFlinger::EffectChain::process_l() 6116{ 6117 sp<ThreadBase> thread = mThread.promote(); 6118 if (thread == 0) { 6119 LOGW("process_l(): cannot promote mixer thread"); 6120 return; 6121 } 6122 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 6123 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 6124 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 6125 bool tracksOnSession = false; 6126 if (!isGlobalSession) { 6127 tracksOnSession = 6128 playbackThread->hasAudioSession(mSessionId) & PlaybackThread::TRACK_SESSION; 6129 } 6130 6131 size_t size = mEffects.size(); 6132 // do not process effect if no track is present in same audio session 6133 if (isGlobalSession || tracksOnSession) { 6134 for (size_t i = 0; i < size; i++) { 6135 mEffects[i]->process(); 6136 } 6137 } 6138 for (size_t i = 0; i < size; i++) { 6139 mEffects[i]->updateState(); 6140 } 6141 // if no track is active, input buffer must be cleared here as the mixer process 6142 // will not do it 6143 if (tracksOnSession && 6144 activeTracks() == 0) { 6145 size_t numSamples = playbackThread->frameCount() * playbackThread->channelCount(); 6146 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6147 } 6148} 6149 6150// addEffect_l() must be called with PlaybackThread::mLock held 6151status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6152{ 6153 effect_descriptor_t desc = effect->desc(); 6154 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6155 6156 Mutex::Autolock _l(mLock); 6157 effect->setChain(this); 6158 sp<ThreadBase> thread = mThread.promote(); 6159 if (thread == 0) { 6160 return NO_INIT; 6161 } 6162 effect->setThread(thread); 6163 6164 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6165 // Auxiliary effects are inserted at the beginning of mEffects vector as 6166 // they are processed first and accumulated in chain input buffer 6167 mEffects.insertAt(effect, 0); 6168 6169 // the input buffer for auxiliary effect contains mono samples in 6170 // 32 bit format. This is to avoid saturation in AudoMixer 6171 // accumulation stage. Saturation is done in EffectModule::process() before 6172 // calling the process in effect engine 6173 size_t numSamples = thread->frameCount(); 6174 int32_t *buffer = new int32_t[numSamples]; 6175 memset(buffer, 0, numSamples * sizeof(int32_t)); 6176 effect->setInBuffer((int16_t *)buffer); 6177 // auxiliary effects output samples to chain input buffer for further processing 6178 // by insert effects 6179 effect->setOutBuffer(mInBuffer); 6180 } else { 6181 // Insert effects are inserted at the end of mEffects vector as they are processed 6182 // after track and auxiliary effects. 6183 // Insert effect order as a function of indicated preference: 6184 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6185 // another effect is present 6186 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6187 // last effect claiming first position 6188 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6189 // first effect claiming last position 6190 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6191 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6192 // already present 6193 6194 int size = (int)mEffects.size(); 6195 int idx_insert = size; 6196 int idx_insert_first = -1; 6197 int idx_insert_last = -1; 6198 6199 for (int i = 0; i < size; i++) { 6200 effect_descriptor_t d = mEffects[i]->desc(); 6201 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6202 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6203 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6204 // check invalid effect chaining combinations 6205 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6206 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6207 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6208 return INVALID_OPERATION; 6209 } 6210 // remember position of first insert effect and by default 6211 // select this as insert position for new effect 6212 if (idx_insert == size) { 6213 idx_insert = i; 6214 } 6215 // remember position of last insert effect claiming 6216 // first position 6217 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6218 idx_insert_first = i; 6219 } 6220 // remember position of first insert effect claiming 6221 // last position 6222 if (iPref == EFFECT_FLAG_INSERT_LAST && 6223 idx_insert_last == -1) { 6224 idx_insert_last = i; 6225 } 6226 } 6227 } 6228 6229 // modify idx_insert from first position if needed 6230 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6231 if (idx_insert_last != -1) { 6232 idx_insert = idx_insert_last; 6233 } else { 6234 idx_insert = size; 6235 } 6236 } else { 6237 if (idx_insert_first != -1) { 6238 idx_insert = idx_insert_first + 1; 6239 } 6240 } 6241 6242 // always read samples from chain input buffer 6243 effect->setInBuffer(mInBuffer); 6244 6245 // if last