AudioFlinger.cpp revision 1d2bff0e588afe183a1baaae731519b4e957bbdb
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 <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <audio_effects/effect_visualizer.h> 54 55#include <cpustats/ThreadCpuUsage.h> 56// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 57 58// ---------------------------------------------------------------------------- 59 60 61namespace android { 62 63static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; 64static const char* kHardwareLockedString = "Hardware lock is taken\n"; 65 66//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 67static const float MAX_GAIN = 4096.0f; 68static const float MAX_GAIN_INT = 0x1000; 69 70// retry counts for buffer fill timeout 71// 50 * ~20msecs = 1 second 72static const int8_t kMaxTrackRetries = 50; 73static const int8_t kMaxTrackStartupRetries = 50; 74// allow less retry attempts on direct output thread. 75// direct outputs can be a scarce resource in audio hardware and should 76// be released as quickly as possible. 77static const int8_t kMaxTrackRetriesDirect = 2; 78 79static const int kDumpLockRetries = 50; 80static const int kDumpLockSleep = 20000; 81 82static const nsecs_t kWarningThrottle = seconds(5); 83 84// RecordThread loop sleep time upon application overrun or audio HAL read error 85static const int kRecordThreadSleepUs = 5000; 86 87// ---------------------------------------------------------------------------- 88 89static bool recordingAllowed() { 90 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 91 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 92 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 93 return ok; 94} 95 96static bool settingsAllowed() { 97 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 98 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 99 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 100 return ok; 101} 102 103// To collect the amplifier usage 104static void addBatteryData(uint32_t params) { 105 sp<IBinder> binder = 106 defaultServiceManager()->getService(String16("media.player")); 107 sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder); 108 if (service.get() == NULL) { 109 LOGW("Cannot connect to the MediaPlayerService for battery tracking"); 110 return; 111 } 112 113 service->addBatteryData(params); 114} 115 116static int load_audio_interface(const char *if_name, const hw_module_t **mod, 117 audio_hw_device_t **dev) 118{ 119 int rc; 120 121 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 122 if (rc) 123 goto out; 124 125 rc = audio_hw_device_open(*mod, dev); 126 LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 127 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 128 if (rc) 129 goto out; 130 131 return 0; 132 133out: 134 *mod = NULL; 135 *dev = NULL; 136 return rc; 137} 138 139static const char *audio_interfaces[] = { 140 "primary", 141 "a2dp", 142 "usb", 143}; 144#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 145 146// ---------------------------------------------------------------------------- 147 148AudioFlinger::AudioFlinger() 149 : BnAudioFlinger(), 150 mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1) 151{ 152} 153 154void AudioFlinger::onFirstRef() 155{ 156 int rc = 0; 157 158 Mutex::Autolock _l(mLock); 159 160 /* TODO: move all this work into an Init() function */ 161 mHardwareStatus = AUDIO_HW_IDLE; 162 163 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 164 const hw_module_t *mod; 165 audio_hw_device_t *dev; 166 167 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 168 if (rc) 169 continue; 170 171 LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 172 mod->name, mod->id); 173 mAudioHwDevs.push(dev); 174 175 if (!mPrimaryHardwareDev) { 176 mPrimaryHardwareDev = dev; 177 LOGI("Using '%s' (%s.%s) as the primary audio interface", 178 mod->name, mod->id, audio_interfaces[i]); 179 } 180 } 181 182 mHardwareStatus = AUDIO_HW_INIT; 183 184 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 185 LOGE("Primary audio interface not found"); 186 return; 187 } 188 189 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 190 audio_hw_device_t *dev = mAudioHwDevs[i]; 191 192 mHardwareStatus = AUDIO_HW_INIT; 193 rc = dev->init_check(dev); 194 if (rc == 0) { 195 AutoMutex lock(mHardwareLock); 196 197 mMode = AUDIO_MODE_NORMAL; 198 mHardwareStatus = AUDIO_HW_SET_MODE; 199 dev->set_mode(dev, mMode); 200 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 201 dev->set_master_volume(dev, 1.0f); 202 mHardwareStatus = AUDIO_HW_IDLE; 203 } 204 } 205} 206 207status_t AudioFlinger::initCheck() const 208{ 209 Mutex::Autolock _l(mLock); 210 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 211 return NO_INIT; 212 return NO_ERROR; 213} 214 215AudioFlinger::~AudioFlinger() 216{ 217 int num_devs = mAudioHwDevs.size(); 218 219 while (!mRecordThreads.isEmpty()) { 220 // closeInput() will remove first entry from mRecordThreads 221 closeInput(mRecordThreads.keyAt(0)); 222 } 223 while (!mPlaybackThreads.isEmpty()) { 224 // closeOutput() will remove first entry from mPlaybackThreads 225 closeOutput(mPlaybackThreads.keyAt(0)); 226 } 227 228 for (int i = 0; i < num_devs; i++) { 229 audio_hw_device_t *dev = mAudioHwDevs[i]; 230 audio_hw_device_close(dev); 231 } 232 mAudioHwDevs.clear(); 233} 234 235audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 236{ 237 /* first matching HW device is returned */ 238 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 239 audio_hw_device_t *dev = mAudioHwDevs[i]; 240 if ((dev->get_supported_devices(dev) & devices) == devices) 241 return dev; 242 } 243 return NULL; 244} 245 246status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 247{ 248 const size_t SIZE = 256; 249 char buffer[SIZE]; 250 String8 result; 251 252 result.append("Clients:\n"); 253 for (size_t i = 0; i < mClients.size(); ++i) { 254 wp<Client> wClient = mClients.valueAt(i); 255 if (wClient != 0) { 256 sp<Client> client = wClient.promote(); 257 if (client != 0) { 258 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 259 result.append(buffer); 260 } 261 } 262 } 263 write(fd, result.string(), result.size()); 264 return NO_ERROR; 265} 266 267 268status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 269{ 270 const size_t SIZE = 256; 271 char buffer[SIZE]; 272 String8 result; 273 int hardwareStatus = mHardwareStatus; 274 275 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 276 result.append(buffer); 277 write(fd, result.string(), result.size()); 278 return NO_ERROR; 279} 280 281status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 282{ 283 const size_t SIZE = 256; 284 char buffer[SIZE]; 285 String8 result; 286 snprintf(buffer, SIZE, "Permission Denial: " 287 "can't dump AudioFlinger from pid=%d, uid=%d\n", 288 IPCThreadState::self()->getCallingPid(), 289 IPCThreadState::self()->getCallingUid()); 290 result.append(buffer); 291 write(fd, result.string(), result.size()); 292 return NO_ERROR; 293} 294 295static bool tryLock(Mutex& mutex) 296{ 297 bool locked = false; 298 for (int i = 0; i < kDumpLockRetries; ++i) { 299 if (mutex.tryLock() == NO_ERROR) { 300 locked = true; 301 break; 302 } 303 usleep(kDumpLockSleep); 304 } 305 return locked; 306} 307 308status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 309{ 310 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 311 dumpPermissionDenial(fd, args); 312 } else { 313 // get state of hardware lock 314 bool hardwareLocked = tryLock(mHardwareLock); 315 if (!hardwareLocked) { 316 String8 result(kHardwareLockedString); 317 write(fd, result.string(), result.size()); 318 } else { 319 mHardwareLock.unlock(); 320 } 321 322 bool locked = tryLock(mLock); 323 324 // failed to lock - AudioFlinger is probably deadlocked 325 if (!locked) { 326 String8 result(kDeadlockedString); 327 write(fd, result.string(), result.size()); 328 } 329 330 dumpClients(fd, args); 331 dumpInternals(fd, args); 332 333 // dump playback threads 334 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 335 mPlaybackThreads.valueAt(i)->dump(fd, args); 336 } 337 338 // dump record threads 339 for (size_t i = 0; i < mRecordThreads.size(); i++) { 340 mRecordThreads.valueAt(i)->dump(fd, args); 341 } 342 343 // dump all hardware devs 344 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 345 audio_hw_device_t *dev = mAudioHwDevs[i]; 346 dev->dump(dev, fd); 347 } 348 if (locked) mLock.unlock(); 349 } 350 return NO_ERROR; 351} 352 353 354// IAudioFlinger interface 355 356 357sp<IAudioTrack> AudioFlinger::createTrack( 358 pid_t pid, 359 int streamType, 360 uint32_t sampleRate, 361 uint32_t format, 362 uint32_t channelMask, 363 int frameCount, 364 uint32_t flags, 365 const sp<IMemory>& sharedBuffer, 366 int output, 367 int *sessionId, 368 status_t *status) 369{ 370 sp<PlaybackThread::Track> track; 371 sp<TrackHandle> trackHandle; 372 sp<Client> client; 373 wp<Client> wclient; 374 status_t lStatus; 375 int lSessionId; 376 377 if (streamType >= AUDIO_STREAM_CNT) { 378 LOGE("invalid stream type"); 379 lStatus = BAD_VALUE; 380 goto Exit; 381 } 382 383 { 384 Mutex::Autolock _l(mLock); 385 PlaybackThread *thread = checkPlaybackThread_l(output); 386 PlaybackThread *effectThread = NULL; 387 if (thread == NULL) { 388 LOGE("unknown output thread"); 389 lStatus = BAD_VALUE; 390 goto Exit; 391 } 392 393 wclient = mClients.valueFor(pid); 394 395 if (wclient != NULL) { 396 client = wclient.promote(); 397 } else { 398 client = new Client(this, pid); 399 mClients.add(pid, client); 400 } 401 402 LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 403 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 404 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 405 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 406 if (mPlaybackThreads.keyAt(i) != output) { 407 // prevent same audio session on different output threads 408 uint32_t sessions = t->hasAudioSession(*sessionId); 409 if (sessions & PlaybackThread::TRACK_SESSION) { 410 lStatus = BAD_VALUE; 411 goto Exit; 412 } 413 // check if an effect with same session ID is waiting for a track to be created 414 if (sessions & PlaybackThread::EFFECT_SESSION) { 415 effectThread = t.get(); 416 } 417 } 418 } 419 lSessionId = *sessionId; 420 } else { 421 // if no audio session id is provided, create one here 422 lSessionId = nextUniqueId(); 423 if (sessionId != NULL) { 424 *sessionId = lSessionId; 425 } 426 } 427 LOGV("createTrack() lSessionId: %d", lSessionId); 428 429 track = thread->createTrack_l(client, streamType, sampleRate, format, 430 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 431 432 // move effect chain to this output thread if an effect on same session was waiting 433 // for a track to be created 434 if (lStatus == NO_ERROR && effectThread != NULL) { 435 Mutex::Autolock _dl(thread->mLock); 436 Mutex::Autolock _sl(effectThread->mLock); 437 moveEffectChain_l(lSessionId, effectThread, thread, true); 438 } 439 } 440 if (lStatus == NO_ERROR) { 441 trackHandle = new TrackHandle(track); 442 } else { 443 // remove local strong reference to Client before deleting the Track so that the Client 444 // destructor is called by the TrackBase destructor with mLock held 445 client.clear(); 446 track.clear(); 447 } 448 449Exit: 450 if(status) { 451 *status = lStatus; 452 } 453 return trackHandle; 454} 455 456uint32_t AudioFlinger::sampleRate(int output) const 457{ 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 if (thread == NULL) { 461 LOGW("sampleRate() unknown thread %d", output); 462 return 0; 463 } 464 return thread->sampleRate(); 465} 466 467int AudioFlinger::channelCount(int output) const 468{ 469 Mutex::Autolock _l(mLock); 470 PlaybackThread *thread = checkPlaybackThread_l(output); 471 if (thread == NULL) { 472 LOGW("channelCount() unknown thread %d", output); 473 return 0; 474 } 475 return thread->channelCount(); 476} 477 478uint32_t AudioFlinger::format(int output) const 479{ 480 Mutex::Autolock _l(mLock); 481 PlaybackThread *thread = checkPlaybackThread_l(output); 482 if (thread == NULL) { 483 LOGW("format() unknown thread %d", output); 484 return 0; 485 } 486 return thread->format(); 487} 488 489size_t AudioFlinger::frameCount(int output) const 490{ 491 Mutex::Autolock _l(mLock); 492 PlaybackThread *thread = checkPlaybackThread_l(output); 493 if (thread == NULL) { 494 LOGW("frameCount() unknown thread %d", output); 495 return 0; 496 } 497 return thread->frameCount(); 498} 499 500uint32_t AudioFlinger::latency(int output) const 501{ 502 Mutex::Autolock _l(mLock); 503 PlaybackThread *thread = checkPlaybackThread_l(output); 504 if (thread == NULL) { 505 LOGW("latency() unknown thread %d", output); 506 return 0; 507 } 508 return thread->latency(); 509} 510 511status_t AudioFlinger::setMasterVolume(float value) 512{ 513 // check calling permissions 514 if (!settingsAllowed()) { 515 return PERMISSION_DENIED; 516 } 517 518 // when hw supports master volume, don't scale in sw mixer 519 { // scope for the lock 520 AutoMutex lock(mHardwareLock); 521 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 522 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 523 value = 1.0f; 524 } 525 mHardwareStatus = AUDIO_HW_IDLE; 526 } 527 528 Mutex::Autolock _l(mLock); 529 mMasterVolume = value; 530 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 531 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 532 533 return NO_ERROR; 534} 535 536status_t AudioFlinger::setMode(int mode) 537{ 538 status_t ret; 539 540 // check calling permissions 541 if (!settingsAllowed()) { 542 return PERMISSION_DENIED; 543 } 544 if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) { 545 LOGW("Illegal value: setMode(%d)", mode); 546 return BAD_VALUE; 547 } 548 549 { // scope for the lock 550 AutoMutex lock(mHardwareLock); 551 mHardwareStatus = AUDIO_HW_SET_MODE; 552 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 553 mHardwareStatus = AUDIO_HW_IDLE; 554 } 555 556 if (NO_ERROR == ret) { 557 Mutex::Autolock _l(mLock); 558 mMode = mode; 559 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 560 mPlaybackThreads.valueAt(i)->setMode(mode); 561 } 562 563 return ret; 564} 565 566status_t AudioFlinger::setMicMute(bool state) 567{ 568 // check calling permissions 569 if (!settingsAllowed()) { 570 return PERMISSION_DENIED; 571 } 572 573 AutoMutex lock(mHardwareLock); 574 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 575 status_t ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 576 mHardwareStatus = AUDIO_HW_IDLE; 577 return ret; 578} 579 580bool AudioFlinger::getMicMute() const 581{ 582 bool state = AUDIO_MODE_INVALID; 583 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 584 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 585 mHardwareStatus = AUDIO_HW_IDLE; 586 return state; 587} 588 589status_t AudioFlinger::setMasterMute(bool muted) 590{ 591 // check calling permissions 592 if (!settingsAllowed()) { 593 return PERMISSION_DENIED; 594 } 595 596 Mutex::Autolock _l(mLock); 597 mMasterMute = muted; 598 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 599 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 600 601 return NO_ERROR; 602} 603 604float AudioFlinger::masterVolume() const 605{ 606 return mMasterVolume; 607} 608 609bool AudioFlinger::masterMute() const 610{ 611 return mMasterMute; 612} 613 614status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 615{ 616 // check calling permissions 617 if (!settingsAllowed()) { 618 return PERMISSION_DENIED; 619 } 620 621 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 622 return BAD_VALUE; 623 } 624 625 AutoMutex lock(mLock); 626 PlaybackThread *thread = NULL; 627 if (output) { 628 thread = checkPlaybackThread_l(output); 629 if (thread == NULL) { 630 return BAD_VALUE; 631 } 632 } 633 634 mStreamTypes[stream].volume = value; 635 636 if (thread == NULL) { 637 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 638 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 639 } 640 } else { 641 thread->setStreamVolume(stream, value); 642 } 643 644 return NO_ERROR; 645} 646 647status_t AudioFlinger::setStreamMute(int stream, bool muted) 648{ 649 // check calling permissions 650 if (!settingsAllowed()) { 651 return PERMISSION_DENIED; 652 } 653 654 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || 655 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 656 return BAD_VALUE; 657 } 658 659 AutoMutex lock(mLock); 660 mStreamTypes[stream].mute = muted; 661 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 662 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 663 664 return NO_ERROR; 665} 666 667float AudioFlinger::streamVolume(int stream, int output) const 668{ 669 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 670 return 0.0f; 671 } 672 673 AutoMutex lock(mLock); 674 float volume; 675 if (output) { 676 PlaybackThread *thread = checkPlaybackThread_l(output); 677 if (thread == NULL) { 678 return 0.0f; 679 } 680 volume = thread->streamVolume(stream); 681 } else { 682 volume = mStreamTypes[stream].volume; 683 } 684 685 return volume; 686} 687 688bool AudioFlinger::streamMute(int stream) const 689{ 690 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 691 return true; 692 } 693 694 return mStreamTypes[stream].mute; 695} 696 697status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 698{ 699 status_t result; 700 701 LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 702 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 703 // check calling permissions 704 if (!settingsAllowed()) { 705 return PERMISSION_DENIED; 706 } 707 708 // ioHandle == 0 means the parameters are global to the audio hardware interface 709 if (ioHandle == 0) { 710 AutoMutex lock(mHardwareLock); 711 mHardwareStatus = AUDIO_SET_PARAMETER; 712 status_t final_result = NO_ERROR; 713 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 714 audio_hw_device_t *dev = mAudioHwDevs[i]; 715 result = dev->set_parameters(dev, keyValuePairs.string()); 716 final_result = result ?: final_result; 717 } 718 mHardwareStatus = AUDIO_HW_IDLE; 719 return final_result; 720 } 721 722 // hold a strong ref on thread in case closeOutput() or closeInput() is called 723 // and the thread is exited once the lock is released 724 sp<ThreadBase> thread; 725 { 726 Mutex::Autolock _l(mLock); 727 thread = checkPlaybackThread_l(ioHandle); 728 if (thread == NULL) { 729 thread = checkRecordThread_l(ioHandle); 730 } else if (thread.get() == primaryPlaybackThread_l()) { 731 // indicate output device change to all input threads for pre processing 732 AudioParameter param = AudioParameter(keyValuePairs); 733 int value; 734 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 735 for (size_t i = 0; i < mRecordThreads.size(); i++) { 736 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 737 } 738 } 739 } 740 } 741 if (thread != NULL) { 742 result = thread->setParameters(keyValuePairs); 743 return result; 744 } 745 return BAD_VALUE; 746} 747 748String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 749{ 750// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 751// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 752 753 if (ioHandle == 0) { 754 String8 out_s8; 755 756 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 757 audio_hw_device_t *dev = mAudioHwDevs[i]; 758 char *s = dev->get_parameters(dev, keys.string()); 759 out_s8 += String8(s); 760 free(s); 761 } 762 return out_s8; 763 } 764 765 Mutex::Autolock _l(mLock); 766 767 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 768 if (playbackThread != NULL) { 769 return playbackThread->getParameters(keys); 770 } 771 RecordThread *recordThread = checkRecordThread_l(ioHandle); 772 if (recordThread != NULL) { 773 return recordThread->getParameters(keys); 774 } 775 return String8(""); 776} 777 778size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 779{ 780 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 781} 782 783unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 784{ 785 if (ioHandle == 0) { 786 return 0; 787 } 788 789 Mutex::Autolock _l(mLock); 790 791 RecordThread *recordThread = checkRecordThread_l(ioHandle); 792 if (recordThread != NULL) { 793 return recordThread->getInputFramesLost(); 794 } 795 return 0; 796} 797 798status_t AudioFlinger::setVoiceVolume(float value) 799{ 800 // check calling permissions 801 if (!settingsAllowed()) { 802 return PERMISSION_DENIED; 803 } 804 805 AutoMutex lock(mHardwareLock); 806 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 807 status_t ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 808 mHardwareStatus = AUDIO_HW_IDLE; 809 810 return ret; 811} 812 813status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 814{ 815 status_t status; 816 817 Mutex::Autolock _l(mLock); 818 819 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 820 if (playbackThread != NULL) { 821 return playbackThread->getRenderPosition(halFrames, dspFrames); 822 } 823 824 return BAD_VALUE; 825} 826 827void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 828{ 829 830 Mutex::Autolock _l(mLock); 831 832 int pid = IPCThreadState::self()->getCallingPid(); 833 if (mNotificationClients.indexOfKey(pid) < 0) { 834 sp<NotificationClient> notificationClient = new NotificationClient(this, 835 client, 836 pid); 837 LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 838 839 mNotificationClients.add(pid, notificationClient); 840 841 sp<IBinder> binder = client->asBinder(); 842 binder->linkToDeath(notificationClient); 843 844 // the config change is always sent from playback or record threads to avoid deadlock 845 // with AudioSystem::gLock 846 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 847 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 848 } 849 850 for (size_t i = 0; i < mRecordThreads.size(); i++) { 851 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 852 } 853 } 854} 855 856void AudioFlinger::removeNotificationClient(pid_t pid) 857{ 858 Mutex::Autolock _l(mLock); 859 860 int index = mNotificationClients.indexOfKey(pid); 861 if (index >= 0) { 862 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 863 LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 864 mNotificationClients.removeItem(pid); 865 } 866} 867 868// audioConfigChanged_l() must be called with AudioFlinger::mLock held 869void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 870{ 871 size_t size = mNotificationClients.size(); 872 for (size_t i = 0; i < size; i++) { 873 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 874 } 875} 876 877// removeClient_l() must be called with AudioFlinger::mLock held 878void AudioFlinger::removeClient_l(pid_t pid) 879{ 880 LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 881 mClients.removeItem(pid); 882} 883 884 885// ---------------------------------------------------------------------------- 886 887AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 888 : Thread(false), 889 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 890 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false), mDevice(device) 891{ 892} 893 894AudioFlinger::ThreadBase::~ThreadBase() 895{ 896 mParamCond.broadcast(); 897 mNewParameters.clear(); 898} 899 900void AudioFlinger::ThreadBase::exit() 901{ 902 // keep a strong ref on ourself so that we wont get 903 // destroyed in the middle of requestExitAndWait() 904 sp <ThreadBase> strongMe = this; 905 906 LOGV("ThreadBase::exit"); 907 { 908 AutoMutex lock(&mLock); 909 mExiting = true; 910 requestExit(); 911 mWaitWorkCV.signal(); 912 } 913 requestExitAndWait(); 914} 915 916uint32_t AudioFlinger::ThreadBase::sampleRate() const 917{ 918 return mSampleRate; 919} 920 921int AudioFlinger::ThreadBase::channelCount() const 922{ 923 return (int)mChannelCount; 924} 925 926uint32_t AudioFlinger::ThreadBase::format() const 927{ 928 return mFormat; 929} 930 931size_t AudioFlinger::ThreadBase::frameCount() const 932{ 933 return mFrameCount; 934} 935 936status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 937{ 938 status_t status; 939 940 LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 941 Mutex::Autolock _l(mLock); 942 943 mNewParameters.add(keyValuePairs); 944 mWaitWorkCV.signal(); 945 // wait condition with timeout in case the thread loop has exited 946 // before the request could be processed 947 if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { 948 status = mParamStatus; 949 mWaitWorkCV.signal(); 950 } else { 951 status = TIMED_OUT; 952 } 953 return status; 954} 955 956void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 957{ 958 Mutex::Autolock _l(mLock); 959 