AudioFlinger.cpp revision 0d255b2d9061ba31f13ada3fc0f7e51916407176
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_hal.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <media/EffectVisualizerApi.h> 54 55// ---------------------------------------------------------------------------- 56// the sim build doesn't have gettid 57 58#ifndef HAVE_GETTID 59# define gettid getpid 60#endif 61 62// ---------------------------------------------------------------------------- 63 64 65namespace android { 66 67static const char* kDeadlockedString = "AudioFlinger may be deadlocked\n"; 68static const char* kHardwareLockedString = "Hardware lock is taken\n"; 69 70//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 71static const float MAX_GAIN = 4096.0f; 72static const float MAX_GAIN_INT = 0x1000; 73 74// retry counts for buffer fill timeout 75// 50 * ~20msecs = 1 second 76static const int8_t kMaxTrackRetries = 50; 77static const int8_t kMaxTrackStartupRetries = 50; 78// allow less retry attempts on direct output thread. 79// direct outputs can be a scarce resource in audio hardware and should 80// be released as quickly as possible. 81static const int8_t kMaxTrackRetriesDirect = 2; 82 83static const int kDumpLockRetries = 50; 84static const int kDumpLockSleep = 20000; 85 86static const nsecs_t kWarningThrottle = seconds(5); 87 88 89// ---------------------------------------------------------------------------- 90 91static bool recordingAllowed() { 92 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 93 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 94 if (!ok) LOGE("Request requires android.permission.RECORD_AUDIO"); 95 return ok; 96} 97 98static bool settingsAllowed() { 99 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 100 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 101 if (!ok) LOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 102 return ok; 103} 104 105// To collect the amplifier usage 106static void addBatteryData(uint32_t params) { 107 sp<IBinder> binder = 108 defaultServiceManager()->getService(String16("media.player")); 109 sp<IMediaPlayerService> service = interface_cast<IMediaPlayerService>(binder); 110 if (service.get() == NULL) { 111 LOGW("Cannot connect to the MediaPlayerService for battery tracking"); 112 return; 113 } 114 115 service->addBatteryData(params); 116} 117 118static int load_audio_interface(const char *if_name, const hw_module_t **mod, 119 audio_hw_device_t **dev) 120{ 121 int rc; 122 123 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 124 if (rc) 125 goto out; 126 127 rc = audio_hw_device_open(*mod, dev); 128 LOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 129 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 130 if (rc) 131 goto out; 132 133 return 0; 134 135out: 136 *mod = NULL; 137 *dev = NULL; 138 return rc; 139} 140 141static const char *audio_interfaces[] = { 142 "primary", 143 "a2dp", 144 "usb", 145}; 146#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 147 148// ---------------------------------------------------------------------------- 149 150AudioFlinger::AudioFlinger() 151 : BnAudioFlinger(), 152 mPrimaryHardwareDev(0), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1) 153{ 154} 155 156void AudioFlinger::onFirstRef() 157{ 158 int rc = 0; 159 160 Mutex::Autolock _l(mLock); 161 162 /* TODO: move all this work into an Init() function */ 163 mHardwareStatus = AUDIO_HW_IDLE; 164 165 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 166 const hw_module_t *mod; 167 audio_hw_device_t *dev; 168 169 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 170 if (rc) 171 continue; 172 173 LOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 174 mod->name, mod->id); 175 mAudioHwDevs.push(dev); 176 177 if (!mPrimaryHardwareDev) { 178 mPrimaryHardwareDev = dev; 179 LOGI("Using '%s' (%s.%s) as the primary audio interface", 180 mod->name, mod->id, audio_interfaces[i]); 181 } 182 } 183 184 mHardwareStatus = AUDIO_HW_INIT; 185 186 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 187 LOGE("Primary audio interface not found"); 188 return; 189 } 190 191 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 192 audio_hw_device_t *dev = mAudioHwDevs[i]; 193 194 mHardwareStatus = AUDIO_HW_INIT; 195 rc = dev->init_check(dev); 196 if (rc == 0) { 197 AutoMutex lock(mHardwareLock); 198 199 mMode = AUDIO_MODE_NORMAL; 200 mHardwareStatus = AUDIO_HW_SET_MODE; 201 dev->set_mode(dev, mMode); 202 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 203 dev->set_master_volume(dev, 1.0f); 204 mHardwareStatus = AUDIO_HW_IDLE; 205 } 206 } 207} 208 209status_t AudioFlinger::initCheck() const 210{ 211 Mutex::Autolock _l(mLock); 212 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 213 return NO_INIT; 214 return NO_ERROR; 215} 216 217AudioFlinger::~AudioFlinger() 218{ 219 int num_devs = mAudioHwDevs.size(); 220 221 while (!mRecordThreads.isEmpty()) { 222 // closeInput() will remove first entry from mRecordThreads 223 closeInput(mRecordThreads.keyAt(0)); 224 } 225 while (!mPlaybackThreads.isEmpty()) { 226 // closeOutput() will remove first entry from mPlaybackThreads 227 closeOutput(mPlaybackThreads.keyAt(0)); 228 } 229 230 for (int i = 0; i < num_devs; i++) { 231 