effect in the chain, output samples to chain 6246 // output buffer, otherwise to chain input buffer 6247 if (idx_insert == size) { 6248 if (idx_insert != 0) { 6249 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6250 mEffects[idx_insert-1]->configure(); 6251 } 6252 effect->setOutBuffer(mOutBuffer); 6253 } else { 6254 effect->setOutBuffer(mInBuffer); 6255 } 6256 mEffects.insertAt(effect, idx_insert); 6257 6258 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6259 } 6260 effect->configure(); 6261 return NO_ERROR; 6262} 6263 6264// removeEffect_l() must be called with PlaybackThread::mLock held 6265size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6266{ 6267 Mutex::Autolock _l(mLock); 6268 int size = (int)mEffects.size(); 6269 int i; 6270 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6271 6272 for (i = 0; i < size; i++) { 6273 if (effect == mEffects[i]) { 6274 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6275 delete[] effect->inBuffer(); 6276 } else { 6277 if (i == size - 1 && i != 0) { 6278 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6279 mEffects[i - 1]->configure(); 6280 } 6281 } 6282 mEffects.removeAt(i); 6283 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6284 break; 6285 } 6286 } 6287 6288 return mEffects.size(); 6289} 6290 6291// setDevice_l() must be called with PlaybackThread::mLock held 6292void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6293{ 6294 size_t size = mEffects.size(); 6295 for (size_t i = 0; i < size; i++) { 6296 mEffects[i]->setDevice(device); 6297 } 6298} 6299 6300// setMode_l() must be called with PlaybackThread::mLock held 6301void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6302{ 6303 size_t size = mEffects.size(); 6304 for (size_t i = 0; i < size; i++) { 6305 mEffects[i]->setMode(mode); 6306 } 6307} 6308 6309// setVolume_l() must be called with PlaybackThread::mLock held 6310bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6311{ 6312 uint32_t newLeft = *left; 6313 uint32_t newRight = *right; 6314 bool hasControl = false; 6315 int ctrlIdx = -1; 6316 size_t size = mEffects.size(); 6317 6318 // first update volume controller 6319 for (size_t i = size; i > 0; i--) { 6320 if (mEffects[i - 1]->isProcessEnabled() && 6321 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6322 ctrlIdx = i - 1; 6323 hasControl = true; 6324 break; 6325 } 6326 } 6327 6328 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6329 if (hasControl) { 6330 *left = mNewLeftVolume; 6331 *right = mNewRightVolume; 6332 } 6333 return hasControl; 6334 } 6335 6336 mVolumeCtrlIdx = ctrlIdx; 6337 mLeftVolume = newLeft; 6338 mRightVolume = newRight; 6339 6340 // second get volume update from volume controller 6341 if (ctrlIdx >= 0) { 6342 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6343 mNewLeftVolume = newLeft; 6344 mNewRightVolume = newRight; 6345 } 6346 // then indicate volume to all other effects in chain. 6347 // Pass altered volume to effects before volume controller 6348 // and requested volume to effects after controller 6349 uint32_t lVol = newLeft; 6350 uint32_t rVol = newRight; 6351 6352 for (size_t i = 0; i < size; i++) { 6353 if ((int)i == ctrlIdx) continue; 6354 // this also works for ctrlIdx == -1 when there is no volume controller 6355 if ((int)i > ctrlIdx) { 6356 lVol = *left; 6357 rVol = *right; 6358 } 6359 mEffects[i]->setVolume(&lVol, &rVol, false); 6360 } 6361 *left = newLeft; 6362 *right = newRight; 6363 6364 return hasControl; 6365} 6366 6367status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6368{ 6369 const size_t SIZE = 256; 6370 char buffer[SIZE]; 6371 String8 result; 6372 6373 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6374 result.append(buffer); 6375 6376 bool locked = tryLock(mLock); 6377 // failed to lock - AudioFlinger is probably deadlocked 6378 if (!locked) { 6379 result.append("\tCould not lock mutex:\n"); 6380 } 6381 6382 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6383 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6384 mEffects.size(), 6385 (uint32_t)mInBuffer, 6386 (uint32_t)mOutBuffer, 6387 mActiveTrackCnt); 6388 result.append(buffer); 6389 write(fd, result.string(), result.size()); 6390 6391 for (size_t i = 0; i < mEffects.size(); ++i) { 6392 sp<EffectModule> effect = mEffects[i]; 6393 if (effect != 0) { 6394 effect->dump(fd, args); 6395 } 6396 } 6397 6398 if (locked) { 6399 mLock.unlock(); 6400 } 6401 6402 return NO_ERROR; 6403} 6404 6405#undef LOG_TAG 6406#define LOG_TAG "AudioFlinger" 6407 6408// ---------------------------------------------------------------------------- 6409 6410status_t AudioFlinger::onTransact( 6411 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6412{ 6413 return BnAudioFlinger::onTransact(code, data, reply, flags); 6414} 6415 6416}; // namespace android 6417