sendConfigEvent_l(event, param); 960} 961 962// sendConfigEvent_l() must be called with ThreadBase::mLock held 963void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 964{ 965 ConfigEvent *configEvent = new ConfigEvent(); 966 configEvent->mEvent = event; 967 configEvent->mParam = param; 968 mConfigEvents.add(configEvent); 969 LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 970 mWaitWorkCV.signal(); 971} 972 973void AudioFlinger::ThreadBase::processConfigEvents() 974{ 975 mLock.lock(); 976 while(!mConfigEvents.isEmpty()) { 977 LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 978 ConfigEvent *configEvent = mConfigEvents[0]; 979 mConfigEvents.removeAt(0); 980 // release mLock before locking AudioFlinger mLock: lock order is always 981 // AudioFlinger then ThreadBase to avoid cross deadlock 982 mLock.unlock(); 983 mAudioFlinger->mLock.lock(); 984 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 985 mAudioFlinger->mLock.unlock(); 986 delete configEvent; 987 mLock.lock(); 988 } 989 mLock.unlock(); 990} 991 992status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 993{ 994 const size_t SIZE = 256; 995 char buffer[SIZE]; 996 String8 result; 997 998 bool locked = tryLock(mLock); 999 if (!locked) { 1000 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1001 write(fd, buffer, strlen(buffer)); 1002 } 1003 1004 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1005 result.append(buffer); 1006 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1007 result.append(buffer); 1008 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1009 result.append(buffer); 1010 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1011 result.append(buffer); 1012 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1013 result.append(buffer); 1014 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1015 result.append(buffer); 1016 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 1017 result.append(buffer); 1018 1019 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1020 result.append(buffer); 1021 result.append(" Index Command"); 1022 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1023 snprintf(buffer, SIZE, "\n %02d ", i); 1024 result.append(buffer); 1025 result.append(mNewParameters[i]); 1026 } 1027 1028 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1029 result.append(buffer); 1030 snprintf(buffer, SIZE, " Index event param\n"); 1031 result.append(buffer); 1032 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1033 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 1034 result.append(buffer); 1035 } 1036 result.append("\n"); 1037 1038 write(fd, result.string(), result.size()); 1039 1040 if (locked) { 1041 mLock.unlock(); 1042 } 1043 return NO_ERROR; 1044} 1045 1046status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1047{ 1048 const size_t SIZE = 256; 1049 char buffer[SIZE]; 1050 String8 result; 1051 1052 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1053 write(fd, buffer, strlen(buffer)); 1054 1055 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1056 sp<EffectChain> chain = mEffectChains[i]; 1057 if (chain != 0) { 1058 chain->dump(fd, args); 1059 } 1060 } 1061 return NO_ERROR; 1062} 1063 1064 1065// ---------------------------------------------------------------------------- 1066 1067AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1068 AudioStreamOut* output, 1069 int id, 1070 uint32_t device) 1071 : ThreadBase(audioFlinger, id, device), 1072 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 1073 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1074{ 1075 readOutputParameters(); 1076 1077 mMasterVolume = mAudioFlinger->masterVolume(); 1078 mMasterMute = mAudioFlinger->masterMute(); 1079 1080 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 1081 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1082 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1083 } 1084} 1085 1086AudioFlinger::PlaybackThread::~PlaybackThread() 1087{ 1088 delete [] mMixBuffer; 1089} 1090 1091status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1092{ 1093 dumpInternals(fd, args); 1094 dumpTracks(fd, args); 1095 dumpEffectChains(fd, args); 1096 return NO_ERROR; 1097} 1098 1099status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1100{ 1101 const size_t SIZE = 256; 1102 char buffer[SIZE]; 1103 String8 result; 1104 1105 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1106 result.append(buffer); 1107 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1108 for (size_t i = 0; i < mTracks.size(); ++i) { 1109 sp<Track> track = mTracks[i]; 1110 if (track != 0) { 1111 track->dump(buffer, SIZE); 1112 result.append(buffer); 1113 } 1114 } 1115 1116 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1117 result.append(buffer); 1118 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1119 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1120 wp<Track> wTrack = mActiveTracks[i]; 1121 if (wTrack != 0) { 1122 sp<Track> track = wTrack.promote(); 1123 if (track != 0) { 1124 track->dump(buffer, SIZE); 1125 result.append(buffer); 1126 } 1127 } 1128 } 1129 write(fd, result.string(), result.size()); 1130 return NO_ERROR; 1131} 1132 1133status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1134{ 1135 const size_t SIZE = 256; 1136 char buffer[SIZE]; 1137 String8 result; 1138 1139 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1140 result.append(buffer); 1141 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1142 result.append(buffer); 1143 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1144 result.append(buffer); 1145 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1146 result.append(buffer); 1147 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1148 result.append(buffer); 1149 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1150 result.append(buffer); 1151 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1152 result.append(buffer); 1153 write(fd, result.string(), result.size()); 1154 1155 dumpBase(fd, args); 1156 1157 return NO_ERROR; 1158} 1159 1160// Thread virtuals 1161status_t AudioFlinger::PlaybackThread::readyToRun() 1162{ 1163 if (mSampleRate == 0) { 1164 LOGE("No working audio driver found."); 1165 return NO_INIT; 1166 } 1167 LOGI("AudioFlinger's thread %p ready to run", this); 1168 return NO_ERROR; 1169} 1170 1171void AudioFlinger::PlaybackThread::onFirstRef() 1172{ 1173 const size_t SIZE = 256; 1174 char buffer[SIZE]; 1175 1176 snprintf(buffer, SIZE, "Playback Thread %p", this); 1177 1178 run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); 1179} 1180 1181// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1182sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1183 const sp<AudioFlinger::Client>& client, 1184 int streamType, 1185 uint32_t sampleRate, 1186 uint32_t format, 1187 uint32_t channelMask, 1188 int frameCount, 1189 const sp<IMemory>& sharedBuffer, 1190 int sessionId, 1191 status_t *status) 1192{ 1193 sp<Track> track; 1194 status_t lStatus; 1195 1196 if (mType == DIRECT) { 1197 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1198 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1199 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1200 "for output %p with format %d", 1201 sampleRate, format, channelMask, mOutput, mFormat); 1202 lStatus = BAD_VALUE; 1203 goto Exit; 1204 } 1205 } 1206 } else { 1207 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1208 if (sampleRate > mSampleRate*2) { 1209 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1210 lStatus = BAD_VALUE; 1211 goto Exit; 1212 } 1213 } 1214 1215 lStatus = initCheck(); 1216 if (lStatus != NO_ERROR) { 1217 LOGE("Audio driver not initialized."); 1218 goto Exit; 1219 } 1220 1221 { // scope for mLock 1222 Mutex::Autolock _l(mLock); 1223 1224 // all tracks in same audio session must share the same routing strategy otherwise 1225 // conflicts will happen when tracks are moved from one output to another by audio policy 1226 // manager 1227 uint32_t strategy = 1228 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1229 for (size_t i = 0; i < mTracks.size(); ++i) { 1230 sp<Track> t = mTracks[i]; 1231 if (t != 0) { 1232 if (sessionId == t->sessionId() && 1233 strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) { 1234 lStatus = BAD_VALUE; 1235 goto Exit; 1236 } 1237 } 1238 } 1239 1240 track = new Track(this, client, streamType, sampleRate, format, 1241 channelMask, frameCount, sharedBuffer, sessionId); 1242 if (track->getCblk() == NULL || track->name() < 0) { 1243 lStatus = NO_MEMORY; 1244 goto Exit; 1245 } 1246 mTracks.add(track); 1247 1248 sp<EffectChain> chain = getEffectChain_l(sessionId); 1249 if (chain != 0) { 1250 LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1251 track->setMainBuffer(chain->inBuffer()); 1252 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1253 chain->incTrackCnt(); 1254 } 1255 } 1256 lStatus = NO_ERROR; 1257 1258Exit: 1259 if(status) { 1260 *status = lStatus; 1261 } 1262 return track; 1263} 1264 1265uint32_t AudioFlinger::PlaybackThread::latency() const 1266{ 1267 if (mOutput) { 1268 return mOutput->stream->get_latency(mOutput->stream); 1269 } 1270 else { 1271 return 0; 1272 } 1273} 1274 1275status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1276{ 1277 mMasterVolume = value; 1278 return NO_ERROR; 1279} 1280 1281status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1282{ 1283 mMasterMute = muted; 1284 return NO_ERROR; 1285} 1286 1287float AudioFlinger::PlaybackThread::masterVolume() const 1288{ 1289 return mMasterVolume; 1290} 1291 1292bool AudioFlinger::PlaybackThread::masterMute() const 1293{ 1294 return mMasterMute; 1295} 1296 1297status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1298{ 1299 mStreamTypes[stream].volume = value; 1300 return NO_ERROR; 1301} 1302 1303status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1304{ 1305 mStreamTypes[stream].mute = muted; 1306 return NO_ERROR; 1307} 1308 1309float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1310{ 1311 return mStreamTypes[stream].volume; 1312} 1313 1314bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1315{ 1316 return mStreamTypes[stream].mute; 1317} 1318 1319// addTrack_l() must be called with ThreadBase::mLock held 1320status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1321{ 1322 status_t status = ALREADY_EXISTS; 1323 1324 // set retry count for buffer fill 1325 track->mRetryCount = kMaxTrackStartupRetries; 1326 if (mActiveTracks.indexOf(track) < 0) { 1327 // the track is newly added, make sure it fills up all its 1328 // buffers before playing. This is to ensure the client will 1329 // effectively get the latency it requested. 1330 track->mFillingUpStatus = Track::FS_FILLING; 1331 track->mResetDone = false; 1332 mActiveTracks.add(track); 1333 if (track->mainBuffer() != mMixBuffer) { 1334 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1335 if (chain != 0) { 1336 LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1337 chain->incActiveTrackCnt(); 1338 } 1339 } 1340 1341 status = NO_ERROR; 1342 } 1343 1344 LOGV("mWaitWorkCV.broadcast"); 1345 mWaitWorkCV.broadcast(); 1346 1347 return status; 1348} 1349 1350// destroyTrack_l() must be called with ThreadBase::mLock held 1351void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1352{ 1353 track->mState = TrackBase::TERMINATED; 1354 if (mActiveTracks.indexOf(track) < 0) { 1355 removeTrack_l(track); 1356 } 1357} 1358 1359void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1360{ 1361 mTracks.remove(track); 1362 deleteTrackName_l(track->name()); 1363 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1364 if (chain != 0) { 1365 chain->decTrackCnt(); 1366 } 1367} 1368 1369String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1370{ 1371 String8 out_s8; 1372 char *s; 1373 1374 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1375 out_s8 = String8(s); 1376 free(s); 1377 return out_s8; 1378} 1379 1380// destroyTrack_l() must be called with AudioFlinger::mLock held 1381void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1382 AudioSystem::OutputDescriptor desc; 1383 void *param2 = 0; 1384 1385 LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1386 1387 switch (event) { 1388 case AudioSystem::OUTPUT_OPENED: 1389 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1390 desc.channels = mChannelMask; 1391 desc.samplingRate = mSampleRate; 1392 desc.format = mFormat; 1393 desc.frameCount = mFrameCount; 1394 desc.latency = latency(); 1395 param2 = &desc; 1396 break; 1397 1398 case AudioSystem::STREAM_CONFIG_CHANGED: 1399 param2 = ¶m; 1400 case AudioSystem::OUTPUT_CLOSED: 1401 default: 1402 break; 1403 } 1404 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1405} 1406 1407void AudioFlinger::PlaybackThread::readOutputParameters() 1408{ 1409 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1410 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1411 mChannelCount = (uint16_t)popcount(mChannelMask); 1412 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1413 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); 1414 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1415 1416 // FIXME - Current mixer implementation only supports stereo output: Always 1417 // Allocate a stereo buffer even if HW output is mono. 1418 if (mMixBuffer != NULL) delete[] mMixBuffer; 1419 mMixBuffer = new int16_t[mFrameCount * 2]; 1420 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1421 1422 // force reconfiguration of effect chains and engines to take new buffer size and audio 1423 // parameters into account 1424 // Note that mLock is not held when readOutputParameters() is called from the constructor 1425 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1426 // matter. 1427 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1428 Vector< sp<EffectChain> > effectChains = mEffectChains; 1429 for (size_t i = 0; i < effectChains.size(); i ++) { 1430 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1431 } 1432} 1433 1434status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1435{ 1436 if (halFrames == 0 || dspFrames == 0) { 1437 return BAD_VALUE; 1438 } 1439 if (initCheck() != NO_ERROR) { 1440 return INVALID_OPERATION; 1441 } 1442 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1443 1444 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1445} 1446 1447uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1448{ 1449 Mutex::Autolock _l(mLock); 1450 uint32_t result = 0; 1451 if (getEffectChain_l(sessionId) != 0) { 1452 result = EFFECT_SESSION; 1453 } 1454 1455 for (size_t i = 0; i < mTracks.size(); ++i) { 1456 sp<Track> track = mTracks[i]; 1457 if (sessionId == track->sessionId() && 1458 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1459 result |= TRACK_SESSION; 1460 break; 1461 } 1462 } 1463 1464 return result; 1465} 1466 1467uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1468{ 1469 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1470 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1471 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1472 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1473 } 1474 for (size_t i = 0; i < mTracks.size(); i++) { 1475 sp<Track> track = mTracks[i]; 1476 if (sessionId == track->sessionId() && 1477 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1478 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1479 } 1480 } 1481 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1482} 1483 1484 1485// ---------------------------------------------------------------------------- 1486 1487AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1488 : PlaybackThread(audioFlinger, output, id, device), 1489 mAudioMixer(0) 1490{ 1491 mType = ThreadBase::MIXER; 1492 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1493 1494 // FIXME - Current mixer implementation only supports stereo output 1495 if (mChannelCount == 1) { 1496 LOGE("Invalid audio hardware channel count"); 1497 } 1498} 1499 1500AudioFlinger::MixerThread::~MixerThread() 1501{ 1502 delete mAudioMixer; 1503} 1504 1505bool AudioFlinger::MixerThread::threadLoop() 1506{ 1507 Vector< sp<Track> > tracksToRemove; 1508 uint32_t mixerStatus = MIXER_IDLE; 1509 nsecs_t standbyTime = systemTime(); 1510 size_t mixBufferSize = mFrameCount * mFrameSize; 1511 // FIXME: Relaxed timing because of a certain device that can't meet latency 1512 // Should be reduced to 2x after the vendor fixes the driver issue 1513 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1514 nsecs_t lastWarning = 0; 1515 bool longStandbyExit = false; 1516 uint32_t activeSleepTime = activeSleepTimeUs(); 1517 uint32_t idleSleepTime = idleSleepTimeUs(); 1518 uint32_t sleepTime = idleSleepTime; 1519 Vector< sp<EffectChain> > effectChains; 1520#ifdef DEBUG_CPU_USAGE 1521 ThreadCpuUsage cpu; 1522 const CentralTendencyStatistics& stats = cpu.statistics(); 1523#endif 1524 1525 while (!exitPending()) 1526 { 1527#ifdef DEBUG_CPU_USAGE 1528 cpu.sampleAndEnable(); 1529 unsigned n = stats.n(); 1530 // cpu.elapsed() is expensive, so don't call it every loop 1531 if ((n & 127) == 1) { 1532 long long elapsed = cpu.elapsed(); 1533 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1534 double perLoop = elapsed / (double) n; 1535 double perLoop100 = perLoop * 0.01; 1536 double mean = stats.mean(); 1537 double stddev = stats.stddev(); 1538 double minimum = stats.minimum(); 1539 double maximum = stats.maximum(); 1540 cpu.resetStatistics(); 1541 LOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1542 elapsed * .000000001, n, perLoop * .000001, 1543 mean * .001, 1544 stddev * .001, 1545 minimum * .001, 1546 maximum * .001, 1547 mean / perLoop100, 1548 stddev / perLoop100, 1549 minimum / perLoop100, 1550 maximum / perLoop100); 1551 } 1552 } 1553#endif 1554 processConfigEvents(); 1555 1556 mixerStatus = MIXER_IDLE; 1557 { // scope for mLock 1558 1559 Mutex::Autolock _l(mLock); 1560 1561 if (checkForNewParameters_l()) { 1562 mixBufferSize = mFrameCount * mFrameSize; 1563 // FIXME: Relaxed timing because of a certain device that can't meet latency 1564 // Should be reduced to 2x after the vendor fixes the driver issue 1565 maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1566 activeSleepTime = activeSleepTimeUs(); 1567 idleSleepTime = idleSleepTimeUs(); 1568 } 1569 1570 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1571 1572 // put audio hardware into standby after short delay 1573 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1574 mSuspended) { 1575 if (!mStandby) { 1576 LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1577 mOutput->stream->common.standby(&mOutput->stream->common); 1578 mStandby = true; 1579 mBytesWritten = 0; 1580 } 1581 1582 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1583 // we're about to wait, flush the binder command buffer 1584 IPCThreadState::self()->flushCommands(); 1585 1586 if (exitPending()) break; 1587 1588 // wait until we have something to do... 1589 LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1590 mWaitWorkCV.wait(mLock); 1591 LOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1592 1593 if (mMasterMute == false) { 1594 char value[PROPERTY_VALUE_MAX]; 1595 property_get("ro.audio.silent", value, "0"); 1596 if (atoi(value)) { 1597 LOGD("Silence is golden"); 1598 setMasterMute(true); 1599 } 1600 } 1601 1602 standbyTime = systemTime() + kStandbyTimeInNsecs; 1603 sleepTime = idleSleepTime; 1604 continue; 1605 } 1606 } 1607 1608 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1609 1610 // prevent any changes in effect chain list and in each effect chain 1611 // during mixing and effect process as the audio buffers could be deleted 1612 // or modified if an effect is created or deleted 1613 lockEffectChains_l(effectChains); 1614 } 1615 1616 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1617 // mix buffers... 1618 mAudioMixer->process(); 1619 sleepTime = 0; 1620 standbyTime = systemTime() + kStandbyTimeInNsecs; 1621 //TODO: delay standby when effects have a tail 1622 } else { 1623 // If no tracks are ready, sleep once for the duration of an output 1624 // buffer size, then write 0s to the output 1625 if (sleepTime == 0) { 1626 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1627 sleepTime = activeSleepTime; 1628 } else { 1629 sleepTime = idleSleepTime; 1630 } 1631 } else if (mBytesWritten != 0 || 1632 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 1633 memset (mMixBuffer, 0, mixBufferSize); 1634 sleepTime = 0; 1635 LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 1636 } 1637 // TODO add standby time extension fct of effect tail 1638 } 1639 1640 if (mSuspended) { 1641 sleepTime = suspendSleepTimeUs(); 1642 } 1643 // sleepTime == 0 means we must write to audio hardware 1644 if (sleepTime == 0) { 1645 for (size_t i = 0; i < effectChains.size(); i ++) { 1646 effectChains[i]->process_l(); 1647 } 1648 // enable changes in effect chain 1649 unlockEffectChains(effectChains); 1650 mLastWriteTime = systemTime(); 1651 mInWrite = true; 1652 mBytesWritten += mixBufferSize; 1653 1654 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1655 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 1656 mNumWrites++; 1657 mInWrite = false; 1658 nsecs_t now = systemTime(); 1659 nsecs_t delta = now - mLastWriteTime; 1660 if (delta > maxPeriod) { 1661 mNumDelayedWrites++; 1662 if ((now - lastWarning) > kWarningThrottle) { 1663 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 1664 ns2ms(delta), mNumDelayedWrites, this); 1665 lastWarning = now; 1666 } 1667 if (mStandby) { 1668 longStandbyExit = true; 1669 } 1670 } 1671 mStandby = false; 1672 } else { 1673 // enable changes in effect chain 1674 unlockEffectChains(effectChains); 1675 usleep(sleepTime); 1676 } 1677 1678 // finally let go of all our tracks, without the lock held 1679 // since we can't guarantee the destructors won't acquire that 1680 // same lock. 1681 tracksToRemove.clear(); 1682 1683 // Effect chains will be actually deleted here if they were removed from 1684 // mEffectChains list during mixing or effects processing 1685 effectChains.clear(); 1686 } 1687 1688 if (!mStandby) { 1689 mOutput->stream->common.standby(&mOutput->stream->common); 1690 } 1691 1692 LOGV("MixerThread %p exiting", this); 1693 return false; 1694} 1695 1696// prepareTracks_l() must be called with ThreadBase::mLock held 1697uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 1698{ 1699 1700 uint32_t mixerStatus = MIXER_IDLE; 1701 // find out which tracks need to be processed 1702 size_t count = activeTracks.size(); 1703 size_t mixedTracks = 0; 1704 size_t tracksWithEffect = 0; 1705 1706 float masterVolume = mMasterVolume; 1707 bool masterMute = mMasterMute; 1708 1709 if (masterMute) { 1710 masterVolume = 0; 1711 } 1712 // Delegate master volume control to effect in output mix effect chain if needed 1713 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 1714 if (chain != 0) { 1715 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 1716 chain->setVolume_l(&v, &v); 1717 masterVolume = (float)((v + (1 << 23)) >> 24); 1718 chain.clear(); 1719 } 1720 1721 for (size_t i=0 ; i<count ; i++) { 1722 sp<Track> t = activeTracks[i].promote(); 1723 if (t == 0) continue; 1724 1725 Track* const track = t.get(); 1726 audio_track_cblk_t* cblk = track->cblk(); 1727 1728 // The first time a track is added we wait 1729 // for all its buffers to be filled before processing it 1730 mAudioMixer->setActiveTrack(track->name()); 1731 if (cblk->framesReady() && track->isReady() && 1732 !track->isPaused() && !track->isTerminated()) 1733 { 1734 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 1735 1736 mixedTracks++; 1737 1738 // track->mainBuffer() != mMixBuffer means there is an effect chain 1739 // connected to the track 1740 chain.clear(); 1741 if (track->mainBuffer() != mMixBuffer) { 1742 chain = getEffectChain_l(track->sessionId()); 1743 // Delegate volume control to effect in track effect chain if needed 1744 if (chain != 0) { 1745 tracksWithEffect++; 1746 } else { 1747 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 1748 track->name(), track->sessionId()); 1749 } 1750 } 1751 1752 1753 int param = AudioMixer::VOLUME; 1754 if (track->mFillingUpStatus == Track::FS_FILLED) { 1755 // no ramp for the first volume setting 1756 track->mFillingUpStatus = Track::FS_ACTIVE; 1757 if (track->mState == TrackBase::RESUMING) { 1758 track->mState = TrackBase::ACTIVE; 1759 param = AudioMixer::RAMP_VOLUME; 1760 } 1761 mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 1762 } else if (cblk->server != 0) { 1763 // If the track is stopped before the first frame was mixed, 1764 // do not apply ramp 1765 param = AudioMixer::RAMP_VOLUME; 1766 } 1767 1768 // compute volume for this track 1769 uint32_t vl, vr, va; 1770 if (track->isMuted() || track->isPausing() || 1771 mStreamTypes[track->type()].mute) { 1772 vl = vr = va = 0; 1773 if (track->isPausing()) { 1774 track->setPaused(); 1775 } 1776 } else { 1777 1778 // read original volumes with volume control 1779 float typeVolume = mStreamTypes[track->type()].volume; 1780 float v = masterVolume * typeVolume; 1781 vl = (uint32_t)(v * cblk->volume[0]) << 12; 1782 vr = (uint32_t)(v * cblk->volume[1]) << 12; 1783 1784 va = (uint32_t)(v * cblk->sendLevel); 1785 } 1786 // Delegate volume control to effect in track effect chain if needed 1787 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1788 // Do not ramp volume if volume is controlled by effect 1789 param = AudioMixer::VOLUME; 1790 track->mHasVolumeController = true; 1791 } else { 1792 // force no volume ramp when volume controller was just disabled or removed 1793 // from effect chain to avoid volume spike 1794 if (track->mHasVolumeController) { 1795 param = AudioMixer::VOLUME; 1796 } 1797 track->mHasVolumeController = false; 1798 } 1799 1800 // Convert volumes from 8.24 to 4.12 format 1801 int16_t left, right, aux; 1802 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1803 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1804 left = int16_t(v_clamped); 1805 v_clamped = (vr + (1 << 11)) >> 12; 1806 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1807 right = int16_t(v_clamped); 1808 1809 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 1810 aux = int16_t(va); 1811 1812 // XXX: these things DON'T need to be done each time 1813 mAudioMixer->setBufferProvider(track); 1814 mAudioMixer->enable(AudioMixer::MIXING); 1815 1816 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1817 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1818 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1819 mAudioMixer->setParameter( 1820 AudioMixer::TRACK, 1821 AudioMixer::FORMAT, (void *)track->format()); 1822 mAudioMixer->setParameter( 1823 AudioMixer::TRACK, 1824 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 1825 mAudioMixer->setParameter( 1826 AudioMixer::RESAMPLE, 1827 AudioMixer::SAMPLE_RATE, 1828 (void *)(cblk->sampleRate)); 1829 mAudioMixer->setParameter( 1830 AudioMixer::TRACK, 1831 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1832 mAudioMixer->setParameter( 1833 AudioMixer::TRACK, 1834 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1835 1836 // reset retry count 1837 track->mRetryCount = kMaxTrackRetries; 1838 mixerStatus = MIXER_TRACKS_READY; 1839 } else { 1840 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1841 if (track->isStopped()) { 1842 track->reset(); 1843 } 1844 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1845 // We have consumed all the buffers of this track. 