audio_hw_device_t *dev = mAudioHwDevs[i]; 232 audio_hw_device_close(dev); 233 } 234 mAudioHwDevs.clear(); 235} 236 237audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 238{ 239 /* first matching HW device is returned */ 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 if ((dev->get_supported_devices(dev) & devices) == devices) 243 return dev; 244 } 245 return NULL; 246} 247 248status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 249{ 250 const size_t SIZE = 256; 251 char buffer[SIZE]; 252 String8 result; 253 254 result.append("Clients:\n"); 255 for (size_t i = 0; i < mClients.size(); ++i) { 256 wp<Client> wClient = mClients.valueAt(i); 257 if (wClient != 0) { 258 sp<Client> client = wClient.promote(); 259 if (client != 0) { 260 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 261 result.append(buffer); 262 } 263 } 264 } 265 write(fd, result.string(), result.size()); 266 return NO_ERROR; 267} 268 269 270status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 271{ 272 const size_t SIZE = 256; 273 char buffer[SIZE]; 274 String8 result; 275 int hardwareStatus = mHardwareStatus; 276 277 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 278 result.append(buffer); 279 write(fd, result.string(), result.size()); 280 return NO_ERROR; 281} 282 283status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 284{ 285 const size_t SIZE = 256; 286 char buffer[SIZE]; 287 String8 result; 288 snprintf(buffer, SIZE, "Permission Denial: " 289 "can't dump AudioFlinger from pid=%d, uid=%d\n", 290 IPCThreadState::self()->getCallingPid(), 291 IPCThreadState::self()->getCallingUid()); 292 result.append(buffer); 293 write(fd, result.string(), result.size()); 294 return NO_ERROR; 295} 296 297static bool tryLock(Mutex& mutex) 298{ 299 bool locked = false; 300 for (int i = 0; i < kDumpLockRetries; ++i) { 301 if (mutex.tryLock() == NO_ERROR) { 302 locked = true; 303 break; 304 } 305 usleep(kDumpLockSleep); 306 } 307 return locked; 308} 309 310status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 311{ 312 if (checkCallingPermission(String16("android.permission.DUMP")) == false) { 313 dumpPermissionDenial(fd, args); 314 } else { 315 // get state of hardware lock 316 bool hardwareLocked = tryLock(mHardwareLock); 317 if (!hardwareLocked) { 318 String8 result(kHardwareLockedString); 319 write(fd, result.string(), result.size()); 320 } else { 321 mHardwareLock.unlock(); 322 } 323 324 bool locked = tryLock(mLock); 325 326 // failed to lock - AudioFlinger is probably deadlocked 327 if (!locked) { 328 String8 result(kDeadlockedString); 329 write(fd, result.string(), result.size()); 330 } 331 332 dumpClients(fd, args); 333 dumpInternals(fd, args); 334 335 // dump playback threads 336 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 337 mPlaybackThreads.valueAt(i)->dump(fd, args); 338 } 339 340 // dump record threads 341 for (size_t i = 0; i < mRecordThreads.size(); i++) { 342 mRecordThreads.valueAt(i)->dump(fd, args); 343 } 344 345 // dump all hardware devs 346 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 347 audio_hw_device_t *dev = mAudioHwDevs[i]; 348 dev->dump(dev, fd); 349 } 350 if (locked) mLock.unlock(); 351 } 352 return NO_ERROR; 353} 354 355 356// IAudioFlinger interface 357 358 359sp<IAudioTrack> AudioFlinger::createTrack( 360 pid_t pid, 361 int streamType, 362 uint32_t sampleRate, 363 uint32_t format, 364 uint32_t channelMask, 365 int frameCount, 366 uint32_t flags, 367 const sp<IMemory>& sharedBuffer, 368 int output, 369 int *sessionId, 370 status_t *status) 371{ 372 sp<PlaybackThread::Track> track; 373 sp<TrackHandle> trackHandle; 374 sp<Client> client; 375 wp<Client> wclient; 376 status_t lStatus; 377 int lSessionId; 378 379 if (streamType >= AUDIO_STREAM_CNT) { 380 LOGE("invalid stream type"); 381 lStatus = BAD_VALUE; 382 goto Exit; 383 } 384 385 { 386 Mutex::Autolock _l(mLock); 387 PlaybackThread *thread = checkPlaybackThread_l(output); 388 PlaybackThread *effectThread = NULL; 389 if (thread == NULL) { 390 LOGE("unknown output thread"); 391 lStatus = BAD_VALUE; 392 goto Exit; 393 } 394 395 wclient = mClients.valueFor(pid); 396 397 if (wclient != NULL) { 398 client = wclient.promote(); 399 } else { 400 client = new Client(this, pid); 401 mClients.add(pid, client); 402 } 403 404 LOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 405 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 406 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 407 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 408 if (mPlaybackThreads.keyAt(i) != output) { 409 // prevent same audio session on different output threads 410 uint32_t sessions = t->hasAudioSession(*sessionId); 411 if (sessions & PlaybackThread::TRACK_SESSION) { 412 lStatus = BAD_VALUE; 413 goto Exit; 414 } 415 // check if an effect with same session ID is waiting for a track to be created 416 if (sessions & PlaybackThread::EFFECT_SESSION) { 417 effectThread = t.get(); 418 } 419 } 420 } 421 lSessionId = *sessionId; 422 } else { 423 // if no audio session id is provided, create one here 424 lSessionId = nextUniqueId_l(); 425 if (sessionId != NULL) { 426 *sessionId = lSessionId; 427 } 428 } 429 LOGV("createTrack() lSessionId: %d", lSessionId); 430 431 track = thread->createTrack_l(client, streamType, sampleRate, format, 432 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 433 434 // move effect chain to this output thread if an effect on same session was waiting 435 // for a track to be created 436 if (lStatus == NO_ERROR && effectThread != NULL) { 437 Mutex::Autolock _dl(thread->mLock); 438 Mutex::Autolock _sl(effectThread->mLock); 439 moveEffectChain_l(lSessionId, effectThread, thread, true); 440 } 441 } 442 if (lStatus == NO_ERROR) { 443 trackHandle = new TrackHandle(track); 444 } else { 445 // remove local strong reference to Client before deleting the Track so that the Client 446 // destructor is called by the TrackBase destructor with mLock held 447 client.clear(); 448 track.clear(); 449 } 450 451Exit: 452 if(status) { 453 *status = lStatus; 454 } 455 return trackHandle; 456} 457 458uint32_t AudioFlinger::sampleRate(int output) const 459{ 460 Mutex::Autolock _l(mLock); 461 PlaybackThread *thread = checkPlaybackThread_l(output); 462 if (thread == NULL) { 463 LOGW("sampleRate() unknown thread %d", output); 464 return 0; 465 } 466 return thread->sampleRate(); 467} 468 469int AudioFlinger::channelCount(int output) const 470{ 471 Mutex::Autolock _l(mLock); 472 PlaybackThread *thread = checkPlaybackThread_l(output); 473 if (thread == NULL) { 474 LOGW("channelCount() unknown thread %d", output); 475 return 0; 476 } 477 return thread->channelCount(); 478} 479 480uint32_t AudioFlinger::format(int output) const 481{ 482 Mutex::Autolock _l(mLock); 483 PlaybackThread *thread = checkPlaybackThread_l(output); 484 if (thread == NULL) { 485 LOGW("format() unknown thread %d", output); 486 return 0; 487 } 488 return thread->format(); 489} 490 491size_t AudioFlinger::frameCount(int output) const 492{ 493 Mutex::Autolock _l(mLock); 494 PlaybackThread *thread = checkPlaybackThread_l(output); 495 if (thread == NULL) { 496 LOGW("frameCount() unknown thread %d", output); 497 return 0; 498 } 499 return thread->frameCount(); 500} 501 502uint32_t AudioFlinger::latency(int output) const 503{ 504 Mutex::Autolock _l(mLock); 505 PlaybackThread *thread = checkPlaybackThread_l(output); 506 if (thread == NULL) { 507 LOGW("latency() unknown thread %d", output); 508 return 0; 509 } 510 return thread->latency(); 511} 512 513status_t AudioFlinger::setMasterVolume(float value) 514{ 515 // check calling permissions 516 if (!settingsAllowed()) { 517 return PERMISSION_DENIED; 518 } 519 520 // when hw supports master volume, don't scale in sw mixer 521 { // scope for the lock 522 AutoMutex lock(mHardwareLock); 523 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 524 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 525 value = 1.0f; 526 } 527 mHardwareStatus = AUDIO_HW_IDLE; 528 } 529 530 Mutex::Autolock _l(mLock); 531 mMasterVolume = value; 532 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 533 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 534 535 return NO_ERROR; 536} 537 538status_t AudioFlinger::setMode(int mode) 539{ 540 status_t ret; 541 542 // check calling permissions 543 if (!settingsAllowed()) { 544 return PERMISSION_DENIED; 545 } 546 if ((mode < 0) || (mode >= AUDIO_MODE_CNT)) { 547 LOGW("Illegal value: setMode(%d)", mode); 548 return BAD_VALUE; 549 } 550 551 { // scope for the lock 552 AutoMutex lock(mHardwareLock); 553 mHardwareStatus = AUDIO_HW_SET_MODE; 554 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 555 mHardwareStatus = AUDIO_HW_IDLE; 556 } 557 558 if (NO_ERROR == ret) { 559 Mutex::Autolock _l(mLock); 560 mMode = mode; 561 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 562 mPlaybackThreads.valueAt(i)->setMode(mode); 563 } 564 565 return ret; 566} 567 568status_t AudioFlinger::setMicMute(bool state) 569{ 570 // check calling permissions 571 if (!settingsAllowed()) { 572 return PERMISSION_DENIED; 573 } 574 575 AutoMutex lock(mHardwareLock); 576 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 577 status_t ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 578 mHardwareStatus = AUDIO_HW_IDLE; 579 return ret; 580} 581 582bool AudioFlinger::getMicMute() const 583{ 584 bool state = AUDIO_MODE_INVALID; 585 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 586 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 587 mHardwareStatus = AUDIO_HW_IDLE; 588 return state; 589} 590 591status_t AudioFlinger::setMasterMute(bool muted) 592{ 593 // check calling permissions 594 if (!settingsAllowed()) { 595 return PERMISSION_DENIED; 596 } 597 598 Mutex::Autolock _l(mLock); 599 mMasterMute = muted; 600 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 601 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 602 603 return NO_ERROR; 604} 605 606float AudioFlinger::masterVolume() const 607{ 608 return mMasterVolume; 609} 610 611bool AudioFlinger::masterMute() const 612{ 613 return mMasterMute; 614} 615 616status_t AudioFlinger::setStreamVolume(int stream, float value, int output) 617{ 618 // check calling permissions 619 if (!settingsAllowed()) { 620 return PERMISSION_DENIED; 621 } 622 623 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 624 return BAD_VALUE; 625 } 626 627 AutoMutex lock(mLock); 628 PlaybackThread *thread = NULL; 629 if (output) { 630 thread = checkPlaybackThread_l(output); 631 if (thread == NULL) { 632 return BAD_VALUE; 633 } 634 } 635 636 mStreamTypes[stream].volume = value; 637 638 if (thread == NULL) { 639 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 640 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 641 } 642 } else { 643 thread->setStreamVolume(stream, value); 644 } 645 646 return NO_ERROR; 647} 648 649status_t AudioFlinger::setStreamMute(int stream, bool muted) 650{ 651 // check calling permissions 652 if (!settingsAllowed()) { 653 return PERMISSION_DENIED; 654 } 655 656 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT || 657 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 658 return BAD_VALUE; 659 } 660 661 AutoMutex lock(mLock); 662 mStreamTypes[stream].mute = muted; 663 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 664 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 665 666 return NO_ERROR; 667} 668 669float AudioFlinger::streamVolume(int stream, int output) const 670{ 671 if (stream < 0 || uint32_t(stream) >= AUDIO_STREAM_CNT) { 672 return 0.0f; 673 } 674 675 AutoMutex lock(mLock); 676 float volume; 677 if (output) { 678 PlaybackThread *thread = checkPlaybackThread_l(output); 679 if (thread == NULL) { 680 return 0.0f; 681 } 682 volume = thread->streamVolume(stream); 683 } else { 684 volume = mStreamTypes[stream].volume; 685 } 686 687 return volume; 688} 689 690bool AudioFlinger::streamMute(int stream) const 691{ 692 if (stream < 0 || stream >= (int)AUDIO_STREAM_CNT) { 693 return true; 694 } 695 696 return mStreamTypes[stream].mute; 697} 698 699status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 700{ 701 status_t result; 702 703 LOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 704 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 705 // check calling permissions 706 if (!settingsAllowed()) { 707 return PERMISSION_DENIED; 708 } 709 710 // ioHandle == 0 means the parameters are global to the audio hardware interface 711 if (ioHandle == 0) { 712 AutoMutex lock(mHardwareLock); 713 mHardwareStatus = AUDIO_SET_PARAMETER; 714 status_t final_result = NO_ERROR; 715 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 716 audio_hw_device_t *dev = mAudioHwDevs[i]; 717 result = dev->set_parameters(dev, keyValuePairs.string()); 718 final_result = result ?: final_result; 719 } 720 mHardwareStatus = AUDIO_HW_IDLE; 721 return final_result; 722 } 723 724 // hold a strong ref on thread in case closeOutput() or closeInput() is called 725 // and the thread is exited once the lock is released 726 sp<ThreadBase> thread; 727 { 728 Mutex::Autolock _l(mLock); 729 thread = checkPlaybackThread_l(ioHandle); 730 if (thread == NULL) { 731 thread = checkRecordThread_l(ioHandle); 732 } 733 } 734 if (thread != NULL) { 735 result = thread->setParameters(keyValuePairs); 736 return result; 737 } 738 return BAD_VALUE; 739} 740 741String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 742{ 743// LOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 744// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 745 746 if (ioHandle == 0) { 747 String8 out_s8; 748 749 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 750 audio_hw_device_t *dev = mAudioHwDevs[i]; 751 char *s = dev->get_parameters(dev, keys.string()); 752 out_s8 += String8(s); 753 free(s); 754 } 755 return out_s8; 756 } 757 758 Mutex::Autolock _l(mLock); 759 760 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 761 if (playbackThread != NULL) { 762 return playbackThread->getParameters(keys); 763 } 764 RecordThread *recordThread = checkRecordThread_l(ioHandle); 765 if (recordThread != NULL) { 766 return recordThread->getParameters(keys); 767 } 768 return String8(""); 769} 770 771size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, int format, int channelCount) 772{ 773 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 774} 775 776unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 777{ 778 if (ioHandle == 0) { 779 return 0; 780 } 781 782 Mutex::Autolock _l(mLock); 783 784 RecordThread *recordThread = checkRecordThread_l(ioHandle); 785 if (recordThread != NULL) { 786 return recordThread->getInputFramesLost(); 787 } 788 return 0; 789} 790 791status_t AudioFlinger::setVoiceVolume(float value) 792{ 793 // check calling permissions 794 if (!settingsAllowed()) { 795 return PERMISSION_DENIED; 796 } 797 798 AutoMutex lock(mHardwareLock); 799 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 800 status_t ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 801 mHardwareStatus = AUDIO_HW_IDLE; 802 803 return ret; 804} 805 806status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 807{ 808 status_t status; 809 810 Mutex::Autolock _l(mLock); 811 812 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 813 if (playbackThread != NULL) { 814 return playbackThread->getRenderPosition(halFrames, dspFrames); 815 } 816 817 return BAD_VALUE; 818} 819 820void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 821{ 822 823 Mutex::Autolock _l(mLock); 824 825 int pid = IPCThreadState::self()->getCallingPid(); 826 if (mNotificationClients.indexOfKey(pid) < 0) { 827 sp<NotificationClient> notificationClient = new NotificationClient(this, 828 client, 829 pid); 830 LOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 831 832 mNotificationClients.add(pid, notificationClient); 833 834 sp<IBinder> binder = client->asBinder(); 835 binder->linkToDeath(notificationClient); 836 837 // the config change is always sent from playback or record threads to avoid deadlock 838 // with AudioSystem::gLock 839 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 840 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 841 } 842 843 for (size_t i = 0; i < mRecordThreads.size(); i++) { 844 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 845 } 846 } 847} 848 849void AudioFlinger::removeNotificationClient(pid_t pid) 850{ 851 Mutex::Autolock _l(mLock); 852 853 int index = mNotificationClients.indexOfKey(pid); 854 if (index >= 0) { 855 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 856 LOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 857 mNotificationClients.removeItem(pid); 858 } 859} 860 861// audioConfigChanged_l() must be called with AudioFlinger::mLock held 862void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 863{ 864 size_t size = mNotificationClients.size(); 865 for (size_t i = 0; i < size; i++) { 866 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 867 } 868} 869 870// removeClient_l() must be called with AudioFlinger::mLock held 871void AudioFlinger::removeClient_l(pid_t pid) 872{ 873 LOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 874 mClients.removeItem(pid); 875} 876 877 878// ---------------------------------------------------------------------------- 879 880AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id) 881 : Thread(false), 882 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 883 mFrameSize(1), mFormat(0), mStandby(false), mId(id), mExiting(false) 884{ 885} 886 887AudioFlinger::ThreadBase::~ThreadBase() 888{ 889 mParamCond.broadcast(); 890 mNewParameters.clear(); 891} 892 893void AudioFlinger::ThreadBase::exit() 894{ 895 // keep a strong ref on ourself so that we wont get 896 // destroyed in the middle of requestExitAndWait() 897 sp <ThreadBase> strongMe = this; 898 899 LOGV("ThreadBase::exit"); 900 { 901 AutoMutex lock(&mLock); 902 mExiting = true; 903 requestExit(); 904 mWaitWorkCV.signal(); 905 } 906 requestExitAndWait(); 907} 908 909uint32_t AudioFlinger::ThreadBase::sampleRate() const 910{ 911 return mSampleRate; 912} 913 914int AudioFlinger::ThreadBase::channelCount() const 915{ 916 return (int)mChannelCount; 917} 918 919uint32_t AudioFlinger::ThreadBase::format() const 920{ 921 return mFormat; 922} 923 924size_t AudioFlinger::ThreadBase::frameCount() const 925{ 926 return mFrameCount; 927} 928 929status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 930{ 931 status_t status; 932 933 LOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 934 Mutex::Autolock _l(mLock); 935 936 mNewParameters.add(keyValuePairs); 937 mWaitWorkCV.signal(); 938 // wait condition with timeout in case the thread loop has exited 939 // before the request could be processed 940 if (mParamCond.waitRelative(mLock, seconds(2)) == NO_ERROR) { 941 status = mParamStatus; 942 mWaitWorkCV.signal(); 943 } else { 944 status = TIMED_OUT; 945 } 946 return status; 947} 948 949void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 950{ 951 Mutex::Autolock _l(mLock); 952 sendConfigEvent_l(event, param); 953} 954 955// sendConfigEvent_l() must be called with ThreadBase::mLock held 956void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 957{ 958 ConfigEvent *configEvent = new ConfigEvent(); 959 configEvent->mEvent = event; 960 configEvent->mParam = param; 961 mConfigEvents.add(configEvent); 962 LOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 963 mWaitWorkCV.signal(); 964} 965 966void AudioFlinger::ThreadBase::processConfigEvents() 967{ 968 mLock.lock(); 969 while(!mConfigEvents.isEmpty()) { 970 LOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 971 ConfigEvent *configEvent = mConfigEvents[0]; 972 mConfigEvents.removeAt(0); 973 // release mLock before locking AudioFlinger mLock: lock order is always 974 // AudioFlinger then ThreadBase to avoid cross deadlock 975 mLock.unlock(); 976 mAudioFlinger->mLock.lock(); 977 audioConfigChanged_l(configEvent->mEvent, configEvent->mParam); 978 mAudioFlinger->mLock.unlock(); 979 delete configEvent; 980 mLock.lock(); 981 } 982 mLock.unlock(); 983} 984 985status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 986{ 987 const size_t SIZE = 256; 988 char buffer[SIZE]; 989 String8 result; 990 991 bool locked = tryLock(mLock); 992 if (!locked) { 993 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 994 write(fd, buffer, strlen(buffer)); 995 } 996 997 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 998 result.append(buffer); 999 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1000 result.append(buffer); 1001 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1002 result.append(buffer); 1003 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1004 result.append(buffer); 1005 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1006 result.append(buffer); 1007 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1008 result.append(buffer); 1009 snprintf(buffer, SIZE, "Frame size: %d\n", mFrameSize); 