1846 // Remove it from the list of active tracks. 1847 tracksToRemove->add(track); 1848 } else { 1849 // No buffers for this track. Give it a few chances to 1850 // fill a buffer, then remove it from active list. 1851 if (--(track->mRetryCount) <= 0) { 1852 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1853 tracksToRemove->add(track); 1854 // indicate to client process that the track was disabled because of underrun 1855 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 1856 } else if (mixerStatus != MIXER_TRACKS_READY) { 1857 mixerStatus = MIXER_TRACKS_ENABLED; 1858 } 1859 } 1860 mAudioMixer->disable(AudioMixer::MIXING); 1861 } 1862 } 1863 1864 // remove all the tracks that need to be... 1865 count = tracksToRemove->size(); 1866 if (UNLIKELY(count)) { 1867 for (size_t i=0 ; i<count ; i++) { 1868 const sp<Track>& track = tracksToRemove->itemAt(i); 1869 mActiveTracks.remove(track); 1870 if (track->mainBuffer() != mMixBuffer) { 1871 chain = getEffectChain_l(track->sessionId()); 1872 if (chain != 0) { 1873 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1874 chain->decActiveTrackCnt(); 1875 } 1876 } 1877 if (track->isTerminated()) { 1878 removeTrack_l(track); 1879 } 1880 } 1881 } 1882 1883 // mix buffer must be cleared if all tracks are connected to an 1884 // effect chain as in this case the mixer will not write to 1885 // mix buffer and track effects will accumulate into it 1886 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1887 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1888 } 1889 1890 return mixerStatus; 1891} 1892 1893void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1894{ 1895 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1896 this, streamType, mTracks.size()); 1897 Mutex::Autolock _l(mLock); 1898 1899 size_t size = mTracks.size(); 1900 for (size_t i = 0; i < size; i++) { 1901 sp<Track> t = mTracks[i]; 1902 if (t->type() == streamType) { 1903 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 1904 t->mCblk->cv.signal(); 1905 } 1906 } 1907} 1908 1909 1910// getTrackName_l() must be called with ThreadBase::mLock held 1911int AudioFlinger::MixerThread::getTrackName_l() 1912{ 1913 return mAudioMixer->getTrackName(); 1914} 1915 1916// deleteTrackName_l() must be called with ThreadBase::mLock held 1917void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1918{ 1919 LOGV("remove track (%d) and delete from mixer", name); 1920 mAudioMixer->deleteTrackName(name); 1921} 1922 1923// checkForNewParameters_l() must be called with ThreadBase::mLock held 1924bool AudioFlinger::MixerThread::checkForNewParameters_l() 1925{ 1926 bool reconfig = false; 1927 1928 while (!mNewParameters.isEmpty()) { 1929 status_t status = NO_ERROR; 1930 String8 keyValuePair = mNewParameters[0]; 1931 AudioParameter param = AudioParameter(keyValuePair); 1932 int value; 1933 1934 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1935 reconfig = true; 1936 } 1937 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1938 if (value != AUDIO_FORMAT_PCM_16_BIT) { 1939 status = BAD_VALUE; 1940 } else { 1941 reconfig = true; 1942 } 1943 } 1944 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1945 if (value != AUDIO_CHANNEL_OUT_STEREO) { 1946 status = BAD_VALUE; 1947 } else { 1948 reconfig = true; 1949 } 1950 } 1951 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1952 // do not accept frame count changes if tracks are open as the track buffer 1953 // size depends on frame count and correct behavior would not be garantied 1954 // if frame count is changed after track creation 1955 if (!mTracks.isEmpty()) { 1956 status = INVALID_OPERATION; 1957 } else { 1958 reconfig = true; 1959 } 1960 } 1961 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1962 // when changing the audio output device, call addBatteryData to notify 1963 // the change 1964 if ((int)mDevice != value) { 1965 uint32_t params = 0; 1966 // check whether speaker is on 1967 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 1968 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 1969 } 1970 1971 int deviceWithoutSpeaker 1972 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 1973 // check if any other device (except speaker) is on 1974 if (value & deviceWithoutSpeaker ) { 1975 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 1976 } 1977 1978 if (params != 0) { 1979 addBatteryData(params); 1980 } 1981 } 1982 1983 // forward device change to effects that have requested to be 1984 // aware of attached audio device. 1985 mDevice = (uint32_t)value; 1986 for (size_t i = 0; i < mEffectChains.size(); i++) { 1987 mEffectChains[i]->setDevice_l(mDevice); 1988 } 1989 } 1990 1991 if (status == NO_ERROR) { 1992 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 1993 keyValuePair.string()); 1994 if (!mStandby && status == INVALID_OPERATION) { 1995 mOutput->stream->common.standby(&mOutput->stream->common); 1996 mStandby = true; 1997 mBytesWritten = 0; 1998 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 1999 keyValuePair.string()); 2000 } 2001 if (status == NO_ERROR && reconfig) { 2002 delete mAudioMixer; 2003 readOutputParameters(); 2004 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2005 for (size_t i = 0; i < mTracks.size() ; i++) { 2006 int name = getTrackName_l(); 2007 if (name < 0) break; 2008 mTracks[i]->mName = name; 2009 // limit track sample rate to 2 x new output sample rate 2010 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2011 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2012 } 2013 } 2014 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2015 } 2016 } 2017 2018 mNewParameters.removeAt(0); 2019 2020 mParamStatus = status; 2021 mParamCond.signal(); 2022 mWaitWorkCV.wait(mLock); 2023 } 2024 return reconfig; 2025} 2026 2027status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2028{ 2029 const size_t SIZE = 256; 2030 char buffer[SIZE]; 2031 String8 result; 2032 2033 PlaybackThread::dumpInternals(fd, args); 2034 2035 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2036 result.append(buffer); 2037 write(fd, result.string(), result.size()); 2038 return NO_ERROR; 2039} 2040 2041uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 2042{ 2043 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2044} 2045 2046uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2047{ 2048 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2049} 2050 2051uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2052{ 2053 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2054} 2055 2056// ---------------------------------------------------------------------------- 2057AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2058 : PlaybackThread(audioFlinger, output, id, device) 2059{ 2060 mType = ThreadBase::DIRECT; 2061} 2062 2063AudioFlinger::DirectOutputThread::~DirectOutputThread() 2064{ 2065} 2066 2067 2068static inline int16_t clamp16(int32_t sample) 2069{ 2070 if ((sample>>15) ^ (sample>>31)) 2071 sample = 0x7FFF ^ (sample>>31); 2072 return sample; 2073} 2074 2075static inline 2076int32_t mul(int16_t in, int16_t v) 2077{ 2078#if defined(__arm__) && !defined(__thumb__) 2079 int32_t out; 2080 asm( "smulbb %[out], %[in], %[v] \n" 2081 : [out]"=r"(out) 2082 : [in]"%r"(in), [v]"r"(v) 2083 : ); 2084 return out; 2085#else 2086 return in * int32_t(v); 2087#endif 2088} 2089 2090void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2091{ 2092 // Do not apply volume on compressed audio 2093 if (!audio_is_linear_pcm(mFormat)) { 2094 return; 2095 } 2096 2097 // convert to signed 16 bit before volume calculation 2098 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2099 size_t count = mFrameCount * mChannelCount; 2100 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2101 int16_t *dst = mMixBuffer + count-1; 2102 while(count--) { 2103 *dst-- = (int16_t)(*src--^0x80) << 8; 2104 } 2105 } 2106 2107 size_t frameCount = mFrameCount; 2108 int16_t *out = mMixBuffer; 2109 if (ramp) { 2110 if (mChannelCount == 1) { 2111 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2112 int32_t vlInc = d / (int32_t)frameCount; 2113 int32_t vl = ((int32_t)mLeftVolShort << 16); 2114 do { 2115 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2116 out++; 2117 vl += vlInc; 2118 } while (--frameCount); 2119 2120 } else { 2121 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2122 int32_t vlInc = d / (int32_t)frameCount; 2123 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2124 int32_t vrInc = d / (int32_t)frameCount; 2125 int32_t vl = ((int32_t)mLeftVolShort << 16); 2126 int32_t vr = ((int32_t)mRightVolShort << 16); 2127 do { 2128 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2129 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2130 out += 2; 2131 vl += vlInc; 2132 vr += vrInc; 2133 } while (--frameCount); 2134 } 2135 } else { 2136 if (mChannelCount == 1) { 2137 do { 2138 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2139 out++; 2140 } while (--frameCount); 2141 } else { 2142 do { 2143 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2144 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2145 out += 2; 2146 } while (--frameCount); 2147 } 2148 } 2149 2150 // convert back to unsigned 8 bit after volume calculation 2151 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2152 size_t count = mFrameCount * mChannelCount; 2153 int16_t *src = mMixBuffer; 2154 uint8_t *dst = (uint8_t *)mMixBuffer; 2155 while(count--) { 2156 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2157 } 2158 } 2159 2160 mLeftVolShort = leftVol; 2161 mRightVolShort = rightVol; 2162} 2163 2164bool AudioFlinger::DirectOutputThread::threadLoop() 2165{ 2166 uint32_t mixerStatus = MIXER_IDLE; 2167 sp<Track> trackToRemove; 2168 sp<Track> activeTrack; 2169 nsecs_t standbyTime = systemTime(); 2170 int8_t *curBuf; 2171 size_t mixBufferSize = mFrameCount*mFrameSize; 2172 uint32_t activeSleepTime = activeSleepTimeUs(); 2173 uint32_t idleSleepTime = idleSleepTimeUs(); 2174 uint32_t sleepTime = idleSleepTime; 2175 // use shorter standby delay as on normal output to release 2176 // hardware resources as soon as possible 2177 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2178 2179 while (!exitPending()) 2180 { 2181 bool rampVolume; 2182 uint16_t leftVol; 2183 uint16_t rightVol; 2184 Vector< sp<EffectChain> > effectChains; 2185 2186 processConfigEvents(); 2187 2188 mixerStatus = MIXER_IDLE; 2189 2190 { // scope for the mLock 2191 2192 Mutex::Autolock _l(mLock); 2193 2194 if (checkForNewParameters_l()) { 2195 mixBufferSize = mFrameCount*mFrameSize; 2196 activeSleepTime = activeSleepTimeUs(); 2197 idleSleepTime = idleSleepTimeUs(); 2198 standbyDelay = microseconds(activeSleepTime*2); 2199 } 2200 2201 // put audio hardware into standby after short delay 2202 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2203 mSuspended) { 2204 // wait until we have something to do... 2205 if (!mStandby) { 2206 LOGV("Audio hardware entering standby, mixer %p\n", this); 2207 mOutput->stream->common.standby(&mOutput->stream->common); 2208 mStandby = true; 2209 mBytesWritten = 0; 2210 } 2211 2212 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2213 // we're about to wait, flush the binder command buffer 2214 IPCThreadState::self()->flushCommands(); 2215 2216 if (exitPending()) break; 2217 2218 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2219 mWaitWorkCV.wait(mLock); 2220 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2221 2222 if (mMasterMute == false) { 2223 char value[PROPERTY_VALUE_MAX]; 2224 property_get("ro.audio.silent", value, "0"); 2225 if (atoi(value)) { 2226 LOGD("Silence is golden"); 2227 setMasterMute(true); 2228 } 2229 } 2230 2231 standbyTime = systemTime() + standbyDelay; 2232 sleepTime = idleSleepTime; 2233 continue; 2234 } 2235 } 2236 2237 effectChains = mEffectChains; 2238 2239 // find out which tracks need to be processed 2240 if (mActiveTracks.size() != 0) { 2241 sp<Track> t = mActiveTracks[0].promote(); 2242 if (t == 0) continue; 2243 2244 Track* const track = t.get(); 2245 audio_track_cblk_t* cblk = track->cblk(); 2246 2247 // The first time a track is added we wait 2248 // for all its buffers to be filled before processing it 2249 if (cblk->framesReady() && track->isReady() && 2250 !track->isPaused() && !track->isTerminated()) 2251 { 2252 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2253 2254 if (track->mFillingUpStatus == Track::FS_FILLED) { 2255 track->mFillingUpStatus = Track::FS_ACTIVE; 2256 mLeftVolFloat = mRightVolFloat = 0; 2257 mLeftVolShort = mRightVolShort = 0; 2258 if (track->mState == TrackBase::RESUMING) { 2259 track->mState = TrackBase::ACTIVE; 2260 rampVolume = true; 2261 } 2262 } else if (cblk->server != 0) { 2263 // If the track is stopped before the first frame was mixed, 2264 // do not apply ramp 2265 rampVolume = true; 2266 } 2267 // compute volume for this track 2268 float left, right; 2269 if (track->isMuted() || mMasterMute || track->isPausing() || 2270 mStreamTypes[track->type()].mute) { 2271 left = right = 0; 2272 if (track->isPausing()) { 2273 track->setPaused(); 2274 } 2275 } else { 2276 float typeVolume = mStreamTypes[track->type()].volume; 2277 float v = mMasterVolume * typeVolume; 2278 float v_clamped = v * cblk->volume[0]; 2279 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2280 left = v_clamped/MAX_GAIN; 2281 v_clamped = v * cblk->volume[1]; 2282 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2283 right = v_clamped/MAX_GAIN; 2284 } 2285 2286 if (left != mLeftVolFloat || right != mRightVolFloat) { 2287 mLeftVolFloat = left; 2288 mRightVolFloat = right; 2289 2290 // If audio HAL implements volume control, 2291 // force software volume to nominal value 2292 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2293 left = 1.0f; 2294 right = 1.0f; 2295 } 2296 2297 // Convert volumes from float to 8.24 2298 uint32_t vl = (uint32_t)(left * (1 << 24)); 2299 uint32_t vr = (uint32_t)(right * (1 << 24)); 2300 2301 // Delegate volume control to effect in track effect chain if needed 2302 // only one effect chain can be present on DirectOutputThread, so if 2303 // there is one, the track is connected to it 2304 if (!effectChains.isEmpty()) { 2305 // Do not ramp volume if volume is controlled by effect 2306 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2307 rampVolume = false; 2308 } 2309 } 2310 2311 // Convert volumes from 8.24 to 4.12 format 2312 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2313 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2314 leftVol = (uint16_t)v_clamped; 2315 v_clamped = (vr + (1 << 11)) >> 12; 2316 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2317 rightVol = (uint16_t)v_clamped; 2318 } else { 2319 leftVol = mLeftVolShort; 2320 rightVol = mRightVolShort; 2321 rampVolume = false; 2322 } 2323 2324 // reset retry count 2325 track->mRetryCount = kMaxTrackRetriesDirect; 2326 activeTrack = t; 2327 mixerStatus = MIXER_TRACKS_READY; 2328 } else { 2329 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2330 if (track->isStopped()) { 2331 track->reset(); 2332 } 2333 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2334 // We have consumed all the buffers of this track. 2335 // Remove it from the list of active tracks. 2336 trackToRemove = track; 2337 } else { 2338 // No buffers for this track. Give it a few chances to 2339 // fill a buffer, then remove it from active list. 2340 if (--(track->mRetryCount) <= 0) { 2341 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2342 trackToRemove = track; 2343 } else { 2344 mixerStatus = MIXER_TRACKS_ENABLED; 2345 } 2346 } 2347 } 2348 } 2349 2350 // remove all the tracks that need to be... 2351 if (UNLIKELY(trackToRemove != 0)) { 2352 mActiveTracks.remove(trackToRemove); 2353 if (!effectChains.isEmpty()) { 2354 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2355 trackToRemove->sessionId()); 2356 effectChains[0]->decActiveTrackCnt(); 2357 } 2358 if (trackToRemove->isTerminated()) { 2359 removeTrack_l(trackToRemove); 2360 } 2361 } 2362 2363 lockEffectChains_l(effectChains); 2364 } 2365 2366 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2367 AudioBufferProvider::Buffer buffer; 2368 size_t frameCount = mFrameCount; 2369 curBuf = (int8_t *)mMixBuffer; 2370 // output audio to hardware 2371 while (frameCount) { 2372 buffer.frameCount = frameCount; 2373 activeTrack->getNextBuffer(&buffer); 2374 if (UNLIKELY(buffer.raw == 0)) { 2375 memset(curBuf, 0, frameCount * mFrameSize); 2376 break; 2377 } 2378 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2379 frameCount -= buffer.frameCount; 2380 curBuf += buffer.frameCount * mFrameSize; 2381 activeTrack->releaseBuffer(&buffer); 2382 } 2383 sleepTime = 0; 2384 standbyTime = systemTime() + standbyDelay; 2385 } else { 2386 if (sleepTime == 0) { 2387 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2388 sleepTime = activeSleepTime; 2389 } else { 2390 sleepTime = idleSleepTime; 2391 } 2392 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2393 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2394 sleepTime = 0; 2395 } 2396 } 2397 2398 if (mSuspended) { 2399 sleepTime = suspendSleepTimeUs(); 2400 } 2401 // sleepTime == 0 means we must write to audio hardware 2402 if (sleepTime == 0) { 2403 if (mixerStatus == MIXER_TRACKS_READY) { 2404 applyVolume(leftVol, rightVol, rampVolume); 2405 } 2406 for (size_t i = 0; i < effectChains.size(); i ++) { 2407 effectChains[i]->process_l(); 2408 } 2409 unlockEffectChains(effectChains); 2410 2411 mLastWriteTime = systemTime(); 2412 mInWrite = true; 2413 mBytesWritten += mixBufferSize; 2414 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2415 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2416 mNumWrites++; 2417 mInWrite = false; 2418 mStandby = false; 2419 } else { 2420 unlockEffectChains(effectChains); 2421 usleep(sleepTime); 2422 } 2423 2424 // finally let go of removed track, without the lock held 2425 // since we can't guarantee the destructors won't acquire that 2426 // same lock. 2427 trackToRemove.clear(); 2428 activeTrack.clear(); 2429 2430 // Effect chains will be actually deleted here if they were removed from 2431 // mEffectChains list during mixing or effects processing 2432 effectChains.clear(); 2433 } 2434 2435 if (!mStandby) { 2436 mOutput->stream->common.standby(&mOutput->stream->common); 2437 } 2438 2439 LOGV("DirectOutputThread %p exiting", this); 2440 return false; 2441} 2442 2443// getTrackName_l() must be called with ThreadBase::mLock held 2444int AudioFlinger::DirectOutputThread::getTrackName_l() 2445{ 2446 return 0; 2447} 2448 2449// deleteTrackName_l() must be called with ThreadBase::mLock held 2450void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2451{ 2452} 2453 2454// checkForNewParameters_l() must be called with ThreadBase::mLock held 2455bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2456{ 2457 bool reconfig = false; 2458 2459 while (!mNewParameters.isEmpty()) { 2460 status_t status = NO_ERROR; 2461 String8 keyValuePair = mNewParameters[0]; 2462 AudioParameter param = AudioParameter(keyValuePair); 2463 int value; 2464 2465 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2466 // do not accept frame count changes if tracks are open as the track buffer 2467 // size depends on frame count and correct behavior would not be garantied 2468 // if frame count is changed after track creation 2469 if (!mTracks.isEmpty()) { 2470 status = INVALID_OPERATION; 2471 } else { 2472 reconfig = true; 2473 } 2474 } 2475 if (status == NO_ERROR) { 2476 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2477 keyValuePair.string()); 2478 if (!mStandby && status == INVALID_OPERATION) { 2479 mOutput->stream->common.standby(&mOutput->stream->common); 2480 mStandby = true; 2481 mBytesWritten = 0; 2482 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2483 keyValuePair.string()); 2484 } 2485 if (status == NO_ERROR && reconfig) { 2486 readOutputParameters(); 2487 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2488 } 2489 } 2490 2491 mNewParameters.removeAt(0); 2492 2493 mParamStatus = status; 2494 mParamCond.signal(); 2495 mWaitWorkCV.wait(mLock); 2496 } 2497 return reconfig; 2498} 2499 2500uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2501{ 2502 uint32_t time; 2503 if (audio_is_linear_pcm(mFormat)) { 2504 time = (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2505 } else { 2506 time = 10000; 2507 } 2508 return time; 2509} 2510 2511uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2512{ 2513 uint32_t time; 2514 if (audio_is_linear_pcm(mFormat)) { 2515 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2516 } else { 2517 time = 10000; 2518 } 2519 return time; 2520} 2521 2522uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2523{ 2524 uint32_t time; 2525 if (audio_is_linear_pcm(mFormat)) { 2526 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2527 } else { 2528 time = 10000; 2529 } 2530 return time; 2531} 2532 2533 2534// ---------------------------------------------------------------------------- 2535 2536AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2537 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2538{ 2539 mType = ThreadBase::DUPLICATING; 2540 addOutputTrack(mainThread); 2541} 2542 2543AudioFlinger::DuplicatingThread::~DuplicatingThread() 2544{ 2545 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2546 mOutputTracks[i]->destroy(); 2547 } 2548 mOutputTracks.clear(); 2549} 2550 2551bool AudioFlinger::DuplicatingThread::threadLoop() 2552{ 2553 Vector< sp<Track> > tracksToRemove; 2554 uint32_t mixerStatus = MIXER_IDLE; 2555 nsecs_t standbyTime = systemTime(); 2556 size_t mixBufferSize = mFrameCount*mFrameSize; 2557 SortedVector< sp<OutputTrack> > outputTracks; 2558 uint32_t writeFrames = 0; 2559 uint32_t activeSleepTime = activeSleepTimeUs(); 2560 uint32_t idleSleepTime = idleSleepTimeUs(); 2561 uint32_t sleepTime = idleSleepTime; 2562 Vector< sp<EffectChain> > effectChains; 2563 2564 while (!exitPending()) 2565 { 2566 processConfigEvents(); 2567 2568 mixerStatus = MIXER_IDLE; 2569 { // scope for the mLock 2570 2571 Mutex::Autolock _l(mLock); 2572 2573 if (checkForNewParameters_l()) { 2574 mixBufferSize = mFrameCount*mFrameSize; 2575 updateWaitTime(); 2576 activeSleepTime = activeSleepTimeUs(); 2577 idleSleepTime = idleSleepTimeUs(); 2578 } 2579 2580 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2581 2582 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2583 outputTracks.add(mOutputTracks[i]); 2584 } 2585 2586 // put audio hardware into standby after short delay 2587 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2588 mSuspended) { 2589 if (!mStandby) { 2590 for (size_t i = 0; i < outputTracks.size(); i++) { 2591 outputTracks[i]->stop(); 2592 } 2593 mStandby = true; 2594 mBytesWritten = 0; 2595 } 2596 2597 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2598 // we're about to wait, flush the binder command buffer 2599 IPCThreadState::self()->flushCommands(); 2600 outputTracks.clear(); 2601 2602 if (exitPending()) break; 2603 2604 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2605 mWaitWorkCV.wait(mLock); 2606 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2607 if (mMasterMute == false) { 2608 char value[PROPERTY_VALUE_MAX]; 2609 property_get("ro.audio.silent", value, "0"); 2610 if (atoi(value)) { 2611 LOGD("Silence is golden"); 2612 setMasterMute(true); 2613 } 2614 } 2615 2616 standbyTime = systemTime() + kStandbyTimeInNsecs; 2617 sleepTime = idleSleepTime; 2618 continue; 2619 } 2620 } 2621 2622 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2623 2624 // prevent any changes in effect chain list and in each effect chain 2625 // during mixing and effect process as the audio buffers could be deleted 2626 // or modified if an effect is created or deleted 2627 lockEffectChains_l(effectChains); 2628 } 2629 2630 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2631 // mix buffers... 2632 if (outputsReady(outputTracks)) { 2633 mAudioMixer->process(); 2634 } else { 2635 memset(mMixBuffer, 0, mixBufferSize); 2636 } 2637 sleepTime = 0; 2638 writeFrames = mFrameCount; 2639 } else { 2640 if (sleepTime == 0) { 2641 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2642 sleepTime = activeSleepTime; 2643 } else { 2644 sleepTime = idleSleepTime; 2645 } 2646 } else if (mBytesWritten != 0) { 2647 // flush remaining overflow buffers in output tracks 2648 for (size_t i = 0; i < outputTracks.size(); i++) { 2649 if (outputTracks[i]->isActive()) { 2650 sleepTime = 0; 2651 writeFrames = 0; 2652 memset(mMixBuffer, 0, mixBufferSize); 2653 break; 2654 } 2655 } 2656 } 2657 } 2658 2659 if (mSuspended) { 2660 sleepTime = suspendSleepTimeUs(); 2661 } 2662 // sleepTime == 0 means we must write to audio hardware 2663 if (sleepTime == 0) { 2664 for (size_t i = 0; i < effectChains.size(); i ++) { 2665 effectChains[i]->process_l(); 2666 } 2667 // enable changes in effect chain 2668 unlockEffectChains(effectChains); 2669 2670 standbyTime = systemTime() + kStandbyTimeInNsecs; 2671 for (size_t i = 0; i < outputTracks.size(); i++) { 2672 outputTracks[i]->write(mMixBuffer, writeFrames); 2673 } 2674 mStandby = false; 2675 mBytesWritten += mixBufferSize; 2676 } else { 2677 // enable changes in effect chain 2678 unlockEffectChains(effectChains); 2679 usleep(sleepTime); 2680 } 2681 2682 // finally let go of all our tracks, without the lock held 2683 // since we can't guarantee the destructors won't acquire that 2684 // same lock. 2685 tracksToRemove.clear(); 2686 outputTracks.clear(); 2687 2688 // Effect chains will be actually deleted here if they were removed from 2689 // mEffectChains list during mixing or effects processing 2690 effectChains.clear(); 2691 } 2692 2693 return false; 2694} 2695 2696void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2697{ 2698 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2699 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2700 this, 2701 mSampleRate, 2702 mFormat, 2703 mChannelMask, 2704 frameCount); 2705 if (outputTrack->cblk() != NULL) { 2706 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 2707 mOutputTracks.add(outputTrack); 2708 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2709 updateWaitTime(); 2710 } 2711} 2712 2713void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2714{ 2715 Mutex::Autolock _l(mLock); 2716 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2717 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2718 mOutputTracks[i]->destroy(); 2719 mOutputTracks.removeAt(i); 2720 updateWaitTime(); 2721 return; 2722 } 2723 } 2724 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2725} 2726 2727void AudioFlinger::DuplicatingThread::updateWaitTime() 2728{ 2729 mWaitTimeMs = UINT_MAX; 2730 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2731 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2732 if (strong != NULL) { 2733 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2734 if (waitTimeMs < mWaitTimeMs) { 2735 mWaitTimeMs = waitTimeMs; 2736 } 2737 } 2738 } 2739} 2740 2741 2742bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2743{ 2744 for (size_t i = 0; i < outputTracks.size(); i++) { 2745 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2746 if (thread == 0) { 2747 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2748 return false; 2749 } 2750 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2751 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2752 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2753 return false; 2754 } 2755 } 2756 return true; 2757} 2758 2759uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2760{ 2761 return (mWaitTimeMs * 1000) / 2; 2762} 2763 2764// ---------------------------------------------------------------------------- 2765 2766// TrackBase constructor must be called with AudioFlinger::mLock held 2767AudioFlinger::ThreadBase::TrackBase::TrackBase( 2768 const wp<ThreadBase>& thread, 2769 const sp<Client>& client, 2770 uint32_t sampleRate, 2771 uint32_t format, 2772 uint32_t channelMask, 2773 int frameCount, 2774 uint32_t flags, 2775 const sp<IMemory>& sharedBuffer, 2776 int sessionId) 2777 : RefBase(), 2778 mThread(thread), 2779 mClient(client), 2780 mCblk(0), 2781 mFrameCount(0), 2782 mState(IDLE), 2783 mClientTid(-1), 2784 mFormat(format), 2785 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2786 mSessionId(sessionId) 2787{ 2788 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2789 2790 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2791 size_t size = sizeof(audio_track_cblk_t); 2792 uint8_t channelCount = popcount(channelMask); 2793 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2794 if (sharedBuffer == 0) { 2795 size += bufferSize; 2796 } 2797 2798 if (client != NULL) { 2799 mCblkMemory = client->heap()->allocate(size); 2800 if (mCblkMemory != 0) { 2801 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2802 if (mCblk) { // construct the shared structure in-place. 2803 new(mCblk) audio_track_cblk_t(); 2804 // clear all buffers 2805 mCblk->frameCount = frameCount; 2806 mCblk->sampleRate = sampleRate; 2807 mChannelCount = channelCount; 2808 mChannelMask = channelMask; 2809 if (sharedBuffer == 0) { 2810 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2811 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2812 // Force underrun condition to avoid false underrun callback until first data is 2813 // written to buffer (other flags are cleared) 2814 mCblk->flags = CBLK_UNDERRUN_ON; 2815 } else { 2816 mBuffer = sharedBuffer->pointer(); 2817 } 2818 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2819 } 2820 } else { 2821 LOGE("not enough memory for AudioTrack size=%u", size); 2822 client->heap()->dump("AudioTrack"); 2823 return; 2824 } 2825 } else { 2826 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2827 if (mCblk) { // construct the shared structure in-place. 2828 new(mCblk) audio_track_cblk_t(); 2829 // clear all buffers 2830 mCblk->frameCount = frameCount; 2831 mCblk->sampleRate = sampleRate; 2832 mChannelCount = channelCount; 2833 mChannelMask = channelMask; 2834 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2835 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2836 // Force underrun condition to avoid false underrun callback until first data is 2837 // written to buffer (other flags are cleared) 2838 mCblk->flags = CBLK_UNDERRUN_ON; 2839 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2840 } 2841 } 2842} 2843 2844AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2845{ 2846 if (mCblk) { 2847 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2848 if (mClient == NULL) { 2849 delete mCblk; 2850 } 2851 } 2852 mCblkMemory.clear(); // and free the shared memory 2853 if (mClient != NULL) { 2854 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2855 mClient.clear(); 2856 } 2857} 2858 2859void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2860{ 2861 buffer->raw = 0; 2862 mFrameCount = buffer->frameCount; 2863 step(); 2864 buffer->frameCount = 0; 2865} 2866 2867bool AudioFlinger::ThreadBase::TrackBase::step() { 2868 bool result; 2869 audio_track_cblk_t* cblk = this->cblk(); 2870 2871 result = cblk->stepServer(mFrameCount); 2872 if (!result) { 2873 LOGV("stepServer failed acquiring cblk mutex"); 2874 mFlags |= STEPSERVER_FAILED; 2875 } 2876 return result; 2877} 2878 2879void AudioFlinger::ThreadBase::TrackBase::reset() { 2880 audio_track_cblk_t* cblk = this->cblk(); 2881 2882 cblk->user = 0; 2883 cblk->server = 0; 2884 cblk->userBase = 0; 2885 cblk->serverBase = 0; 2886 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2887 LOGV("TrackBase::reset"); 2888} 2889 2890sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2891{ 2892 return mCblkMemory; 2893} 2894 2895int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2896 return (int)mCblk->sampleRate; 2897} 2898 2899int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2900 return (const int)mChannelCount; 2901} 2902 2903uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 2904 return mChannelMask; 2905} 2906 2907void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2908 audio_track_cblk_t* cblk = this->cblk(); 2909 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2910 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2911 2912 // Check validity of returned pointer in case the track control block would have been corrupted. 2913 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2914 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2915 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2916 server %d, serverBase %d, user %d, userBase %d", 2917 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2918 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 2919 return 0; 2920 } 2921 2922 return bufferStart; 2923} 2924 2925// ---------------------------------------------------------------------------- 2926 2927// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2928AudioFlinger::PlaybackThread::Track::Track( 2929 const wp<ThreadBase>& thread, 2930 const sp<Client>& client, 2931 int streamType, 2932 uint32_t sampleRate, 2933 uint32_t format, 2934 uint32_t channelMask, 2935 int frameCount, 2936 const sp<IMemory>& sharedBuffer, 2937 int sessionId) 2938 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 2939 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 2940 mAuxEffectId(0), mHasVolumeController(false) 2941{ 2942 if (mCblk != NULL) { 2943 sp<ThreadBase> baseThread = thread.promote(); 2944 if (baseThread != 0) { 2945 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2946 mName = playbackThread->getTrackName_l(); 2947 mMainBuffer = playbackThread->mixBuffer(); 2948 } 2949 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2950 if (mName < 0) { 2951 LOGE("no more track names available"); 2952 } 2953 mVolume[0] = 1.0f; 2954 mVolume[1] = 1.0f; 2955 mStreamType = streamType; 2956 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2957 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2958 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 2959 } 2960} 2961 2962AudioFlinger::PlaybackThread::Track::~Track() 2963{ 2964 LOGV("PlaybackThread::Track destructor"); 2965 sp<ThreadBase> thread = mThread.promote(); 2966 if (thread != 0) { 2967 Mutex::Autolock _l(thread->mLock); 2968 mState = TERMINATED; 2969 } 2970} 2971 2972void AudioFlinger::PlaybackThread::Track::destroy() 2973{ 2974 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2975 // by removing it from mTracks vector, so there is a risk that this Tracks's 2976 // desctructor is called. As the destructor needs to lock mLock, 2977 // we must acquire a strong reference on this Track before locking mLock 2978 // here so that the destructor is called only when exiting this function. 2979 // On the other hand, as long as Track::destroy() is only called by 2980 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2981 // this Track with its member mTrack. 2982 sp<Track> keep(this); 2983 { // scope for mLock 2984 sp<ThreadBase> thread = mThread.promote(); 2985 if (thread != 0) { 2986 if (!isOutputTrack()) { 2987 if (mState == ACTIVE || mState == RESUMING) { 2988 AudioSystem::stopOutput(thread->id(), 2989 (audio_stream_type_t)mStreamType, 2990 mSessionId); 2991 2992 // to track the speaker usage 2993 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2994 } 2995 AudioSystem::releaseOutput(thread->id()); 2996 } 2997 Mutex::Autolock _l(thread->mLock); 2998 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2999 playbackThread->destroyTrack_l(this); 3000 } 3001 } 3002} 3003 3004void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3005{ 3006 snprintf(buffer, size, " %05d %05d %03u %03u 0x%08x %05u %04u %1d %1d %1d %05u %05u %05u 0x%08x 0x%08x 0x%08x 0x%08x\n", 3007 mName - AudioMixer::TRACK0, 3008 (mClient == NULL) ? getpid() : mClient->pid(), 3009 mStreamType, 3010 mFormat, 3011 mChannelMask, 3012 mSessionId, 3013 mFrameCount, 3014 mState, 3015 mMute, 3016 mFillingUpStatus, 3017 mCblk->sampleRate, 3018 mCblk->volume[0], 3019 mCblk->volume[1], 3020 mCblk->server, 3021 mCblk->user, 3022 (int)mMainBuffer, 3023 (int)mAuxBuffer); 3024} 3025 3026status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3027{ 3028 audio_track_cblk_t* cblk = this->cblk(); 3029 uint32_t framesReady; 3030 uint32_t framesReq = buffer->frameCount; 3031 3032 // Check if last stepServer failed, try to step now 3033 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3034 if (!step()) goto getNextBuffer_exit; 3035 LOGV("stepServer recovered"); 3036 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3037 } 3038 3039 framesReady = cblk->framesReady(); 3040 3041 if (LIKELY(framesReady)) { 3042 uint32_t s = cblk->server; 3043 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3044 3045 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3046 if (framesReq > framesReady) { 3047 framesReq = framesReady; 3048 } 3049 if (s + framesReq > bufferEnd) { 3050 framesReq = bufferEnd - s; 3051 } 3052 3053 buffer->raw = getBuffer(s, framesReq); 3054 if (buffer->raw == 0) goto getNextBuffer_exit; 3055 3056 buffer->frameCount = framesReq; 3057 return NO_ERROR; 3058 } 3059 3060getNextBuffer_exit: 3061 buffer->raw = 0; 3062 buffer->frameCount = 0; 3063 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3064 return NOT_ENOUGH_DATA; 3065} 3066 3067bool AudioFlinger::PlaybackThread::Track::isReady() const { 3068 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3069 3070 if (mCblk->framesReady() >= mCblk->frameCount || 3071 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3072 mFillingUpStatus = FS_FILLED; 3073 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3074 return true; 3075 } 3076 return false; 3077} 3078 3079status_t AudioFlinger::PlaybackThread::Track::start() 3080{ 3081 status_t status = NO_ERROR; 3082 LOGV("start(%d), calling thread %d session %d", 3083 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3084 sp<ThreadBase> thread = mThread.promote(); 3085 if (thread != 0) { 3086 Mutex::Autolock _l(thread->mLock); 3087 int state = mState; 3088 // here the track could be either new, or restarted 3089 // in both cases "unstop" the track 3090 if (mState == PAUSED) { 3091 mState = TrackBase::RESUMING; 3092 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3093 } else { 3094 mState = TrackBase::ACTIVE; 3095 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3096 } 3097 3098 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3099 thread->mLock.unlock(); 3100 status = AudioSystem::startOutput(thread->id(), 3101 (audio_stream_type_t)mStreamType, 3102 mSessionId); 3103 thread->mLock.lock(); 3104 3105 // to track the speaker usage 3106 if (status == NO_ERROR) { 3107 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3108 } 3109 } 3110 if (status == NO_ERROR) { 3111 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3112 playbackThread->addTrack_l(this); 3113 } else { 3114 mState = state; 3115 } 3116 } else { 3117 status = BAD_VALUE; 3118 } 3119 return status; 3120} 3121 3122void AudioFlinger::PlaybackThread::Track::stop() 3123{ 3124 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3125 sp<ThreadBase> thread = mThread.promote(); 3126 if (thread != 0) { 3127 Mutex::Autolock _l(thread->mLock); 3128 int state = mState; 3129 if (mState > STOPPED) { 3130 mState = STOPPED; 3131 // If the track is not active (PAUSED and buffers full), flush buffers 3132 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3133 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3134 reset(); 3135 } 3136 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3137 } 3138 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3139 thread->mLock.unlock(); 3140 AudioSystem::stopOutput(thread->id(), 3141 (audio_stream_type_t)mStreamType, 3142 mSessionId); 3143 thread->mLock.lock(); 3144 3145 // to track the speaker usage 3146 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3147 } 3148 } 3149} 3150 3151void AudioFlinger::PlaybackThread::Track::pause() 3152{ 3153 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3154 sp<ThreadBase> thread = mThread.promote(); 3155 if (thread != 0) { 3156 Mutex::Autolock _l(thread->mLock); 3157 if (mState == ACTIVE || mState == RESUMING) { 3158 mState = PAUSING; 3159 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3160 if (!isOutputTrack()) { 3161 thread->mLock.unlock(); 3162 AudioSystem::stopOutput(thread->id(), 3163 (audio_stream_type_t)mStreamType, 3164 mSessionId); 3165 thread->mLock.lock(); 3166 3167 // to track the speaker usage 3168 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3169 } 3170 } 3171 } 3172} 3173 3174void AudioFlinger::PlaybackThread::Track::flush() 3175{ 3176 LOGV("flush(%d)", mName); 3177 sp<ThreadBase> thread = mThread.promote(); 3178 if (thread != 0) { 3179 Mutex::Autolock _l(thread->mLock); 3180 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3181 return; 3182 } 3183 // No point remaining in PAUSED state after a flush => go to 3184 // STOPPED state 3185 mState = STOPPED; 3186 3187 // do not reset the track if it is still in the process of being stopped or paused. 3188 // this will be done by prepareTracks_l() when the track is stopped. 3189 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3190 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3191 reset(); 3192 } 3193 } 3194} 3195 3196void AudioFlinger::PlaybackThread::Track::reset() 3197{ 3198 // Do not reset twice to avoid discarding data written just after a flush and before 3199 // the audioflinger thread detects the track is stopped. 3200 if (!mResetDone) { 3201 TrackBase::reset(); 3202 // Force underrun condition to avoid false underrun callback until first data is 3203 // written to buffer 3204 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3205 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3206 mFillingUpStatus = FS_FILLING; 3207 mResetDone = true; 3208 } 3209} 3210 3211void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3212{ 3213 mMute = muted; 3214} 3215 3216void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3217{ 3218 mVolume[0] = left; 3219 mVolume[1] = right; 3220} 3221 3222status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3223{ 3224 status_t status = DEAD_OBJECT; 3225 sp<ThreadBase> thread = mThread.promote(); 3226 if (thread != 0) { 3227 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3228 status = playbackThread->attachAuxEffect(this, EffectId); 3229 } 3230 return status; 3231} 3232 3233void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3234{ 3235 mAuxEffectId = EffectId; 3236 mAuxBuffer = buffer; 3237} 3238 3239// ---------------------------------------------------------------------------- 3240 3241// RecordTrack constructor must be called with AudioFlinger::mLock held 3242AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3243 const wp<ThreadBase>& thread, 3244 const sp<Client>& client, 3245 uint32_t sampleRate, 3246 uint32_t format, 3247 uint32_t channelMask, 3248 int frameCount, 3249 uint32_t flags, 3250 int sessionId) 3251 : TrackBase(thread, client, sampleRate, format, 3252 channelMask, frameCount, flags, 0, sessionId), 3253 mOverflow(false) 3254{ 3255 if (mCblk != NULL) { 3256 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3257 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3258 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3259 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3260 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3261 } else { 3262 mCblk->frameSize = sizeof(int8_t); 3263 } 3264 } 3265} 3266 3267AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3268{ 3269 sp<ThreadBase> thread = mThread.promote(); 3270 if (thread != 0) { 3271 AudioSystem::releaseInput(thread->id()); 3272 } 3273} 3274 3275status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3276{ 3277 audio_track_cblk_t* cblk = this->cblk(); 3278 uint32_t framesAvail; 3279 uint32_t framesReq = buffer->frameCount; 3280 3281 // Check if last stepServer failed, try to step now 3282 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3283 if (!step()) goto getNextBuffer_exit; 3284 LOGV("stepServer recovered"); 3285 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3286 } 3287 3288 framesAvail = cblk->framesAvailable_l(); 3289 3290 if (LIKELY(framesAvail)) { 3291 uint32_t s = cblk->server; 3292 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3293 3294 if (framesReq > framesAvail) { 3295 framesReq = framesAvail; 3296 } 3297 if (s + framesReq > bufferEnd) { 3298 framesReq = bufferEnd - s; 3299 } 3300 3301 buffer->raw = getBuffer(s, framesReq); 3302 if (buffer->raw == 0) goto getNextBuffer_exit; 3303 3304 buffer->frameCount = framesReq; 3305 return NO_ERROR; 3306 } 3307 3308getNextBuffer_exit: 3309 buffer->raw = 0; 3310 buffer->frameCount = 0; 3311 return NOT_ENOUGH_DATA; 3312} 3313 3314status_t AudioFlinger::RecordThread::RecordTrack::start() 3315{ 3316 sp<ThreadBase> thread = mThread.promote(); 3317 if (thread != 0) { 3318 RecordThread *recordThread = (RecordThread *)thread.get(); 3319 return recordThread->start(this); 3320 } else { 3321 return BAD_VALUE; 3322 } 3323} 3324 3325void AudioFlinger::RecordThread::RecordTrack::stop() 3326{ 3327 sp<ThreadBase> thread = mThread.promote(); 3328 if (thread != 0) { 3329 RecordThread *recordThread = (RecordThread *)thread.get(); 3330 recordThread->stop(this); 3331 TrackBase::reset(); 3332 // Force overerrun condition to avoid false overrun callback until first data is 3333 // read from buffer 3334 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3335 } 3336} 3337 3338void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3339{ 3340 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3341 (mClient == NULL) ? getpid() : mClient->pid(), 3342 mFormat, 3343 mChannelMask, 3344 mSessionId, 3345 mFrameCount, 3346 mState, 3347 mCblk->sampleRate, 3348 mCblk->server, 3349 mCblk->user); 3350} 3351 3352 3353// ---------------------------------------------------------------------------- 3354 3355AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3356 const wp<ThreadBase>& thread, 3357 DuplicatingThread *sourceThread, 3358 