1010 result.append(buffer); 1011 1012 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1013 result.append(buffer); 1014 result.append(" Index Command"); 1015 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1016 snprintf(buffer, SIZE, "\n %02d ", i); 1017 result.append(buffer); 1018 result.append(mNewParameters[i]); 1019 } 1020 1021 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1022 result.append(buffer); 1023 snprintf(buffer, SIZE, " Index event param\n"); 1024 result.append(buffer); 1025 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1026 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i]->mEvent, mConfigEvents[i]->mParam); 1027 result.append(buffer); 1028 } 1029 result.append("\n"); 1030 1031 write(fd, result.string(), result.size()); 1032 1033 if (locked) { 1034 mLock.unlock(); 1035 } 1036 return NO_ERROR; 1037} 1038 1039 1040// ---------------------------------------------------------------------------- 1041 1042AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1043 : ThreadBase(audioFlinger, id), 1044 mMixBuffer(0), mSuspended(0), mBytesWritten(0), mOutput(output), 1045 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1046 mDevice(device) 1047{ 1048 readOutputParameters(); 1049 1050 mMasterVolume = mAudioFlinger->masterVolume(); 1051 mMasterMute = mAudioFlinger->masterMute(); 1052 1053 for (int stream = 0; stream < AUDIO_STREAM_CNT; stream++) { 1054 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1055 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1056 } 1057} 1058 1059AudioFlinger::PlaybackThread::~PlaybackThread() 1060{ 1061 delete [] mMixBuffer; 1062} 1063 1064status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1065{ 1066 dumpInternals(fd, args); 1067 dumpTracks(fd, args); 1068 dumpEffectChains(fd, args); 1069 return NO_ERROR; 1070} 1071 1072status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1073{ 1074 const size_t SIZE = 256; 1075 char buffer[SIZE]; 1076 String8 result; 1077 1078 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1079 result.append(buffer); 1080 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1081 for (size_t i = 0; i < mTracks.size(); ++i) { 1082 sp<Track> track = mTracks[i]; 1083 if (track != 0) { 1084 track->dump(buffer, SIZE); 1085 result.append(buffer); 1086 } 1087 } 1088 1089 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1090 result.append(buffer); 1091 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1092 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1093 wp<Track> wTrack = mActiveTracks[i]; 1094 if (wTrack != 0) { 1095 sp<Track> track = wTrack.promote(); 1096 if (track != 0) { 1097 track->dump(buffer, SIZE); 1098 result.append(buffer); 1099 } 1100 } 1101 } 1102 write(fd, result.string(), result.size()); 1103 return NO_ERROR; 1104} 1105 1106status_t AudioFlinger::PlaybackThread::dumpEffectChains(int fd, const Vector<String16>& args) 1107{ 1108 const size_t SIZE = 256; 1109 char buffer[SIZE]; 1110 String8 result; 1111 1112 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1113 write(fd, buffer, strlen(buffer)); 1114 1115 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1116 sp<EffectChain> chain = mEffectChains[i]; 1117 if (chain != 0) { 1118 chain->dump(fd, args); 1119 } 1120 } 1121 return NO_ERROR; 1122} 1123 1124status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1125{ 1126 const size_t SIZE = 256; 1127 char buffer[SIZE]; 1128 String8 result; 1129 1130 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1131 result.append(buffer); 1132 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1133 result.append(buffer); 1134 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1135 result.append(buffer); 1136 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1137 result.append(buffer); 1138 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1139 result.append(buffer); 1140 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1141 result.append(buffer); 1142 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1143 result.append(buffer); 1144 write(fd, result.string(), result.size()); 1145 1146 dumpBase(fd, args); 1147 1148 return NO_ERROR; 1149} 1150 1151// Thread virtuals 1152status_t AudioFlinger::PlaybackThread::readyToRun() 1153{ 1154 if (mSampleRate == 0) { 1155 LOGE("No working audio driver found."); 1156 return NO_INIT; 1157 } 1158 LOGI("AudioFlinger's thread %p ready to run", this); 1159 return NO_ERROR; 1160} 1161 1162void AudioFlinger::PlaybackThread::onFirstRef() 1163{ 1164 const size_t SIZE = 256; 1165 char buffer[SIZE]; 1166 1167 snprintf(buffer, SIZE, "Playback Thread %p", this); 1168 1169 run(buffer, ANDROID_PRIORITY_URGENT_AUDIO); 1170} 1171 1172// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1173sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1174 const sp<AudioFlinger::Client>& client, 1175 int streamType, 1176 uint32_t sampleRate, 1177 uint32_t format, 1178 uint32_t channelMask, 1179 int frameCount, 1180 const sp<IMemory>& sharedBuffer, 1181 int sessionId, 1182 status_t *status) 1183{ 1184 sp<Track> track; 1185 status_t lStatus; 1186 1187 if (mType == DIRECT) { 1188 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1189 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1190 LOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1191 "for output %p with format %d", 1192 sampleRate, format, channelMask, mOutput, mFormat); 1193 lStatus = BAD_VALUE; 1194 goto Exit; 1195 } 1196 } 1197 } else { 1198 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1199 if (sampleRate > mSampleRate*2) { 1200 LOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1201 lStatus = BAD_VALUE; 1202 goto Exit; 1203 } 1204 } 1205 1206 if (mOutput == 0) { 1207 LOGE("Audio driver not initialized."); 1208 lStatus = NO_INIT; 1209 goto Exit; 1210 } 1211 1212 { // scope for mLock 1213 Mutex::Autolock _l(mLock); 1214 1215 // all tracks in same audio session must share the same routing strategy otherwise 1216 // conflicts will happen when tracks are moved from one output to another by audio policy 1217 // manager 1218 uint32_t strategy = 1219 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1220 for (size_t i = 0; i < mTracks.size(); ++i) { 1221 sp<Track> t = mTracks[i]; 1222 if (t != 0) { 1223 if (sessionId == t->sessionId() && 1224 strategy != AudioSystem::getStrategyForStream((audio_stream_type_t)t->type())) { 1225 lStatus = BAD_VALUE; 1226 goto Exit; 1227 } 1228 } 1229 } 1230 1231 track = new Track(this, client, streamType, sampleRate, format, 1232 channelMask, frameCount, sharedBuffer, sessionId); 1233 if (track->getCblk() == NULL || track->name() < 0) { 1234 lStatus = NO_MEMORY; 1235 goto Exit; 1236 } 1237 mTracks.add(track); 1238 1239 sp<EffectChain> chain = getEffectChain_l(sessionId); 1240 if (chain != 0) { 1241 LOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1242 track->setMainBuffer(chain->inBuffer()); 1243 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1244 chain->incTrackCnt(); 1245 } 1246 } 1247 lStatus = NO_ERROR; 1248 1249Exit: 1250 if(status) { 1251 *status = lStatus; 1252 } 1253 return track; 1254} 1255 1256uint32_t AudioFlinger::PlaybackThread::latency() const 1257{ 1258 if (mOutput) { 1259 return mOutput->stream->get_latency(mOutput->stream); 1260 } 1261 else { 1262 return 0; 1263 } 1264} 1265 1266status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1267{ 1268 mMasterVolume = value; 1269 return NO_ERROR; 1270} 1271 1272status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1273{ 1274 mMasterMute = muted; 1275 return NO_ERROR; 1276} 1277 1278float AudioFlinger::PlaybackThread::masterVolume() const 1279{ 1280 return mMasterVolume; 1281} 1282 1283bool AudioFlinger::PlaybackThread::masterMute() const 1284{ 1285 return mMasterMute; 1286} 1287 1288status_t AudioFlinger::PlaybackThread::setStreamVolume(int stream, float value) 1289{ 1290 mStreamTypes[stream].volume = value; 1291 return NO_ERROR; 1292} 1293 1294status_t AudioFlinger::PlaybackThread::setStreamMute(int stream, bool muted) 1295{ 1296 mStreamTypes[stream].mute = muted; 1297 return NO_ERROR; 1298} 1299 1300float AudioFlinger::PlaybackThread::streamVolume(int stream) const 1301{ 1302 return mStreamTypes[stream].volume; 1303} 1304 1305bool AudioFlinger::PlaybackThread::streamMute(int stream) const 1306{ 1307 return mStreamTypes[stream].mute; 1308} 1309 1310// addTrack_l() must be called with ThreadBase::mLock held 1311status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1312{ 1313 status_t status = ALREADY_EXISTS; 1314 1315 // set retry count for buffer fill 1316 track->mRetryCount = kMaxTrackStartupRetries; 1317 if (mActiveTracks.indexOf(track) < 0) { 1318 // the track is newly added, make sure it fills up all its 1319 // buffers before playing. This is to ensure the client will 1320 // effectively get the latency it requested. 1321 track->mFillingUpStatus = Track::FS_FILLING; 1322 track->mResetDone = false; 1323 mActiveTracks.add(track); 1324 if (track->mainBuffer() != mMixBuffer) { 1325 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1326 if (chain != 0) { 1327 LOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1328 chain->incActiveTrackCnt(); 1329 } 1330 } 1331 1332 status = NO_ERROR; 1333 } 1334 1335 LOGV("mWaitWorkCV.broadcast"); 1336 mWaitWorkCV.broadcast(); 1337 1338 return status; 1339} 1340 1341// destroyTrack_l() must be called with ThreadBase::mLock held 1342void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1343{ 1344 track->mState = TrackBase::TERMINATED; 1345 if (mActiveTracks.indexOf(track) < 0) { 1346 removeTrack_l(track); 1347 } 1348} 1349 1350void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1351{ 1352 mTracks.remove(track); 1353 deleteTrackName_l(track->name()); 1354 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1355 if (chain != 0) { 1356 chain->decTrackCnt(); 1357 } 1358} 1359 1360String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1361{ 1362 String8 out_s8; 1363 char *s; 1364 1365 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1366 out_s8 = String8(s); 1367 free(s); 1368 return out_s8; 1369} 1370 1371// destroyTrack_l() must be called with AudioFlinger::mLock held 1372void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1373 AudioSystem::OutputDescriptor desc; 1374 void *param2 = 0; 1375 1376 LOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1377 1378 switch (event) { 1379 case AudioSystem::OUTPUT_OPENED: 1380 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1381 