uint32_t sampleRate, 3359 uint32_t format, 3360 uint32_t channelMask, 3361 int frameCount) 3362 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3363 mActive(false), mSourceThread(sourceThread) 3364{ 3365 3366 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3367 if (mCblk != NULL) { 3368 mCblk->flags |= CBLK_DIRECTION_OUT; 3369 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3370 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3371 mOutBuffer.frameCount = 0; 3372 playbackThread->mTracks.add(this); 3373 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3374 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3375 mCblk, mBuffer, mCblk->buffers, 3376 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3377 } else { 3378 LOGW("Error creating output track on thread %p", playbackThread); 3379 } 3380} 3381 3382AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3383{ 3384 clearBufferQueue(); 3385} 3386 3387status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3388{ 3389 status_t status = Track::start(); 3390 if (status != NO_ERROR) { 3391 return status; 3392 } 3393 3394 mActive = true; 3395 mRetryCount = 127; 3396 return status; 3397} 3398 3399void AudioFlinger::PlaybackThread::OutputTrack::stop() 3400{ 3401 Track::stop(); 3402 clearBufferQueue(); 3403 mOutBuffer.frameCount = 0; 3404 mActive = false; 3405} 3406 3407bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3408{ 3409 Buffer *pInBuffer; 3410 Buffer inBuffer; 3411 uint32_t channelCount = mChannelCount; 3412 bool outputBufferFull = false; 3413 inBuffer.frameCount = frames; 3414 inBuffer.i16 = data; 3415 3416 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3417 3418 if (!mActive && frames != 0) { 3419 start(); 3420 sp<ThreadBase> thread = mThread.promote(); 3421 if (thread != 0) { 3422 MixerThread *mixerThread = (MixerThread *)thread.get(); 3423 if (mCblk->frameCount > frames){ 3424 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3425 uint32_t startFrames = (mCblk->frameCount - frames); 3426 pInBuffer = new Buffer; 3427 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3428 pInBuffer->frameCount = startFrames; 3429 pInBuffer->i16 = pInBuffer->mBuffer; 3430 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3431 mBufferQueue.add(pInBuffer); 3432 } else { 3433 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3434 } 3435 } 3436 } 3437 } 3438 3439 while (waitTimeLeftMs) { 3440 // First write pending buffers, then new data 3441 if (mBufferQueue.size()) { 3442 pInBuffer = mBufferQueue.itemAt(0); 3443 } else { 3444 pInBuffer = &inBuffer; 3445 } 3446 3447 if (pInBuffer->frameCount == 0) { 3448 break; 3449 } 3450 3451 if (mOutBuffer.frameCount == 0) { 3452 mOutBuffer.frameCount = pInBuffer->frameCount; 3453 nsecs_t startTime = systemTime(); 3454 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3455 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3456 outputBufferFull = true; 3457 break; 3458 } 3459 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3460 if (waitTimeLeftMs >= waitTimeMs) { 3461 waitTimeLeftMs -= waitTimeMs; 3462 } else { 3463 waitTimeLeftMs = 0; 3464 } 3465 } 3466 3467 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3468 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3469 mCblk->stepUser(outFrames); 3470 pInBuffer->frameCount -= outFrames; 3471 pInBuffer->i16 += outFrames * channelCount; 3472 mOutBuffer.frameCount -= outFrames; 3473 mOutBuffer.i16 += outFrames * channelCount; 3474 3475 if (pInBuffer->frameCount == 0) { 3476 if (mBufferQueue.size()) { 3477 mBufferQueue.removeAt(0); 3478 delete [] pInBuffer->mBuffer; 3479 delete pInBuffer; 3480 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3481 } else { 3482 break; 3483 } 3484 } 3485 } 3486 3487 // If we could not write all frames, allocate a buffer and queue it for next time. 3488 if (inBuffer.frameCount) { 3489 sp<ThreadBase> thread = mThread.promote(); 3490 if (thread != 0 && !thread->standby()) { 3491 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3492 pInBuffer = new Buffer; 3493 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3494 pInBuffer->frameCount = inBuffer.frameCount; 3495 pInBuffer->i16 = pInBuffer->mBuffer; 3496 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3497 mBufferQueue.add(pInBuffer); 3498 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3499 } else { 3500 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3501 } 3502 } 3503 } 3504 3505 // Calling write() with a 0 length buffer, means that no more data will be written: 3506 // If no more buffers are pending, fill output track buffer to make sure it is started 3507 // by output mixer. 3508 if (frames == 0 && mBufferQueue.size() == 0) { 3509 if (mCblk->user < mCblk->frameCount) { 3510 frames = mCblk->frameCount - mCblk->user; 3511 pInBuffer = new Buffer; 3512 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3513 pInBuffer->frameCount = frames; 3514 pInBuffer->i16 = pInBuffer->mBuffer; 3515 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3516 mBufferQueue.add(pInBuffer); 3517 } else if (mActive) { 3518 stop(); 3519 } 3520 } 3521 3522 return outputBufferFull; 3523} 3524 3525status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3526{ 3527 int active; 3528 status_t result; 3529 audio_track_cblk_t* cblk = mCblk; 3530 uint32_t framesReq = buffer->frameCount; 3531 3532// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3533 buffer->frameCount = 0; 3534 3535 uint32_t framesAvail = cblk->framesAvailable(); 3536 3537 3538 if (framesAvail == 0) { 3539 Mutex::Autolock _l(cblk->lock); 3540 goto start_loop_here; 3541 while (framesAvail == 0) { 3542 active = mActive; 3543 if (UNLIKELY(!active)) { 3544 LOGV("Not active and NO_MORE_BUFFERS"); 3545 return AudioTrack::NO_MORE_BUFFERS; 3546 } 3547 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3548 if (result != NO_ERROR) { 3549 return AudioTrack::NO_MORE_BUFFERS; 3550 } 3551 // read the server count again 3552 start_loop_here: 3553 framesAvail = cblk->framesAvailable_l(); 3554 } 3555 } 3556 3557// if (framesAvail < framesReq) { 3558// return AudioTrack::NO_MORE_BUFFERS; 3559// } 3560 3561 if (framesReq > framesAvail) { 3562 framesReq = framesAvail; 3563 } 3564 3565 uint32_t u = cblk->user; 3566 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3567 3568 if (u + framesReq > bufferEnd) { 3569 framesReq = bufferEnd - u; 3570 } 3571 3572 buffer->frameCount = framesReq; 3573 buffer->raw = (void *)cblk->buffer(u); 3574 return NO_ERROR; 3575} 3576 3577 3578void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3579{ 3580 size_t size = mBufferQueue.size(); 3581 Buffer *pBuffer; 3582 3583 for (size_t i = 0; i < size; i++) { 3584 pBuffer = mBufferQueue.itemAt(i); 3585 delete [] pBuffer->mBuffer; 3586 delete pBuffer; 3587 } 3588 mBufferQueue.clear(); 3589} 3590 3591// ---------------------------------------------------------------------------- 3592 3593AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3594 : RefBase(), 3595 mAudioFlinger(audioFlinger), 3596 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3597 mPid(pid) 3598{ 3599 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3600} 3601 3602// Client destructor must be called with AudioFlinger::mLock held 3603AudioFlinger::Client::~Client() 3604{ 3605 mAudioFlinger->removeClient_l(mPid); 3606} 3607 3608const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3609{ 3610 return mMemoryDealer; 3611} 3612 3613// ---------------------------------------------------------------------------- 3614 3615AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3616 const sp<IAudioFlingerClient>& client, 3617 pid_t pid) 3618 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3619{ 3620} 3621 3622AudioFlinger::NotificationClient::~NotificationClient() 3623{ 3624 mClient.clear(); 3625} 3626 3627void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3628{ 3629 sp<NotificationClient> keep(this); 3630 { 3631 mAudioFlinger->removeNotificationClient(mPid); 3632 } 3633} 3634 3635// ---------------------------------------------------------------------------- 3636 3637AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3638 : BnAudioTrack(), 3639 mTrack(track) 3640{ 3641} 3642 3643AudioFlinger::TrackHandle::~TrackHandle() { 3644 // just stop the track on deletion, associated resources 3645 // will be freed from the main thread once all pending buffers have 3646 // been played. Unless it's not in the active track list, in which 3647 // case we free everything now... 3648 mTrack->destroy(); 3649} 3650 3651status_t AudioFlinger::TrackHandle::start() { 3652 return mTrack->start(); 3653} 3654 3655void AudioFlinger::TrackHandle::stop() { 3656 mTrack->stop(); 3657} 3658 3659void AudioFlinger::TrackHandle::flush() { 3660 mTrack->flush(); 3661} 3662 3663void AudioFlinger::TrackHandle::mute(bool e) { 3664 mTrack->mute(e); 3665} 3666 3667void AudioFlinger::TrackHandle::pause() { 3668 mTrack->pause(); 3669} 3670 3671void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3672 mTrack->setVolume(left, right); 3673} 3674 3675sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3676 return mTrack->getCblk(); 3677} 3678 3679status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3680{ 3681 return mTrack->attachAuxEffect(EffectId); 3682} 3683 3684status_t AudioFlinger::TrackHandle::onTransact( 3685 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3686{ 3687 return BnAudioTrack::onTransact(code, data, reply, flags); 3688} 3689 3690// ---------------------------------------------------------------------------- 3691 3692sp<IAudioRecord> AudioFlinger::openRecord( 3693 pid_t pid, 3694 int input, 3695 uint32_t sampleRate, 3696 uint32_t format, 3697 uint32_t channelMask, 3698 int frameCount, 3699 uint32_t flags, 3700 int *sessionId, 3701 status_t *status) 3702{ 3703 sp<RecordThread::RecordTrack> recordTrack; 3704 sp<RecordHandle> recordHandle; 3705 sp<Client> client; 3706 wp<Client> wclient; 3707 status_t lStatus; 3708 RecordThread *thread; 3709 size_t inFrameCount; 3710 int lSessionId; 3711 3712 // check calling permissions 3713 if (!recordingAllowed()) { 3714 lStatus = PERMISSION_DENIED; 3715 goto Exit; 3716 } 3717 3718 // add client to list 3719 { // scope for mLock 3720 Mutex::Autolock _l(mLock); 3721 thread = checkRecordThread_l(input); 3722 if (thread == NULL) { 3723 lStatus = BAD_VALUE; 3724 goto Exit; 3725 } 3726 3727 wclient = mClients.valueFor(pid); 3728 if (wclient != NULL) { 3729 client = wclient.promote(); 3730 } else { 3731 client = new Client(this, pid); 3732 mClients.add(pid, client); 3733 } 3734 3735 // If no audio session id is provided, create one here 3736 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 3737 lSessionId = *sessionId; 3738 } else { 3739 lSessionId = nextUniqueId(); 3740 if (sessionId != NULL) { 3741 *sessionId = lSessionId; 3742 } 3743 } 3744 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3745 recordTrack = thread->createRecordTrack_l(client, 3746 sampleRate, 3747 format, 3748 channelMask, 3749 frameCount, 3750 flags, 3751 lSessionId, 3752 &lStatus); 3753 } 3754 if (lStatus != NO_ERROR) { 3755 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3756 // destructor is called by the TrackBase destructor with mLock held 3757 client.clear(); 3758 recordTrack.clear(); 3759 goto Exit; 3760 } 3761 3762 // return to handle to client 3763 recordHandle = new RecordHandle(recordTrack); 3764 lStatus = NO_ERROR; 3765 3766Exit: 3767 if (status) { 3768 *status = lStatus; 3769 } 3770 return recordHandle; 3771} 3772 3773// ---------------------------------------------------------------------------- 3774 3775AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3776 : BnAudioRecord(), 3777 mRecordTrack(recordTrack) 3778{ 3779} 3780 3781AudioFlinger::RecordHandle::~RecordHandle() { 3782 stop(); 3783} 3784 3785status_t AudioFlinger::RecordHandle::start() { 3786 LOGV("RecordHandle::start()"); 3787 return mRecordTrack->start(); 3788} 3789 3790void AudioFlinger::RecordHandle::stop() { 3791 LOGV("RecordHandle::stop()"); 3792 mRecordTrack->stop(); 3793} 3794 3795sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3796 return mRecordTrack->getCblk(); 3797} 3798 3799status_t AudioFlinger::RecordHandle::onTransact( 3800 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3801{ 3802 return BnAudioRecord::onTransact(code, data, reply, flags); 3803} 3804 3805// ---------------------------------------------------------------------------- 3806 3807AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 3808 AudioStreamIn *input, 3809 uint32_t sampleRate, 3810 uint32_t channels, 3811 int id, 3812 uint32_t device) : 3813 ThreadBase(audioFlinger, id, device), 3814 mInput(input), mTrack(NULL), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3815{ 3816 mType = ThreadBase::RECORD; 3817 mReqChannelCount = popcount(channels); 3818 mReqSampleRate = sampleRate; 3819 readInputParameters(); 3820} 3821 3822 3823AudioFlinger::RecordThread::~RecordThread() 3824{ 3825 delete[] mRsmpInBuffer; 3826 if (mResampler != 0) { 3827 delete mResampler; 3828 delete[] mRsmpOutBuffer; 3829 } 3830} 3831 3832void AudioFlinger::RecordThread::onFirstRef() 3833{ 3834 const size_t SIZE = 256; 3835 char buffer[SIZE]; 3836 3837 snprintf(buffer, SIZE, "Record Thread %p", this); 3838 3839 run(buffer, PRIORITY_URGENT_AUDIO); 3840} 3841 3842bool AudioFlinger::RecordThread::threadLoop() 3843{ 3844 AudioBufferProvider::Buffer buffer; 3845 sp<RecordTrack> activeTrack; 3846 Vector< sp<EffectChain> > effectChains; 3847 3848 nsecs_t lastWarning = 0; 3849 3850 // start recording 3851 while (!exitPending()) { 3852 3853 processConfigEvents(); 3854 3855 { // scope for mLock 3856 Mutex::Autolock _l(mLock); 3857 checkForNewParameters_l(); 3858 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3859 if (!mStandby) { 3860 mInput->stream->common.standby(&mInput->stream->common); 3861 mStandby = true; 3862 } 3863 3864 if (exitPending()) break; 3865 3866 LOGV("RecordThread: loop stopping"); 3867 // go to sleep 3868 mWaitWorkCV.wait(mLock); 3869 LOGV("RecordThread: loop starting"); 3870 continue; 3871 } 3872 if (mActiveTrack != 0) { 3873 if (mActiveTrack->mState == TrackBase::PAUSING) { 3874 if (!mStandby) { 3875 mInput->stream->common.standby(&mInput->stream->common); 3876 mStandby = true; 3877 } 3878 mActiveTrack.clear(); 3879 mStartStopCond.broadcast(); 3880 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3881 if (mReqChannelCount != mActiveTrack->channelCount()) { 3882 mActiveTrack.clear(); 3883 mStartStopCond.broadcast(); 3884 } else if (mBytesRead != 0) { 3885 // record start succeeds only if first read from audio input 3886 // succeeds 3887 if (mBytesRead > 0) { 3888 mActiveTrack->mState = TrackBase::ACTIVE; 3889 } else { 3890 mActiveTrack.clear(); 3891 } 3892 mStartStopCond.broadcast(); 3893 } 3894 mStandby = false; 3895 } 3896 } 3897 lockEffectChains_l(effectChains); 3898 } 3899 3900 if (mActiveTrack != 0) { 3901 if (mActiveTrack->mState != TrackBase::ACTIVE && 3902 mActiveTrack->mState != TrackBase::RESUMING) { 3903 unlockEffectChains(effectChains); 3904 usleep(kRecordThreadSleepUs); 3905 continue; 3906 } 3907 for (size_t i = 0; i < effectChains.size(); i ++) { 3908 effectChains[i]->process_l(); 3909 } 3910 // enable changes in effect chain 3911 unlockEffectChains(effectChains); 3912 3913 buffer.frameCount = mFrameCount; 3914 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3915 size_t framesOut = buffer.frameCount; 3916 if (mResampler == 0) { 3917 // no resampling 3918 while (framesOut) { 3919 size_t framesIn = mFrameCount - mRsmpInIndex; 3920 if (framesIn) { 3921 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3922 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3923 if (framesIn > framesOut) 3924 framesIn = framesOut; 3925 mRsmpInIndex += framesIn; 3926 framesOut -= framesIn; 3927 if ((int)mChannelCount == mReqChannelCount || 3928 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3929 memcpy(dst, src, framesIn * mFrameSize); 3930 } else { 3931 int16_t *src16 = (int16_t *)src; 3932 int16_t *dst16 = (int16_t *)dst; 3933 if (mChannelCount == 1) { 3934 while (framesIn--) { 3935 *dst16++ = *src16; 3936 *dst16++ = *src16++; 3937 } 3938 } else { 3939 while (framesIn--) { 3940 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3941 src16 += 2; 3942 } 3943 } 3944 } 3945 } 3946 if (framesOut && mFrameCount == mRsmpInIndex) { 3947 if (framesOut == mFrameCount && 3948 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3949 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 3950 framesOut = 0; 3951 } else { 3952 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 3953 mRsmpInIndex = 0; 3954 } 3955 if (mBytesRead < 0) { 3956 LOGE("Error reading audio input"); 3957 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3958 // Force input into standby so that it tries to 3959 // recover at next read attempt 3960 mInput->stream->common.standby(&mInput->stream->common); 3961 usleep(kRecordThreadSleepUs); 3962 } 3963 mRsmpInIndex = mFrameCount; 3964 framesOut = 0; 3965 buffer.frameCount = 0; 3966 } 3967 } 3968 } 3969 } else { 3970 // resampling 3971 3972 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3973 // alter output frame count as if we were expecting stereo samples 3974 if (mChannelCount == 1 && mReqChannelCount == 1) { 3975 framesOut >>= 1; 3976 } 3977 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3978 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3979 // are 32 bit aligned which should be always true. 3980 if (mChannelCount == 2 && mReqChannelCount == 1) { 3981 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3982 // the resampler always outputs stereo samples: do post stereo to mono conversion 3983 int16_t *src = (int16_t *)mRsmpOutBuffer; 3984 int16_t *dst = buffer.i16; 3985 while (framesOut--) { 3986 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3987 src += 2; 3988 } 3989 } else { 3990 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3991 } 3992 3993 } 3994 mActiveTrack->releaseBuffer(&buffer); 3995 mActiveTrack->overflow(); 3996 } 3997 // client isn't retrieving buffers fast enough 3998 else { 3999 if (!mActiveTrack->setOverflow()) { 4000 nsecs_t now = systemTime(); 4001 if ((now - lastWarning) > kWarningThrottle) { 4002 LOGW("RecordThread: buffer overflow"); 4003 lastWarning = now; 4004 } 4005 } 4006 // Release the processor for a while before asking for a new buffer. 4007 // This will give the application more chance to read from the buffer and 4008 // clear the overflow. 4009 usleep(kRecordThreadSleepUs); 4010 } 4011 } else { 4012 unlockEffectChains(effectChains); 4013 } 4014 effectChains.clear(); 4015 } 4016 4017 if (!mStandby) { 4018 mInput->stream->common.standby(&mInput->stream->common); 4019 } 4020 mActiveTrack.clear(); 4021 4022 mStartStopCond.broadcast(); 4023 4024 LOGV("RecordThread %p exiting", this); 4025 return false; 4026} 4027 4028 4029sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4030 const sp<AudioFlinger::Client>& client, 4031 uint32_t sampleRate, 4032 int format, 4033 int channelMask, 4034 int frameCount, 4035 uint32_t flags, 4036 int sessionId, 4037 status_t *status) 4038{ 4039 sp<RecordTrack> track; 4040 status_t lStatus; 4041 4042 lStatus = initCheck(); 4043 if (lStatus != NO_ERROR) { 4044 LOGE("Audio driver not initialized."); 4045 goto Exit; 4046 } 4047 4048 { // scope for mLock 4049 Mutex::Autolock _l(mLock); 4050 4051 track = new RecordTrack(this, client, sampleRate, 4052 format, channelMask, frameCount, flags, sessionId); 4053 4054 if (track->getCblk() == NULL) { 4055 lStatus = NO_MEMORY; 4056 goto Exit; 4057 } 4058 4059 mTrack = track.get(); 4060 4061 } 4062 lStatus = NO_ERROR; 4063 4064Exit: 4065 if (status) { 4066 *status = lStatus; 4067 } 4068 return track; 4069} 4070 4071status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4072{ 4073 LOGV("RecordThread::start"); 4074 sp <ThreadBase> strongMe = this; 4075 status_t status = NO_ERROR; 4076 { 4077 AutoMutex lock(&mLock); 4078 if (mActiveTrack != 0) { 4079 if (recordTrack != mActiveTrack.get()) { 4080 status = -EBUSY; 4081 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4082 mActiveTrack->mState = TrackBase::ACTIVE; 4083 } 4084 return status; 4085 } 4086 4087 recordTrack->mState = TrackBase::IDLE; 4088 mActiveTrack = recordTrack; 4089 mLock.unlock(); 4090 status_t status = AudioSystem::startInput(mId); 4091 mLock.lock(); 4092 if (status != NO_ERROR) { 4093 mActiveTrack.clear(); 4094 return status; 4095 } 4096 mRsmpInIndex = mFrameCount; 4097 mBytesRead = 0; 4098 if (mResampler != NULL) { 4099 mResampler->reset(); 4100 } 4101 mActiveTrack->mState = TrackBase::RESUMING; 4102 // signal thread to start 4103 LOGV("Signal record thread"); 4104 mWaitWorkCV.signal(); 4105 // do not wait for mStartStopCond if exiting 4106 if (mExiting) { 4107 mActiveTrack.clear(); 4108 status = INVALID_OPERATION; 4109 goto startError; 4110 } 4111 mStartStopCond.wait(mLock); 4112 if (mActiveTrack == 0) { 4113 LOGV("Record failed to start"); 4114 status = BAD_VALUE; 4115 goto startError; 4116 } 4117 LOGV("Record started OK"); 4118 return status; 4119 } 4120startError: 4121 AudioSystem::stopInput(mId); 4122 return status; 4123} 4124 4125void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4126 LOGV("RecordThread::stop"); 4127 sp <ThreadBase> strongMe = this; 4128 { 4129 AutoMutex lock(&mLock); 4130 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4131 mActiveTrack->mState = TrackBase::PAUSING; 4132 // do not wait for mStartStopCond if exiting 4133 if (mExiting) { 4134 return; 4135 } 4136 mStartStopCond.wait(mLock); 4137 // if we have been restarted, recordTrack == mActiveTrack.get() here 4138 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4139 mLock.unlock(); 4140 AudioSystem::stopInput(mId); 4141 mLock.lock(); 4142 LOGV("Record stopped OK"); 4143 } 4144 } 4145 } 4146} 4147 4148status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4149{ 4150 const size_t SIZE = 256; 4151 char buffer[SIZE]; 4152 String8 result; 4153 pid_t pid = 0; 4154 4155 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4156 result.append(buffer); 4157 4158 if (mActiveTrack != 0) { 4159 result.append("Active Track:\n"); 4160 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4161 mActiveTrack->dump(buffer, SIZE); 4162 result.append(buffer); 4163 4164 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4165 result.append(buffer); 4166 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4167 result.append(buffer); 4168 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4169 result.append(buffer); 4170 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4171 result.append(buffer); 4172 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4173 result.append(buffer); 4174 4175 4176 } else { 4177 result.append("No record client\n"); 4178 } 4179 write(fd, result.string(), result.size()); 4180 4181 dumpBase(fd, args); 4182 dumpEffectChains(fd, args); 4183 4184 return NO_ERROR; 4185} 4186 4187status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4188{ 4189 size_t framesReq = buffer->frameCount; 4190 size_t framesReady = mFrameCount - mRsmpInIndex; 4191 int channelCount; 4192 4193 if (framesReady == 0) { 4194 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4195 if (mBytesRead < 0) { 4196 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4197 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4198 // Force input into standby so that it tries to 4199 // recover at next read attempt 4200 mInput->stream->common.standby(&mInput->stream->common); 4201 usleep(kRecordThreadSleepUs); 4202 } 4203 buffer->raw = 0; 4204 buffer->frameCount = 0; 4205 return NOT_ENOUGH_DATA; 4206 } 4207 mRsmpInIndex = 0; 4208 framesReady = mFrameCount; 4209 } 4210 4211 if (framesReq > framesReady) { 4212 framesReq = framesReady; 4213 } 4214 4215 if (mChannelCount == 1 && mReqChannelCount == 2) { 4216 channelCount = 1; 4217 } else { 4218 channelCount = 2; 4219 } 4220 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4221 buffer->frameCount = framesReq; 4222 return NO_ERROR; 4223} 4224 4225void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4226{ 4227 mRsmpInIndex += buffer->frameCount; 4228 buffer->frameCount = 0; 4229} 4230 4231bool AudioFlinger::RecordThread::checkForNewParameters_l() 4232{ 4233 bool reconfig = false; 4234 4235 while (!mNewParameters.isEmpty()) { 4236 status_t status = NO_ERROR; 4237 String8 keyValuePair = mNewParameters[0]; 4238 AudioParameter param = AudioParameter(keyValuePair); 4239 int value; 4240 int reqFormat = mFormat; 4241 int reqSamplingRate = mReqSampleRate; 4242 int reqChannelCount = mReqChannelCount; 4243 4244 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4245 reqSamplingRate = value; 4246 reconfig = true; 4247 } 4248 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4249 reqFormat = value; 4250 reconfig = true; 4251 } 4252 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4253 reqChannelCount = popcount(value); 4254 reconfig = true; 4255 } 4256 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4257 // do not accept frame count changes if tracks are open as the track buffer 4258 // size depends on frame count and correct behavior would not be garantied 4259 // if frame count is changed after track creation 4260 if (mActiveTrack != 0) { 4261 status = INVALID_OPERATION; 4262 } else { 4263 reconfig = true; 4264 } 4265 } 4266 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4267 // forward device change to effects that have requested to be 4268 // aware of attached audio device. 4269 for (size_t i = 0; i < mEffectChains.size(); i++) { 4270 mEffectChains[i]->setDevice_l(value); 4271 } 4272 // store input device and output device but do not forward output device to audio HAL. 4273 // Note that status is ignored by the caller for output device 4274 // (see AudioFlinger::setParameters() 4275 if (value & AUDIO_DEVICE_OUT_ALL) { 4276 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4277 status = BAD_VALUE; 4278 } else { 4279 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4280 } 4281 mDevice |= (uint32_t)value; 4282 } 4283 if (status == NO_ERROR) { 4284 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4285 if (status == INVALID_OPERATION) { 4286 mInput->stream->common.standby(&mInput->stream->common); 4287 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4288 } 4289 if (reconfig) { 4290 if (status == BAD_VALUE && 4291 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4292 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4293 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4294 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4295 (reqChannelCount < 3)) { 4296 status = NO_ERROR; 4297 } 4298 if (status == NO_ERROR) { 4299 readInputParameters(); 4300 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4301 } 4302 } 4303 } 4304 4305 mNewParameters.removeAt(0); 4306 4307 mParamStatus = status; 4308 mParamCond.signal(); 4309 mWaitWorkCV.wait(mLock); 4310 } 4311 return reconfig; 4312} 4313 4314String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4315{ 4316 char *s; 4317 String8 out_s8; 4318 4319 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4320 out_s8 = String8(s); 4321 free(s); 4322 return out_s8; 4323} 4324 4325void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4326 AudioSystem::OutputDescriptor desc; 4327 void *param2 = 0; 4328 4329 switch (event) { 4330 case AudioSystem::INPUT_OPENED: 4331 case AudioSystem::INPUT_CONFIG_CHANGED: 4332 desc.channels = mChannelMask; 4333 desc.samplingRate = mSampleRate; 4334 desc.format = mFormat; 4335 desc.frameCount = mFrameCount; 4336 desc.latency = 0; 4337 param2 = &desc; 4338 break; 4339 4340 case AudioSystem::INPUT_CLOSED: 4341 default: 4342 break; 4343 } 4344 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4345} 4346 4347void AudioFlinger::RecordThread::readInputParameters() 4348{ 4349 if (mRsmpInBuffer) delete mRsmpInBuffer; 4350 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4351 if (mResampler) delete mResampler; 4352 mResampler = 0; 4353 4354 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4355 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4356 mChannelCount = (uint16_t)popcount(mChannelMask); 4357 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4358 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); 4359 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4360 mFrameCount = mInputBytes / mFrameSize; 4361 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4362 4363 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4364 { 4365 int channelCount; 4366 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4367 // stereo to mono post process as the resampler always outputs stereo. 4368 if (mChannelCount == 1 && mReqChannelCount == 2) { 4369 channelCount = 1; 4370 } else { 4371 channelCount = 2; 4372 } 4373 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4374 mResampler->setSampleRate(mSampleRate); 4375 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4376 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4377 4378 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4379 if (mChannelCount == 1 && mReqChannelCount == 1) { 4380 mFrameCount >>= 1; 4381 } 4382 4383 } 4384 mRsmpInIndex = mFrameCount; 4385} 4386 4387unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4388{ 4389 return mInput->stream->get_input_frames_lost(mInput->stream); 4390} 4391 4392uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4393{ 4394 Mutex::Autolock _l(mLock); 4395 uint32_t result = 0; 4396 if (getEffectChain_l(sessionId) != 0) { 4397 result = EFFECT_SESSION; 4398 } 4399 4400 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4401 result |= TRACK_SESSION; 4402 } 4403 4404 return result; 4405} 4406 4407// ---------------------------------------------------------------------------- 4408 4409int AudioFlinger::openOutput(uint32_t *pDevices, 4410 uint32_t *pSamplingRate, 4411 uint32_t *pFormat, 4412 uint32_t *pChannels, 4413 uint32_t *pLatencyMs, 4414 uint32_t flags) 4415{ 4416 status_t status; 4417 PlaybackThread *thread = NULL; 4418 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4419 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4420 uint32_t format = pFormat ? *pFormat : 0; 4421 uint32_t channels = pChannels ? *pChannels : 0; 4422 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4423 audio_stream_out_t *outStream; 4424 audio_hw_device_t *outHwDev; 4425 4426 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4427 pDevices ? *pDevices : 0, 4428 samplingRate, 4429 format, 4430 channels, 4431 flags); 4432 4433 if (pDevices == NULL || *pDevices == 0) { 4434 return 0; 4435 } 4436 4437 Mutex::Autolock _l(mLock); 4438 4439 outHwDev = findSuitableHwDev_l(*pDevices); 4440 if (outHwDev == NULL) 4441 return 0; 4442 4443 status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, 4444 &channels, &samplingRate, &outStream); 4445 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4446 outStream, 4447 samplingRate, 4448 format, 4449 channels, 4450 status); 4451 4452 mHardwareStatus = AUDIO_HW_IDLE; 4453 if (outStream != NULL) { 4454 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4455 int id = nextUniqueId(); 4456 4457 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4458 (format != AUDIO_FORMAT_PCM_16_BIT) || 4459 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4460 thread = new DirectOutputThread(this, output, id, *pDevices); 4461 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4462 } else { 4463 thread = new MixerThread(this, output, id, *pDevices); 4464 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4465 } 4466 mPlaybackThreads.add(id, thread); 4467 4468 if (pSamplingRate) *pSamplingRate = samplingRate; 4469 if (pFormat) *pFormat = format; 4470 if (pChannels) *pChannels = channels; 4471 if (pLatencyMs) *pLatencyMs = thread->latency(); 4472 4473 // notify client processes of the new output creation 4474 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4475 return id; 4476 } 4477 4478 return 0; 4479} 4480 4481int AudioFlinger::openDuplicateOutput(int output1, int output2) 4482{ 4483 Mutex::Autolock _l(mLock); 4484 MixerThread *thread1 = checkMixerThread_l(output1); 4485 MixerThread *thread2 = checkMixerThread_l(output2); 4486 4487 if (thread1 == NULL || thread2 == NULL) { 4488 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4489 return 0; 4490 } 4491 4492 int id = nextUniqueId(); 4493 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4494 thread->addOutputTrack(thread2); 4495 mPlaybackThreads.add(id, thread); 4496 // notify client processes of the new output creation 4497 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4498 return id; 4499} 4500 4501status_t AudioFlinger::closeOutput(int output) 4502{ 4503 // keep strong reference on the playback thread so that 4504 // it is not destroyed while exit() is executed 4505 sp <PlaybackThread> thread; 4506 { 4507 Mutex::Autolock _l(mLock); 4508 thread = checkPlaybackThread_l(output); 4509 if (thread == NULL) { 4510 return BAD_VALUE; 4511 } 4512 4513 LOGV("closeOutput() %d", output); 4514 4515 if (thread->type() == ThreadBase::MIXER) { 4516 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4517 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 4518 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4519 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4520 } 4521 } 4522 } 4523 void *param2 = 0; 4524 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4525 mPlaybackThreads.removeItem(output); 4526 } 4527 thread->exit(); 4528 4529 if (thread->type() != ThreadBase::DUPLICATING) { 4530 AudioStreamOut *out = thread->getOutput(); 4531 out->hwDev->close_output_stream(out->hwDev, out->stream); 4532 delete out; 4533 } 4534 return NO_ERROR; 4535} 4536 4537status_t AudioFlinger::suspendOutput(int output) 4538{ 4539 Mutex::Autolock _l(mLock); 4540 PlaybackThread *thread = checkPlaybackThread_l(output); 4541 4542 if (thread == NULL) { 4543 return BAD_VALUE; 4544 } 4545 4546 LOGV("suspendOutput() %d", output); 4547 thread->suspend(); 4548 4549 return NO_ERROR; 4550} 4551 4552status_t AudioFlinger::restoreOutput(int output) 4553{ 4554 Mutex::Autolock _l(mLock); 4555 PlaybackThread *thread = checkPlaybackThread_l(output); 4556 4557 if (thread == NULL) { 4558 return BAD_VALUE; 4559 } 4560 4561 LOGV("restoreOutput() %d", output); 4562 4563 thread->restore(); 4564 4565 return NO_ERROR; 4566} 4567 4568int AudioFlinger::openInput(uint32_t *pDevices, 4569 uint32_t *pSamplingRate, 4570 uint32_t *pFormat, 4571 uint32_t *pChannels, 4572 uint32_t acoustics) 4573{ 4574 status_t status; 4575 RecordThread *thread = NULL; 4576 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4577 uint32_t format = pFormat ? *pFormat : 0; 4578 uint32_t channels = pChannels ? *pChannels : 0; 4579 uint32_t reqSamplingRate = samplingRate; 4580 uint32_t reqFormat = format; 4581 uint32_t reqChannels = channels; 4582 audio_stream_in_t *inStream; 4583 audio_hw_device_t *inHwDev; 4584 4585 if (pDevices == NULL || *pDevices == 0) { 4586 return 0; 4587 } 4588 4589 Mutex::Autolock _l(mLock); 4590 4591 inHwDev = findSuitableHwDev_l(*pDevices); 4592 if (inHwDev == NULL) 4593 return 0; 4594 4595 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 4596 &channels, &samplingRate, 4597 (audio_in_acoustics_t)acoustics, 4598 &inStream); 4599 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4600 inStream, 4601 samplingRate, 4602 format, 4603 channels, 4604 acoustics, 4605 status); 4606 4607 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4608 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4609 // or stereo to mono conversions on 16 bit PCM inputs. 4610 if (inStream == NULL && status == BAD_VALUE && 4611 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 4612 (samplingRate <= 2 * reqSamplingRate) && 4613 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 4614 LOGV("openInput() reopening with proposed sampling rate and channels"); 4615 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 4616 &channels, &samplingRate, 4617 (audio_in_acoustics_t)acoustics, 4618 &inStream); 4619 } 4620 4621 if (inStream != NULL) { 4622 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 4623 4624 int id = nextUniqueId(); 4625 // Start record thread 4626 // RecorThread require both input and output device indication to forward to audio 4627 // pre processing modules 4628 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 4629 thread = new RecordThread(this, 4630 input, 4631 reqSamplingRate, 4632 reqChannels, 4633 id, 4634 device); 4635 mRecordThreads.add(id, thread); 4636 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4637 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4638 if (pFormat) *pFormat = format; 4639 if (pChannels) *pChannels = reqChannels; 4640 4641 input->stream->common.standby(&input->stream->common); 4642 4643 // notify client processes of the new input creation 4644 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4645 return id; 4646 } 4647 4648 return 0; 4649} 4650 4651status_t AudioFlinger::closeInput(int input) 4652{ 4653 // keep strong reference on the record thread so that 4654 // it is not destroyed while exit() is executed 4655 sp <RecordThread> thread; 4656 { 4657 Mutex::Autolock _l(mLock); 4658 thread = checkRecordThread_l(input); 4659 if (thread == NULL) { 4660 return BAD_VALUE; 4661 } 4662 4663 LOGV("closeInput() %d", input); 4664 void *param2 = 0; 4665 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4666 mRecordThreads.removeItem(input); 4667 } 4668 thread->exit(); 4669 4670 AudioStreamIn *in = thread->getInput(); 4671 in->hwDev->close_input_stream(in->hwDev, in->stream); 4672 delete in; 4673 4674 return NO_ERROR; 4675} 4676 4677status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4678{ 4679 Mutex::Autolock _l(mLock); 4680 MixerThread *dstThread = checkMixerThread_l(output); 4681 if (dstThread == NULL) { 4682 LOGW("setStreamOutput() bad output id %d", output); 4683 return BAD_VALUE; 4684 } 4685 4686 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4687 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4688 4689 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4690 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4691 if (thread != dstThread && 4692 thread->type() != ThreadBase::DIRECT) { 4693 MixerThread *srcThread = (MixerThread *)thread; 4694 srcThread->invalidateTracks(stream); 4695 } 4696 } 4697 4698 return NO_ERROR; 4699} 4700 4701 4702int AudioFlinger::newAudioSessionId() 4703{ 4704 return nextUniqueId(); 4705} 4706 4707// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4708AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4709{ 4710 PlaybackThread *thread = NULL; 4711 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4712 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4713 } 4714 return thread; 4715} 4716 4717// checkMixerThread_l() must be called with AudioFlinger::mLock held 4718AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4719{ 4720 PlaybackThread *thread = checkPlaybackThread_l(output); 4721 if (thread != NULL) { 4722 if (thread->type() == ThreadBase::DIRECT) { 4723 thread = NULL; 4724 } 4725 } 4726 return (MixerThread *)thread; 4727} 4728 4729// checkRecordThread_l() must be called with AudioFlinger::mLock held 4730AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4731{ 4732 RecordThread *thread = NULL; 4733 if (mRecordThreads.indexOfKey(input) >= 0) { 4734 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4735 } 4736 return thread; 4737} 4738 4739uint32_t AudioFlinger::nextUniqueId() 4740{ 4741 return android_atomic_inc(&mNextUniqueId); 4742} 4743 4744AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 4745{ 4746 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4747 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4748 if (thread->getOutput()->hwDev == mPrimaryHardwareDev) { 4749 return thread; 4750 } 4751 } 4752 return NULL; 4753} 4754 4755uint32_t AudioFlinger::primaryOutputDevice_l() 4756{ 4757 PlaybackThread *thread = primaryPlaybackThread_l(); 4758 4759 if (thread == NULL) { 4760 return 0; 4761 } 4762 4763 return thread->device(); 4764} 4765 4766 4767// ---------------------------------------------------------------------------- 4768// Effect management 4769// ---------------------------------------------------------------------------- 4770 4771 4772status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4773{ 4774 Mutex::Autolock _l(mLock); 4775 return EffectQueryNumberEffects(numEffects); 4776} 4777 4778status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4779{ 4780 Mutex::Autolock _l(mLock); 4781 return EffectQueryEffect(index, descriptor); 4782} 4783 4784status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4785{ 4786 Mutex::Autolock _l(mLock); 4787 return EffectGetDescriptor(pUuid, descriptor); 4788} 4789 4790 4791// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4792static const effect_uuid_t VISUALIZATION_UUID_ = 4793 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4794 4795sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4796 effect_descriptor_t *pDesc, 4797 const sp<IEffectClient>& effectClient, 4798 int32_t priority, 4799 int io, 4800 int sessionId, 4801 status_t *status, 4802 int *id, 4803 int *enabled) 4804{ 4805 status_t lStatus = NO_ERROR; 4806 sp<EffectHandle> handle; 4807 effect_descriptor_t desc; 4808 sp<Client> client; 4809 wp<Client> wclient; 4810 4811 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 4812 pid, effectClient.get(), priority, sessionId, io); 4813 4814 if (pDesc == NULL) { 4815 lStatus = BAD_VALUE; 4816 goto Exit; 4817 } 4818 4819 // check audio settings permission for global effects 4820 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 4821 lStatus = PERMISSION_DENIED; 4822 goto Exit; 4823 } 4824 4825 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 4826 // that can only be created by audio policy manager (running in same process) 4827 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 4828 lStatus = PERMISSION_DENIED; 4829 goto Exit; 4830 } 4831 4832 // check recording permission for visualizer 4833 if ((memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4834 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) && 4835 !recordingAllowed()) { 4836 lStatus = PERMISSION_DENIED; 4837 goto Exit; 4838 } 4839 4840 if (io == 0) { 4841 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 4842 // output must be specified by AudioPolicyManager when using session 4843 // AUDIO_SESSION_OUTPUT_STAGE 4844 lStatus = BAD_VALUE; 4845 goto Exit; 4846 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 4847 // if the output returned by getOutputForEffect() is removed before we lock the 4848 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 4849 // and we will exit safely 4850 io = AudioSystem::getOutputForEffect(&desc); 4851 } 4852 } 4853 4854 { 4855 Mutex::Autolock _l(mLock); 4856 4857 4858 if (!EffectIsNullUuid(&pDesc->uuid)) { 4859 // if uuid is specified, request effect descriptor 4860 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4861 if (lStatus < 0) { 4862 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4863 goto Exit; 4864 } 4865 } else { 4866 // if uuid is not specified, look for an available implementation 4867 // of the required type in effect factory 4868 if (EffectIsNullUuid(&pDesc->type)) { 4869 LOGW("createEffect() no effect type"); 4870 lStatus = BAD_VALUE; 4871 goto Exit; 4872 } 4873 uint32_t numEffects = 0; 4874 effect_descriptor_t d; 4875 bool found = false; 4876 4877 lStatus = EffectQueryNumberEffects(&numEffects); 4878 if (lStatus < 0) { 4879 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4880 goto Exit; 4881 } 4882 for (uint32_t i = 0; i < numEffects; i++) { 4883 lStatus = EffectQueryEffect(i, &desc); 4884 if (lStatus < 0) { 4885 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4886 continue; 4887 } 4888 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4889 // If matching type found save effect descriptor. If the session is 4890 // 0 and the effect is not auxiliary, continue enumeration in case 4891 // an auxiliary version of this effect type is available 4892 found = true; 4893 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4894 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 4895 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4896 break; 4897 } 4898 } 4899 } 4900 if (!found) { 4901 lStatus = BAD_VALUE; 4902 LOGW("createEffect() effect not found"); 4903 goto Exit; 4904 } 4905 // For same effect type, chose auxiliary version over insert version if 4906 // connect to output mix (Compliance to OpenSL ES) 4907 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 4908 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4909 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4910 } 4911 } 4912 4913 // Do not allow auxiliary effects on a session different from 0 (output mix) 4914 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 4915 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4916 lStatus = INVALID_OPERATION; 4917 goto Exit; 4918 } 4919 4920 // return effect descriptor 4921 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4922 4923 // If output is not specified try to find a matching audio session ID in one of the 4924 // output threads. 4925 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 4926 // because of code checking output when entering the function. 4927 // Note: io is never 0 when creating an effect on an input 4928 if (io == 0) { 4929 // look for the thread where the specified audio session is present 4930 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4931 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4932 io = mPlaybackThreads.keyAt(i); 4933 break; 4934 } 4935 } 4936 if (io == 0) { 4937 for (size_t i = 0; i < mRecordThreads.size(); i++) { 4938 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4939 io = mRecordThreads.keyAt(i); 4940 break; 4941 } 4942 } 4943 } 4944 // If no output thread contains the requested session ID, default to 4945 // first output. The effect chain will be moved to the correct output 4946 // thread when a track with the same session ID is created 4947 if (io == 0 && mPlaybackThreads.size()) { 4948 io = mPlaybackThreads.keyAt(0); 4949 } 4950 LOGV("createEffect() got io %d for effect %s", io, desc.name); 4951 } 4952 ThreadBase *thread = checkRecordThread_l(io); 4953 if (thread == NULL) { 4954 thread = checkPlaybackThread_l(io); 4955 if (thread == NULL) { 4956 LOGE("createEffect() unknown output thread"); 4957 lStatus = BAD_VALUE; 4958 goto Exit; 4959 } 4960 } 4961 4962 wclient = mClients.valueFor(pid); 4963 4964 if (wclient != NULL) { 4965 client = wclient.promote(); 4966 } else { 4967 client = new Client(this, pid); 4968 mClients.add(pid, client); 4969 } 4970 4971 // create effect on selected output trhead 4972 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4973 &desc, enabled, &lStatus); 4974 if (handle != 0 && id != NULL) { 4975 *id = handle->id(); 4976 } 4977 } 4978 4979Exit: 4980 if(status) { 4981 *status = lStatus; 4982 } 4983 return handle; 4984} 4985 4986status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4987{ 4988 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4989 session, srcOutput, dstOutput); 4990 Mutex::Autolock _l(mLock); 4991 if (srcOutput == dstOutput) { 4992 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4993 return NO_ERROR; 4994 } 4995 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4996 if (srcThread == NULL) { 4997 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4998 return BAD_VALUE; 4999 } 5000 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5001 if (dstThread == NULL) { 5002 LOGW("moveEffects() bad dstOutput %d", dstOutput); 5003 return BAD_VALUE; 5004 } 5005 5006 Mutex::Autolock _dl(dstThread->mLock); 5007 Mutex::Autolock _sl(srcThread->mLock); 5008 moveEffectChain_l(session, srcThread, dstThread, false); 5009 5010 return NO_ERROR; 5011} 5012 5013// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 5014status_t AudioFlinger::moveEffectChain_l(int session, 5015 AudioFlinger::PlaybackThread *srcThread, 5016 AudioFlinger::PlaybackThread *dstThread, 5017 bool reRegister) 5018{ 5019 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5020 session, srcThread, dstThread); 5021 5022 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 5023 if (chain == 0) { 5024 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5025 session, srcThread); 5026 return INVALID_OPERATION; 5027 } 5028 5029 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5030 // so that a new chain is created with correct parameters when first effect is added. This is 5031 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 5032 // removed. 5033 srcThread->removeEffectChain_l(chain); 5034 5035 // transfer all effects one by one so that new effect chain is created on new thread with 5036 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5037 int dstOutput = dstThread->id(); 5038 sp<EffectChain> dstChain; 5039 uint32_t strategy; 5040 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5041 while (effect != 0) { 5042 srcThread->removeEffect_l(effect); 5043 dstThread->addEffect_l(effect); 5044 // if the move request is not received from audio policy manager, the effect must be 5045 // re-registered with the new strategy and output 5046 if (dstChain == 0) { 5047 dstChain = effect->chain().promote(); 5048 if (dstChain == 0) { 5049 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5050 srcThread->addEffect_l(effect); 5051 return NO_INIT; 5052 } 5053 strategy = dstChain->strategy(); 5054 } 5055 if (reRegister) { 5056 AudioSystem::unregisterEffect(effect->id()); 5057 AudioSystem::registerEffect(&effect->desc(), 5058 dstOutput, 5059 strategy, 5060 session, 5061 effect->id()); 5062 } 5063 effect = chain->getEffectFromId_l(0); 5064 } 5065 5066 return NO_ERROR; 5067} 5068 5069 5070// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5071sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5072 const sp<AudioFlinger::Client>& client, 5073 const sp<IEffectClient>& effectClient, 5074 int32_t priority, 5075 int sessionId, 5076 effect_descriptor_t *desc, 5077 int *enabled, 5078 status_t *status 5079 ) 5080{ 5081 sp<EffectModule> effect; 5082 sp<EffectHandle> handle; 5083 status_t lStatus; 5084 sp<EffectChain> chain; 5085 bool chainCreated = false; 5086 bool effectCreated = false; 5087 bool effectRegistered = false; 5088 5089 lStatus = initCheck(); 5090 if (lStatus != NO_ERROR) { 5091 LOGW("createEffect_l() Audio driver not initialized."); 5092 goto Exit; 5093 } 5094 5095 // Do not allow effects with session ID 0 on direct output or duplicating threads 5096 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5097 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5098 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5099 desc->name, sessionId); 5100 lStatus = BAD_VALUE; 5101 goto Exit; 5102 } 5103 // Only Pre processor effects are allowed on input threads and only on input threads 5104 if ((mType == RECORD && 5105 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5106 (mType != RECORD && 5107 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5108 LOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5109 desc->name, desc->flags, mType); 5110 lStatus = BAD_VALUE; 5111 goto Exit; 5112 } 5113 5114 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5115 5116 { // scope for mLock 5117 Mutex::Autolock _l(mLock); 5118 5119 // check for existing effect chain with the requested audio session 5120 chain = getEffectChain_l(sessionId); 5121 if (chain == 0) { 5122 // create a new chain for this session 5123 LOGV("createEffect_l() new effect chain for session %d", sessionId); 5124 chain = new EffectChain(this, sessionId); 5125 addEffectChain_l(chain); 5126 chain->setStrategy(getStrategyForSession_l(sessionId)); 5127 chainCreated = true; 5128 } else { 5129 effect = chain->getEffectFromDesc_l(desc); 5130 } 5131 5132 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 5133 5134 if (effect == 0) { 5135 int id = mAudioFlinger->nextUniqueId(); 5136 // Check CPU and memory usage 5137 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5138 if (lStatus != NO_ERROR) { 5139 goto Exit; 5140 } 5141 effectRegistered = true; 5142 // create a new effect module if none present in the chain 5143 effect = new EffectModule(this, chain, desc, id, sessionId); 5144 lStatus = effect->status(); 5145 if (lStatus != NO_ERROR) { 5146 goto Exit; 5147 } 5148 lStatus = chain->addEffect_l(effect); 5149 if (lStatus != NO_ERROR) { 5150 goto Exit; 5151 } 5152 effectCreated = true; 5153 5154 effect->setDevice(mDevice); 5155 effect->setMode(mAudioFlinger->getMode()); 5156 } 5157 // create effect handle and connect it to effect module 5158 handle = new EffectHandle(effect, client, effectClient, priority); 5159 lStatus = effect->addHandle(handle); 5160 if (enabled) { 5161 *enabled = (int)effect->isEnabled(); 5162 } 5163 } 5164 5165Exit: 5166 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5167 Mutex::Autolock _l(mLock); 5168 if (effectCreated) { 5169 chain->removeEffect_l(effect); 5170 } 5171 if (effectRegistered) { 5172 AudioSystem::unregisterEffect(effect->id()); 5173 } 5174 if (chainCreated) { 5175 removeEffectChain_l(chain); 5176 } 5177 handle.clear(); 5178 } 5179 5180 if(status) { 5181 *status = lStatus; 5182 } 5183 return handle; 5184} 5185 5186sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5187{ 5188 sp<EffectModule> effect; 5189 5190 sp<EffectChain> chain = getEffectChain_l(sessionId); 5191 if (chain != 0) { 5192 effect = chain->getEffectFromId_l(effectId); 5193 } 5194 return effect; 5195} 5196 5197// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5198// PlaybackThread::mLock held 5199status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5200{ 5201 // check for existing effect chain with the requested audio session 5202 int sessionId = effect->sessionId(); 5203 sp<EffectChain> chain = getEffectChain_l(sessionId); 5204 bool chainCreated = false; 5205 5206 if (chain == 0) { 5207 // create a new chain for this session 5208 LOGV("addEffect_l() new effect chain for session %d", sessionId); 5209 chain = new EffectChain(this, sessionId); 5210 addEffectChain_l(chain); 5211 chain->setStrategy(getStrategyForSession_l(sessionId)); 5212 chainCreated = true; 5213 } 5214 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5215 5216 if (chain->getEffectFromId_l(effect->id()) != 0) { 5217 LOGW("addEffect_l() %p effect %s already present in chain %p", 5218 this, effect->desc().name, chain.get()); 5219 return BAD_VALUE; 5220 } 5221 5222 status_t status = chain->addEffect_l(effect); 5223 if (status != NO_ERROR) { 5224 if (chainCreated) { 5225 removeEffectChain_l(chain); 5226 } 5227 return status; 5228 } 5229 5230 effect->setDevice(mDevice); 5231 effect->setMode(mAudioFlinger->getMode()); 5232 return NO_ERROR; 5233} 5234 5235void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5236 5237 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5238 effect_descriptor_t desc = effect->desc(); 5239 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5240 detachAuxEffect_l(effect->id()); 5241 } 5242 5243 sp<EffectChain> chain = effect->chain().promote(); 5244 if (chain != 0) { 5245 // remove effect chain if removing last effect 5246 if (chain->removeEffect_l(effect) == 0) { 5247 removeEffectChain_l(chain); 5248 } 5249 } else { 5250 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5251 } 5252} 5253 5254void AudioFlinger::ThreadBase::lockEffectChains_l( 5255 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5256{ 5257 effectChains = mEffectChains; 5258 for (size_t i = 0; i < mEffectChains.size(); i++) { 5259 mEffectChains[i]->lock(); 5260 } 5261} 5262 5263void AudioFlinger::ThreadBase::unlockEffectChains( 5264 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5265{ 5266 for (size_t i = 0; i < effectChains.size(); i++) { 5267 effectChains[i]->unlock(); 5268 } 5269} 5270 5271sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5272{ 5273 Mutex::Autolock _l(mLock); 5274 return getEffectChain_l(sessionId); 5275} 5276 5277sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5278{ 5279 sp<EffectChain> chain; 5280 5281 size_t size = mEffectChains.size(); 5282 for (size_t i = 0; i < size; i++) { 5283 if (mEffectChains[i]->sessionId() == sessionId) { 5284 chain = mEffectChains[i]; 5285 break; 5286 } 5287 } 5288 return chain; 5289} 5290 5291void AudioFlinger::ThreadBase::setMode(uint32_t mode) 5292{ 5293 Mutex::Autolock _l(mLock); 5294 size_t size = mEffectChains.size(); 5295 for (size_t i = 0; i < size; i++) { 5296 mEffectChains[i]->setMode_l(mode); 5297 } 5298} 5299 5300void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5301 const wp<EffectHandle>& handle) { 5302 Mutex::Autolock _l(mLock); 5303 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5304 // delete the effect module if removing last handle on it 5305 if (effect->removeHandle(handle) == 0) { 5306 removeEffect_l(effect); 5307 AudioSystem::unregisterEffect(effect->id()); 5308 } 5309} 5310 5311status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5312{ 5313 int session = chain->sessionId(); 5314 int16_t *buffer = mMixBuffer; 5315 bool ownsBuffer = false; 5316 5317 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5318 if (session > 0) { 5319 // Only one effect chain can be present in direct output thread and it uses 5320 // the mix buffer as input 5321 if (mType != DIRECT) { 5322 size_t numSamples = mFrameCount * mChannelCount; 5323 buffer = new int16_t[numSamples]; 5324 memset(buffer, 0, numSamples * sizeof(int16_t)); 5325 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5326 ownsBuffer = true; 5327 } 5328 5329 // Attach all tracks with same session ID to this chain. 5330 for (size_t i = 0; i < mTracks.size(); ++i) { 5331 sp<Track> track = mTracks[i]; 5332 if (session == track->sessionId()) { 5333 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5334 track->setMainBuffer(buffer); 5335 chain->incTrackCnt(); 5336 } 5337 } 5338 5339 // indicate all active tracks in the chain 5340 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5341 sp<Track> track = mActiveTracks[i].promote(); 5342 if (track == 0) continue; 5343 if (session == track->sessionId()) { 5344 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5345 chain->incActiveTrackCnt(); 5346 } 5347 } 5348 } 5349 5350 chain->setInBuffer(buffer, ownsBuffer); 5351 chain->setOutBuffer(mMixBuffer); 5352 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5353 // chains list in order to be processed last as it contains output stage effects 5354 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5355 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5356 // after track specific effects and before output stage 5357 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5358 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5359 // Effect chain for other sessions are inserted at beginning of effect 5360 // chains list to be processed before output mix effects. Relative order between other 5361 // sessions is not important 5362 size_t size = mEffectChains.size(); 5363 size_t i = 0; 5364 for (i = 0; i < size; i++) { 5365 if (mEffectChains[i]->sessionId() < session) break; 5366 } 5367 mEffectChains.insertAt(chain, i); 5368 5369 return NO_ERROR; 5370} 5371 5372size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5373{ 5374 int session = chain->sessionId(); 5375 5376 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5377 5378 for (size_t i = 0; i < mEffectChains.size(); i++) { 5379 if (chain == mEffectChains[i]) { 5380 mEffectChains.removeAt(i); 5381 // detach all active tracks from the chain 5382 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5383 sp<Track> track = mActiveTracks[i].promote(); 5384 if (track == 0) continue; 5385 if (session == track->sessionId()) { 5386 LOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5387 chain.get(), session); 5388 chain->decActiveTrackCnt(); 5389 } 5390 } 5391 5392 // detach all tracks with same session ID from this chain 5393 for (size_t i = 0; i < mTracks.size(); ++i) { 5394 sp<Track> track = mTracks[i]; 5395 if (session == track->sessionId()) { 5396 track->setMainBuffer(mMixBuffer); 5397 chain->decTrackCnt(); 5398 } 5399 } 5400 break; 5401 } 5402 } 5403 return mEffectChains.size(); 5404} 5405 5406status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5407 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5408{ 5409 Mutex::Autolock _l(mLock); 5410 return attachAuxEffect_l(track, EffectId); 5411} 5412 5413status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5414 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5415{ 5416 status_t status = NO_ERROR; 5417 5418 if (EffectId == 0) { 5419 track->setAuxBuffer(0, NULL); 5420 } else { 5421 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 5422 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 5423 if (effect != 0) { 5424 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5425 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5426 } else { 5427 status = INVALID_OPERATION; 5428 } 5429 } else { 5430 status = BAD_VALUE; 5431 } 5432 } 5433 return status; 5434} 5435 5436void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5437{ 5438 for (size_t i = 0; i < mTracks.size(); ++i) { 5439 sp<Track> track = mTracks[i]; 5440 if (track->auxEffectId() == effectId) { 5441 attachAuxEffect_l(track, 0); 5442 } 5443 } 5444} 5445 5446status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 5447{ 5448 // only one chain per input thread 5449 if (mEffectChains.size() != 0) { 5450 return INVALID_OPERATION; 5451 } 5452 LOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 5453 5454 chain->setInBuffer(NULL); 5455 chain->setOutBuffer(NULL); 5456 5457 mEffectChains.add(chain); 5458 5459 return NO_ERROR; 5460} 5461 5462size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 5463{ 5464 LOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 5465 LOGW_IF(mEffectChains.size() != 1, 5466 "removeEffectChain_l() %p invalid chain size %d on thread %p", 5467 chain.get(), mEffectChains.size(), this); 5468 if (mEffectChains.size() == 1) { 5469 mEffectChains.removeAt(0); 5470 } 5471 return 0; 5472} 5473 5474// ---------------------------------------------------------------------------- 5475// EffectModule implementation 5476// ---------------------------------------------------------------------------- 5477 5478#undef LOG_TAG 5479#define LOG_TAG "AudioFlinger::EffectModule" 5480 5481AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5482 const wp<AudioFlinger::EffectChain>& chain, 5483 effect_descriptor_t *desc, 5484 int id, 5485 int sessionId) 5486 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5487 mStatus(NO_INIT), mState(IDLE) 5488{ 5489 LOGV("Constructor %p", this); 5490 int lStatus; 5491 sp<ThreadBase> thread = mThread.promote(); 5492 if (thread == 0) { 5493 return; 5494 } 5495 5496 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5497 5498 // create effect engine from effect factory 5499 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 5500 5501 if (mStatus != NO_ERROR) { 5502 return; 5503 } 5504 lStatus = init(); 5505 if (lStatus < 0) { 5506 mStatus = lStatus; 5507 goto Error; 5508 } 5509 5510 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5511 return; 5512Error: 5513 EffectRelease(mEffectInterface); 5514 mEffectInterface = NULL; 5515 LOGV("Constructor Error %d", mStatus); 5516} 5517 5518AudioFlinger::EffectModule::~EffectModule() 5519{ 5520 LOGV("Destructor %p", this); 5521 if (mEffectInterface != NULL) { 5522 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 5523 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 5524 sp<ThreadBase> thread = mThread.promote(); 5525 if (thread != 0) { 5526 thread->stream()->remove_audio_effect(thread->stream(), mEffectInterface); 5527 } 5528 } 5529 // release effect engine 5530 EffectRelease(mEffectInterface); 5531 } 5532} 5533 5534status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5535{ 5536 status_t status; 5537 5538 Mutex::Autolock _l(mLock); 5539 // First handle in mHandles has highest priority and controls the effect module 5540 int priority = handle->priority(); 5541 size_t size = mHandles.size(); 5542 sp<EffectHandle> h; 5543 size_t i; 5544 for (i = 0; i < size; i++) { 5545 h = mHandles[i].promote(); 5546 if (h == 0) continue; 5547 if (h->priority() <= priority) break; 5548 } 5549 // if inserted in first place, move effect control from previous owner to this handle 5550 if (i == 0) { 5551 if (h != 0) { 5552 h->setControl(false, true); 5553 } 5554 handle->setControl(true, false); 5555 status = NO_ERROR; 5556 } else { 5557 status = ALREADY_EXISTS; 5558 } 5559 mHandles.insertAt(handle, i); 5560 return status; 5561} 5562 5563size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5564{ 5565 Mutex::Autolock _l(mLock); 5566 size_t size = mHandles.size(); 5567 size_t i; 5568 for (i = 0; i < size; i++) { 5569 if (mHandles[i] == handle) break; 5570 } 5571 if (i == size) { 5572 return size; 5573 } 5574 mHandles.removeAt(i); 5575 size = mHandles.size(); 5576 // if removed from first place, move effect control from this handle to next in line 5577 if (i == 0 && size != 0) { 5578 sp<EffectHandle> h = mHandles[0].promote(); 5579 if (h != 0) { 5580 h->setControl(true, true); 5581 } 5582 } 5583 5584 // Release effect engine here so that it is done immediately. Otherwise it will be released 5585 // by the destructor when the last strong reference on the this object is released which can 5586 // happen after next process is called on this effect. 5587 if (size == 0 && mEffectInterface != NULL) { 5588 // release effect engine 5589 EffectRelease(mEffectInterface); 5590 mEffectInterface = NULL; 5591 } 5592 5593 return size; 5594} 5595 5596void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5597{ 5598 // keep a strong reference on this EffectModule to avoid calling the 5599 // destructor before we exit 5600 sp<EffectModule> keep(this); 5601 { 5602 sp<ThreadBase> thread = mThread.promote(); 5603 if (thread != 0) { 5604 thread->disconnectEffect(keep, handle); 5605 } 5606 } 5607} 5608 5609void AudioFlinger::EffectModule::updateState() { 5610 Mutex::Autolock _l(mLock); 5611 5612 switch (mState) { 5613 case RESTART: 5614 reset_l(); 5615 // FALL THROUGH 5616 5617 case STARTING: 5618 // clear auxiliary effect input buffer for next accumulation 5619 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5620 memset(mConfig.inputCfg.buffer.raw, 5621 0, 5622 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5623 } 5624 start_l(); 5625 mState = ACTIVE; 5626 break; 5627 case STOPPING: 5628 stop_l(); 5629 mDisableWaitCnt = mMaxDisableWaitCnt; 5630 mState = STOPPED; 5631 break; 5632 case STOPPED: 5633 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5634 // turn off sequence. 5635 if (--mDisableWaitCnt == 0) { 5636 reset_l(); 5637 mState = IDLE; 5638 } 5639 break; 5640 default: //IDLE , ACTIVE 5641 break; 5642 } 5643} 5644 5645void AudioFlinger::EffectModule::process() 5646{ 5647 Mutex::Autolock _l(mLock); 5648 5649 if (mEffectInterface == NULL || 5650 mConfig.inputCfg.buffer.raw == NULL || 5651 mConfig.outputCfg.buffer.raw == NULL) { 5652 return; 5653 } 5654 5655 if (isProcessEnabled()) { 5656 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5657 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5658 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5659 mConfig.inputCfg.buffer.s32, 5660 mConfig.inputCfg.buffer.frameCount/2); 5661 } 5662 5663 // do the actual processing in the effect engine 5664 int ret = (*mEffectInterface)->process(mEffectInterface, 5665 &mConfig.inputCfg.buffer, 5666 &mConfig.outputCfg.buffer); 5667 5668 // force transition to IDLE state when engine is ready 5669 if (mState == STOPPED && ret == -ENODATA) { 5670 mDisableWaitCnt = 1; 5671 } 5672 5673 // clear auxiliary effect input buffer for next accumulation 5674 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5675 memset(mConfig.inputCfg.buffer.raw, 0, 5676 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5677 } 5678 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5679 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5680 // If an insert effect is idle and input buffer is different from output buffer, 5681 // accumulate input onto output 5682 sp<EffectChain> chain = mChain.promote(); 5683 if (chain != 0 && chain->activeTrackCnt() != 0) { 5684 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 5685 int16_t *in = mConfig.inputCfg.buffer.s16; 5686 int16_t *out = mConfig.outputCfg.buffer.s16; 5687 for (size_t i = 0; i < frameCnt; i++) { 5688 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 5689 } 5690 } 5691 } 5692} 5693 5694void AudioFlinger::EffectModule::reset_l() 5695{ 5696 if (mEffectInterface == NULL) { 5697 return; 5698 } 5699 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5700} 5701 5702status_t AudioFlinger::EffectModule::configure() 5703{ 5704 uint32_t channels; 5705 if (mEffectInterface == NULL) { 5706 return NO_INIT; 5707 } 5708 5709 sp<ThreadBase> thread = mThread.promote(); 5710 if (thread == 0) { 5711 return DEAD_OBJECT; 5712 } 5713 5714 // TODO: handle configuration of effects replacing track process 5715 if (thread->channelCount() == 1) { 5716 channels = AUDIO_CHANNEL_OUT_MONO; 5717 } else { 5718 channels = AUDIO_CHANNEL_OUT_STEREO; 5719 } 5720 5721 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5722 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 5723 } else { 5724 mConfig.inputCfg.channels = channels; 5725 } 5726 mConfig.outputCfg.channels = channels; 5727 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 5728 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 5729 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5730 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5731 mConfig.inputCfg.bufferProvider.cookie = NULL; 5732 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5733 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5734 mConfig.outputCfg.bufferProvider.cookie = NULL; 5735 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5736 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5737 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5738 // Insert effect: 5739 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 5740 // always overwrites output buffer: input buffer == output buffer 5741 // - in other sessions: 5742 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5743 // other effect: overwrites output buffer: input buffer == output buffer 5744 // Auxiliary effect: 5745 // accumulates in output buffer: input buffer != output buffer 5746 // Therefore: accumulate <=> input buffer != output buffer 5747 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5748 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5749 } else { 5750 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5751 } 5752 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5753 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5754 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5755 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5756 5757 LOGV("configure() %p thread %p buffer %p framecount %d", 5758 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5759 5760 status_t cmdStatus; 5761 uint32_t size = sizeof(int); 5762 status_t status = (*mEffectInterface)->command(mEffectInterface, 5763 EFFECT_CMD_CONFIGURE, 5764 sizeof(effect_config_t), 5765 &mConfig, 5766 &size, 5767 &cmdStatus); 5768 if (status == 0) { 5769 status = cmdStatus; 5770 } 5771 5772 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5773 (1000 * mConfig.outputCfg.buffer.frameCount); 5774 5775 return status; 5776} 5777 5778status_t AudioFlinger::EffectModule::init() 5779{ 5780 Mutex::Autolock _l(mLock); 5781 if (mEffectInterface == NULL) { 5782 return NO_INIT; 5783 } 5784 status_t cmdStatus; 5785 uint32_t size = sizeof(status_t); 5786 status_t status = (*mEffectInterface)->command(mEffectInterface, 5787 EFFECT_CMD_INIT, 5788 0, 5789 NULL, 5790 &size, 5791 &cmdStatus); 5792 if (status == 0) { 5793 status = cmdStatus; 5794 } 5795 return status; 5796} 5797 5798status_t AudioFlinger::EffectModule::start_l() 5799{ 5800 if (mEffectInterface == NULL) { 5801 return NO_INIT; 5802 } 5803 status_t cmdStatus; 5804 uint32_t size = sizeof(status_t); 5805 status_t status = (*mEffectInterface)->command(mEffectInterface, 5806 EFFECT_CMD_ENABLE, 5807 0, 5808 NULL, 5809 &size, 5810 &cmdStatus); 5811 if (status == 0) { 5812 status = cmdStatus; 5813 } 5814 if (status == 0 && 5815 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 5816 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 5817 sp<ThreadBase> thread = mThread.promote(); 5818 if (thread != 0) { 5819 thread->stream()->add_audio_effect(thread->stream(), mEffectInterface); 5820 } 5821 } 5822 return status; 5823} 5824 5825status_t AudioFlinger::EffectModule::stop_l() 5826{ 5827 if (mEffectInterface == NULL) { 5828 return NO_INIT; 5829 } 5830 status_t cmdStatus; 5831 uint32_t size = sizeof(status_t); 5832 status_t status = (*mEffectInterface)->command(mEffectInterface, 5833 EFFECT_CMD_DISABLE, 5834 0, 5835 NULL, 5836 &size, 5837 &cmdStatus); 5838 if (status == 0) { 5839 status = cmdStatus; 5840 } 5841 if (status == 0 && 5842 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 5843 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 5844 sp<ThreadBase> thread = mThread.promote(); 5845 if (thread != 0) { 5846 thread->stream()->remove_audio_effect(thread->stream(), mEffectInterface); 5847 } 5848 } 5849 return status; 5850} 5851 5852status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5853 uint32_t cmdSize, 5854 void *pCmdData, 5855 uint32_t *replySize, 5856 void *pReplyData) 5857{ 5858 Mutex::Autolock _l(mLock); 5859// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5860 5861 if (mEffectInterface == NULL) { 5862 return NO_INIT; 5863 } 5864 status_t status = (*mEffectInterface)->command(mEffectInterface, 5865 cmdCode, 5866 cmdSize, 5867 pCmdData, 5868 replySize, 5869 pReplyData); 5870 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5871 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5872 for (size_t i = 1; i < mHandles.size(); i++) { 5873 sp<EffectHandle> h = mHandles[i].promote(); 5874 if (h != 0) { 5875 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5876 } 5877 } 5878 } 5879 return status; 5880} 5881 5882status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5883{ 5884 Mutex::Autolock _l(mLock); 5885 LOGV("setEnabled %p enabled %d", this, enabled); 5886 5887 if (enabled != isEnabled()) { 5888 switch (mState) { 5889 // going from disabled to enabled 5890 case IDLE: 5891 mState = STARTING; 5892 break; 5893 case STOPPED: 5894 mState = RESTART; 5895 break; 5896 case STOPPING: 5897 mState = ACTIVE; 5898 break; 5899 5900 // going from enabled to disabled 5901 case RESTART: 5902 mState = STOPPED; 5903 break; 5904 case STARTING: 5905 mState = IDLE; 5906 break; 5907 case ACTIVE: 5908 mState = STOPPING; 5909 break; 5910 } 5911 for (size_t i = 1; i < mHandles.size(); i++) { 5912 sp<EffectHandle> h = mHandles[i].promote(); 5913 if (h != 0) { 5914 h->setEnabled(enabled); 5915 } 5916 } 5917 } 5918 return NO_ERROR; 5919} 5920 5921bool AudioFlinger::EffectModule::isEnabled() 5922{ 5923 switch (mState) { 5924 case RESTART: 5925 case STARTING: 5926 case ACTIVE: 5927 return true; 5928 case IDLE: 5929 case STOPPING: 5930 case STOPPED: 5931 default: 5932 return false; 5933 } 5934} 5935 5936bool AudioFlinger::EffectModule::isProcessEnabled() 5937{ 5938 switch (mState) { 5939 case RESTART: 5940 case ACTIVE: 5941 case STOPPING: 5942 case STOPPED: 5943 return true; 5944 case IDLE: 5945 case STARTING: 5946 default: 5947 return false; 5948 } 5949} 5950 5951status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5952{ 5953 Mutex::Autolock _l(mLock); 5954 status_t status = NO_ERROR; 5955 5956 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5957 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5958 if (isProcessEnabled() && 5959 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5960 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5961 status_t cmdStatus; 5962 uint32_t volume[2]; 5963 uint32_t *pVolume = NULL; 5964 uint32_t size = sizeof(volume); 5965 volume[0] = *left; 5966 volume[1] = *right; 5967 if (controller) { 5968 pVolume = volume; 5969 } 5970 status = (*mEffectInterface)->command(mEffectInterface, 5971 EFFECT_CMD_SET_VOLUME, 5972 size, 5973 volume, 5974 &size, 5975 pVolume); 5976 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5977 *left = volume[0]; 5978 *right = volume[1]; 5979 } 5980 } 5981 return status; 5982} 5983 5984status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5985{ 5986 Mutex::Autolock _l(mLock); 5987 status_t status = NO_ERROR; 5988 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5989 // audio pre processing modules on RecordThread can receive both output and 5990 // input device indication in the same call 5991 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 5992 if (dev) { 5993 status_t cmdStatus; 5994 uint32_t size = sizeof(status_t); 5995 5996 status = (*mEffectInterface)->command(mEffectInterface, 5997 EFFECT_CMD_SET_DEVICE, 5998 sizeof(uint32_t), 5999 &dev, 6000 &size, 6001 &cmdStatus); 6002 if (status == NO_ERROR) { 6003 status = cmdStatus; 6004 } 6005 } 6006 dev = device & AUDIO_DEVICE_IN_ALL; 6007 if (dev) { 6008 status_t cmdStatus; 6009 uint32_t size = sizeof(status_t); 6010 6011 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6012 EFFECT_CMD_SET_INPUT_DEVICE, 6013 sizeof(uint32_t), 6014 &dev, 6015 &size, 6016 &cmdStatus); 6017 if (status2 == NO_ERROR) { 6018 status2 = cmdStatus; 6019 } 6020 if (status == NO_ERROR) { 6021 status = status2; 6022 } 6023 } 6024 } 6025 return status; 6026} 6027 6028status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 6029{ 6030 Mutex::Autolock _l(mLock); 6031 status_t status = NO_ERROR; 6032 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6033 status_t cmdStatus; 6034 uint32_t size = sizeof(status_t); 6035 status = (*mEffectInterface)->command(mEffectInterface, 6036 EFFECT_CMD_SET_AUDIO_MODE, 6037 sizeof(int), 6038 &mode, 6039 &size, 6040 &cmdStatus); 6041 if (status == NO_ERROR) { 6042 status = cmdStatus; 6043 } 6044 } 6045 return status; 6046} 6047 6048status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6049{ 6050 const size_t SIZE = 256; 6051 char buffer[SIZE]; 6052 String8 result; 6053 6054 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6055 result.append(buffer); 6056 6057 bool locked = tryLock(mLock); 6058 // failed to lock - AudioFlinger is probably deadlocked 6059 if (!locked) { 6060 result.append("\t\tCould not lock Fx mutex:\n"); 6061 } 6062 6063 result.append("\t\tSession Status State Engine:\n"); 6064 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6065 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6066 result.append(buffer); 6067 6068 result.append("\t\tDescriptor:\n"); 6069 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6070 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6071 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6072 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6073 result.append(buffer); 6074 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6075 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6076 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6077 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6078 result.append(buffer); 6079 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6080 mDescriptor.apiVersion, 6081 mDescriptor.flags); 6082 result.append(buffer); 6083 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6084 mDescriptor.name); 6085 result.append(buffer); 6086 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6087 mDescriptor.implementor); 6088 result.append(buffer); 6089 6090 result.append("\t\t- Input configuration:\n"); 6091 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6092 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6093 (uint32_t)mConfig.inputCfg.buffer.raw, 6094 mConfig.inputCfg.buffer.frameCount, 6095 mConfig.inputCfg.samplingRate, 6096 mConfig.inputCfg.channels, 6097 mConfig.inputCfg.format); 6098 result.append(buffer); 6099 6100 result.append("\t\t- Output configuration:\n"); 6101 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6102 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6103 (uint32_t)mConfig.outputCfg.buffer.raw, 6104 mConfig.outputCfg.buffer.frameCount, 6105 mConfig.outputCfg.samplingRate, 6106 mConfig.outputCfg.channels, 6107 mConfig.outputCfg.format); 6108 result.append(buffer); 6109 6110 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6111 result.append(buffer); 6112 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6113 for (size_t i = 0; i < mHandles.size(); ++i) { 6114 sp<EffectHandle> handle = mHandles[i].promote(); 6115 if (handle != 0) { 6116 handle->dump(buffer, SIZE); 6117 result.append(buffer); 6118 } 6119 } 6120 6121 result.append("\n"); 6122 6123 write(fd, result.string(), result.length()); 6124 6125 if (locked) { 6126 mLock.unlock(); 6127 } 6128 6129 return NO_ERROR; 6130} 6131 6132// ---------------------------------------------------------------------------- 6133// EffectHandle implementation 6134// ---------------------------------------------------------------------------- 6135 6136#undef LOG_TAG 6137#define LOG_TAG "AudioFlinger::EffectHandle" 6138 6139AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6140 const sp<AudioFlinger::Client>& client, 6141 const sp<IEffectClient>& effectClient, 6142 int32_t priority) 6143 : BnEffect(), 6144 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 6145{ 6146 LOGV("constructor %p", this); 6147 6148 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6149 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6150 if (mCblkMemory != 0) { 6151 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6152 6153 if (mCblk) { 6154 new(mCblk) effect_param_cblk_t(); 6155 mBuffer = (uint8_t *)mCblk + bufOffset; 6156 } 6157 } else { 6158 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6159 return; 6160 } 6161} 6162 6163AudioFlinger::EffectHandle::~EffectHandle() 6164{ 6165 LOGV("Destructor %p", this); 6166 disconnect(); 6167} 6168 6169status_t AudioFlinger::EffectHandle::enable() 6170{ 6171 if (!mHasControl) return INVALID_OPERATION; 6172 if (mEffect == 0) return DEAD_OBJECT; 6173 6174 return mEffect->setEnabled(true); 6175} 6176 6177status_t AudioFlinger::EffectHandle::disable() 6178{ 6179 if (!mHasControl) return INVALID_OPERATION; 6180 if (mEffect == NULL) return DEAD_OBJECT; 6181 6182 return mEffect->setEnabled(false); 6183} 6184 6185void AudioFlinger::EffectHandle::disconnect() 6186{ 6187 if (mEffect == 0) { 6188 return; 6189 } 6190 mEffect->disconnect(this); 6191 // release sp on module => module destructor can be called now 6192 mEffect.clear(); 6193 if (mCblk) { 6194 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6195 } 6196 mCblkMemory.clear(); // and free the shared memory 6197 if (mClient != 0) { 6198 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6199 mClient.clear(); 6200 } 6201} 6202 6203status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6204 uint32_t cmdSize, 6205 void *pCmdData, 6206 uint32_t *replySize, 6207 void *pReplyData) 6208{ 6209// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6210// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6211 6212 // only get parameter command is permitted for applications not controlling the effect 6213 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6214 return INVALID_OPERATION; 6215 } 6216 if (mEffect == 0) return DEAD_OBJECT; 6217 6218 // handle commands that are not forwarded transparently to effect engine 6219 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6220 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6221 // no risk to block the whole media server process or mixer threads is we are stuck here 6222 Mutex::Autolock _l(mCblk->lock); 6223 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6224 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6225 mCblk->serverIndex = 0; 6226 mCblk->clientIndex = 0; 6227 return BAD_VALUE; 6228 } 6229 status_t status = NO_ERROR; 6230 while (mCblk->serverIndex < mCblk->clientIndex) { 6231 int reply; 6232 uint32_t rsize = sizeof(int); 6233 int *p = (int *)(mBuffer + mCblk->serverIndex); 6234 int size = *p++; 6235 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6236 LOGW("command(): invalid parameter block size"); 6237 break; 6238 } 6239 effect_param_t *param = (effect_param_t *)p; 6240 if (param->psize == 0 || param->vsize == 0) { 6241 LOGW("command(): null parameter or value size"); 6242 mCblk->serverIndex += size; 6243 continue; 6244 } 6245 uint32_t psize = sizeof(effect_param_t) + 6246 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 6247 param->vsize; 6248 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 6249 psize, 6250 p, 6251 &rsize, 6252 &reply); 6253 // stop at first error encountered 6254 if (ret != NO_ERROR) { 6255 status = ret; 6256 *(int *)pReplyData = reply; 6257 break; 6258 } else if (reply != NO_ERROR) { 6259 *(int *)pReplyData = reply; 6260 break; 6261 } 6262 mCblk->serverIndex += size; 6263 } 6264 mCblk->serverIndex = 0; 6265 mCblk->clientIndex = 0; 6266 return status; 6267 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6268 *(int *)pReplyData = NO_ERROR; 6269 return enable(); 6270 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6271 *(int *)pReplyData = NO_ERROR; 6272 return disable(); 6273 } 6274 6275 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6276} 6277 6278sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 6279 return mCblkMemory; 6280} 6281 6282void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 6283{ 6284 LOGV("setControl %p control %d", this, hasControl); 6285 6286 mHasControl = hasControl; 6287 if (signal && mEffectClient != 0) { 6288 mEffectClient->controlStatusChanged(hasControl); 6289 } 6290} 6291 6292void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6293 uint32_t cmdSize, 6294 void *pCmdData, 6295 uint32_t replySize, 6296 void *pReplyData) 6297{ 6298 if (mEffectClient != 0) { 6299 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6300 } 6301} 6302 6303 6304 6305void AudioFlinger::EffectHandle::setEnabled(bool enabled) 6306{ 6307 if (mEffectClient != 0) { 6308 mEffectClient->enableStatusChanged(enabled); 6309 } 6310} 6311 6312status_t AudioFlinger::EffectHandle::onTransact( 6313 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6314{ 6315 return BnEffect::onTransact(code, data, reply, flags); 6316} 6317 6318 6319void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 6320{ 6321 bool locked = tryLock(mCblk->lock); 6322 6323 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 6324 (mClient == NULL) ? getpid() : mClient->pid(), 6325 mPriority, 6326 mHasControl, 6327 !locked, 6328 mCblk->clientIndex, 6329 mCblk->serverIndex 6330 ); 6331 6332 if (locked) { 6333 mCblk->lock.unlock(); 6334 } 6335} 6336 6337#undef LOG_TAG 6338#define LOG_TAG "AudioFlinger::EffectChain" 6339 6340AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 6341 int sessionId) 6342 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), 6343 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 6344 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 6345{ 6346 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 6347} 6348 6349AudioFlinger::EffectChain::~EffectChain() 6350{ 6351 if (mOwnInBuffer) { 6352 delete mInBuffer; 6353 } 6354 6355} 6356 6357// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6358sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6359{ 6360 sp<EffectModule> effect; 6361 size_t size = mEffects.size(); 6362 6363 for (size_t i = 0; i < size; i++) { 6364 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6365 effect = mEffects[i]; 6366 break; 6367 } 6368 } 6369 return effect; 6370} 6371 6372// getEffectFromId_l() must be called with PlaybackThread::mLock held 6373sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6374{ 6375 sp<EffectModule> effect; 6376 size_t size = mEffects.size(); 6377 6378 for (size_t i = 0; i < size; i++) { 6379 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6380 if (id == 0 || mEffects[i]->id() == id) { 6381 effect = mEffects[i]; 6382 break; 6383 } 6384 } 6385 return effect; 6386} 6387 6388// Must be called with EffectChain::mLock locked 6389void AudioFlinger::EffectChain::process_l() 6390{ 6391 sp<ThreadBase> thread = mThread.promote(); 6392 if (thread == 0) { 6393 LOGW("process_l(): cannot promote mixer thread"); 6394 return; 6395 } 6396 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 6397 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 6398 bool tracksOnSession = false; 6399 if (!isGlobalSession) { 6400 tracksOnSession = (trackCnt() != 0); 6401 } 6402 6403 // if no track is active, input buffer must be cleared here as the mixer process 6404 // will not do it 6405 if (tracksOnSession && 6406 activeTrackCnt() == 0) { 6407 size_t numSamples = thread->frameCount() * thread->channelCount(); 6408 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6409 } 6410 6411 size_t size = mEffects.size(); 6412 // do not process effect if no track is present in same audio session 6413 if (isGlobalSession || tracksOnSession) { 6414 for (size_t i = 0; i < size; i++) { 6415 mEffects[i]->process(); 6416 } 6417 } 6418 for (size_t i = 0; i < size; i++) { 6419 mEffects[i]->updateState(); 6420 } 6421} 6422 6423// addEffect_l() must be called with PlaybackThread::mLock held 6424status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6425{ 6426 effect_descriptor_t desc = effect->desc(); 6427 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6428 6429 Mutex::Autolock _l(mLock); 6430 effect->setChain(this); 6431 sp<ThreadBase> thread = mThread.promote(); 6432 if (thread == 0) { 6433 return NO_INIT; 6434 } 6435 effect->setThread(thread); 6436 6437 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6438 // Auxiliary effects are inserted at the beginning of mEffects vector as 6439 // they are processed first and accumulated in chain input buffer 6440 mEffects.insertAt(effect, 0); 6441 6442 // the input buffer for auxiliary effect contains mono samples in 6443 // 32 bit format. This is to avoid saturation in AudoMixer 6444 // accumulation stage. Saturation is done in EffectModule::process() before 6445 // calling the process in effect engine 6446 size_t numSamples = thread->frameCount(); 6447 int32_t *buffer = new int32_t[numSamples]; 6448 memset(buffer, 0, numSamples * sizeof(int32_t)); 6449 effect->setInBuffer((int16_t *)buffer); 6450 // auxiliary effects output samples to chain input buffer for further processing 6451 // by insert effects 6452 effect->setOutBuffer(mInBuffer); 6453 } else { 6454 // Insert effects are inserted at the end of mEffects vector as they are processed 6455 // after track and auxiliary effects. 6456 // Insert effect order as a function of indicated preference: 6457 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6458 // another effect is present 6459 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6460 // last effect claiming first position 6461 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6462 // first effect claiming last position 6463 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6464 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6465 // already present 6466 6467 int size = (int)mEffects.size(); 6468 int idx_insert = size; 6469 int idx_insert_first = -1; 6470 int idx_insert_last = -1; 6471 6472 for (int i = 0; i < size; i++) { 6473 effect_descriptor_t d = mEffects[i]->desc(); 6474 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6475 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6476 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6477 // check invalid effect chaining combinations 6478 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6479 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6480 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6481 return INVALID_OPERATION; 6482 } 6483 // remember position of first insert effect and by default 6484 // select this as insert position for new effect 6485 if (idx_insert == size) { 6486 idx_insert = i; 6487 } 6488 // remember position of last insert effect claiming 6489 // first position 6490 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6491 idx_insert_first = i; 6492 } 6493 // remember position of first insert effect claiming 6494 // last position 6495 if (iPref == EFFECT_FLAG_INSERT_LAST && 6496 idx_insert_last == -1) { 6497 idx_insert_last = i; 6498 } 6499 } 6500 } 6501 6502 // modify idx_insert from first position if needed 6503 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6504 if (idx_insert_last != -1) { 6505 idx_insert = idx_insert_last; 6506 } else { 6507 idx_insert = size; 6508 } 6509 } else { 6510 if (idx_insert_first != -1) { 6511 idx_insert = idx_insert_first + 1; 6512 } 6513 } 6514 6515 // always read samples from chain input buffer 6516 effect->setInBuffer(mInBuffer); 6517 6518 // if last effect in the chain, output samples to chain 6519 // output buffer, otherwise to chain input buffer 6520 if (idx_insert == size) { 6521 if (idx_insert != 0) { 6522 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6523 mEffects[idx_insert-1]->configure(); 6524 } 6525 effect->setOutBuffer(mOutBuffer); 6526 } else { 6527 effect->setOutBuffer(mInBuffer); 6528 } 6529 mEffects.insertAt(effect, idx_insert); 6530 6531 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6532 } 6533 effect->configure(); 6534 return NO_ERROR; 6535} 6536 6537// removeEffect_l() must be called with PlaybackThread::mLock held 6538size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6539{ 6540 Mutex::Autolock _l(mLock); 6541 int size = (int)mEffects.size(); 6542 int i; 6543 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6544 6545 for (i = 0; i < size; i++) { 6546 if (effect == mEffects[i]) { 6547 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6548 delete[] effect->inBuffer(); 6549 } else { 6550 if (i == size - 1 && i != 0) { 6551 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6552 mEffects[i - 1]->configure(); 6553 } 6554 } 6555 mEffects.removeAt(i); 6556 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6557 break; 6558 } 6559 } 6560 6561 return mEffects.size(); 6562} 6563 6564// setDevice_l() must be called with PlaybackThread::mLock held 6565void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6566{ 6567 size_t size = mEffects.size(); 6568 for (size_t i = 0; i < size; i++) { 6569 mEffects[i]->setDevice(device); 6570 } 6571} 6572 6573// setMode_l() must be called with PlaybackThread::mLock held 6574void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6575{ 6576 size_t size = mEffects.size(); 6577 for (size_t i = 0; i < size; i++) { 6578 mEffects[i]->setMode(mode); 6579 } 6580} 6581 6582// setVolume_l() must be called with PlaybackThread::mLock held 6583bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6584{ 6585 uint32_t newLeft = *left; 6586 uint32_t newRight = *right; 6587 bool hasControl = false; 6588 int ctrlIdx = -1; 6589 size_t size = mEffects.size(); 6590 6591 // first update volume controller 6592 for (size_t i = size; i > 0; i--) { 6593 if (mEffects[i - 1]->isProcessEnabled() && 6594 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6595 ctrlIdx = i - 1; 6596 hasControl = true; 6597 break; 6598 } 6599 } 6600 6601 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6602 if (hasControl) { 6603 *left = mNewLeftVolume; 6604 *right = mNewRightVolume; 6605 } 6606 return hasControl; 6607 } 6608 6609 mVolumeCtrlIdx = ctrlIdx; 6610 mLeftVolume = newLeft; 6611 mRightVolume = newRight; 6612 6613 // second get volume update from volume controller 6614 if (ctrlIdx >= 0) { 6615 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6616 mNewLeftVolume = newLeft; 6617 mNewRightVolume = newRight; 6618 } 6619 // then indicate volume to all other effects in chain. 6620 // Pass altered volume to effects before volume controller 6621 // and requested volume to effects after controller 6622 uint32_t lVol = newLeft; 6623 uint32_t rVol = newRight; 6624 6625 for (size_t i = 0; i < size; i++) { 6626 if ((int)i == ctrlIdx) continue; 6627 // this also works for ctrlIdx == -1 when there is no volume controller 6628 if ((int)i > ctrlIdx) { 6629 lVol = *left; 6630 rVol = *right; 6631 } 6632 mEffects[i]->setVolume(&lVol, &rVol, false); 6633 } 6634 *left = newLeft; 6635 *right = newRight; 6636 6637 return hasControl; 6638} 6639 6640status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6641{ 6642 const size_t SIZE = 256; 6643 char buffer[SIZE]; 6644 String8 result; 6645 6646 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6647 result.append(buffer); 6648 6649 bool locked = tryLock(mLock); 6650 // failed to lock - AudioFlinger is probably deadlocked 6651 if (!locked) { 6652 result.append("\tCould not lock mutex:\n"); 6653 } 6654 6655 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6656 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6657 mEffects.size(), 6658 (uint32_t)mInBuffer, 6659 (uint32_t)mOutBuffer, 6660 mActiveTrackCnt); 6661 result.append(buffer); 6662 write(fd, result.string(), result.size()); 6663 6664 for (size_t i = 0; i < mEffects.size(); ++i) { 6665 sp<EffectModule> effect = mEffects[i]; 6666 if (effect != 0) { 6667 effect->dump(fd, args); 6668 } 6669 } 6670 6671 if (locked) { 6672 mLock.unlock(); 6673 } 6674 6675 return NO_ERROR; 6676} 6677 6678#undef LOG_TAG 6679#define LOG_TAG "AudioFlinger" 6680 6681// ---------------------------------------------------------------------------- 6682 6683status_t AudioFlinger::onTransact( 6684 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6685{ 6686 return BnAudioFlinger::onTransact(code, data, reply, flags); 6687} 6688 6689}; // namespace android 6690