desc.channels = mChannelMask; 1382 desc.samplingRate = mSampleRate; 1383 desc.format = mFormat; 1384 desc.frameCount = mFrameCount; 1385 desc.latency = latency(); 1386 param2 = &desc; 1387 break; 1388 1389 case AudioSystem::STREAM_CONFIG_CHANGED: 1390 param2 = ¶m; 1391 case AudioSystem::OUTPUT_CLOSED: 1392 default: 1393 break; 1394 } 1395 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1396} 1397 1398void AudioFlinger::PlaybackThread::readOutputParameters() 1399{ 1400 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1401 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1402 mChannelCount = (uint16_t)popcount(mChannelMask); 1403 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1404 mFrameSize = (uint16_t)audio_stream_frame_size(&mOutput->stream->common); 1405 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1406 1407 // FIXME - Current mixer implementation only supports stereo output: Always 1408 // Allocate a stereo buffer even if HW output is mono. 1409 if (mMixBuffer != NULL) delete[] mMixBuffer; 1410 mMixBuffer = new int16_t[mFrameCount * 2]; 1411 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1412 1413 // force reconfiguration of effect chains and engines to take new buffer size and audio 1414 // parameters into account 1415 // Note that mLock is not held when readOutputParameters() is called from the constructor 1416 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1417 // matter. 1418 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1419 Vector< sp<EffectChain> > effectChains = mEffectChains; 1420 for (size_t i = 0; i < effectChains.size(); i ++) { 1421 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1422 } 1423} 1424 1425status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1426{ 1427 if (halFrames == 0 || dspFrames == 0) { 1428 return BAD_VALUE; 1429 } 1430 if (mOutput == 0) { 1431 return INVALID_OPERATION; 1432 } 1433 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1434 1435 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1436} 1437 1438uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1439{ 1440 Mutex::Autolock _l(mLock); 1441 uint32_t result = 0; 1442 if (getEffectChain_l(sessionId) != 0) { 1443 result = EFFECT_SESSION; 1444 } 1445 1446 for (size_t i = 0; i < mTracks.size(); ++i) { 1447 sp<Track> track = mTracks[i]; 1448 if (sessionId == track->sessionId() && 1449 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1450 result |= TRACK_SESSION; 1451 break; 1452 } 1453 } 1454 1455 return result; 1456} 1457 1458uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1459{ 1460 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1461 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1462 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1463 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1464 } 1465 for (size_t i = 0; i < mTracks.size(); i++) { 1466 sp<Track> track = mTracks[i]; 1467 if (sessionId == track->sessionId() && 1468 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1469 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1470 } 1471 } 1472 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1473} 1474 1475sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain(int sessionId) 1476{ 1477 Mutex::Autolock _l(mLock); 1478 return getEffectChain_l(sessionId); 1479} 1480 1481sp<AudioFlinger::EffectChain> AudioFlinger::PlaybackThread::getEffectChain_l(int sessionId) 1482{ 1483 sp<EffectChain> chain; 1484 1485 size_t size = mEffectChains.size(); 1486 for (size_t i = 0; i < size; i++) { 1487 if (mEffectChains[i]->sessionId() == sessionId) { 1488 chain = mEffectChains[i]; 1489 break; 1490 } 1491 } 1492 return chain; 1493} 1494 1495void AudioFlinger::PlaybackThread::setMode(uint32_t mode) 1496{ 1497 Mutex::Autolock _l(mLock); 1498 size_t size = mEffectChains.size(); 1499 for (size_t i = 0; i < size; i++) { 1500 mEffectChains[i]->setMode_l(mode); 1501 } 1502} 1503 1504// ---------------------------------------------------------------------------- 1505 1506AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1507 : PlaybackThread(audioFlinger, output, id, device), 1508 mAudioMixer(0) 1509{ 1510 mType = PlaybackThread::MIXER; 1511 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1512 1513 // FIXME - Current mixer implementation only supports stereo output 1514 if (mChannelCount == 1) { 1515 LOGE("Invalid audio hardware channel count"); 1516 } 1517} 1518 1519AudioFlinger::MixerThread::~MixerThread() 1520{ 1521 delete mAudioMixer; 1522} 1523 1524bool AudioFlinger::MixerThread::threadLoop() 1525{ 1526 Vector< sp<Track> > tracksToRemove; 1527 uint32_t mixerStatus = MIXER_IDLE; 1528 nsecs_t standbyTime = systemTime(); 1529 size_t mixBufferSize = mFrameCount * mFrameSize; 1530 // FIXME: Relaxed timing because of a certain device that can't meet latency 1531 // Should be reduced to 2x after the vendor fixes the driver issue 1532 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1533 nsecs_t lastWarning = 0; 1534 bool longStandbyExit = false; 1535 uint32_t activeSleepTime = activeSleepTimeUs(); 1536 uint32_t idleSleepTime = idleSleepTimeUs(); 1537 uint32_t sleepTime = idleSleepTime; 1538 Vector< sp<EffectChain> > effectChains; 1539 1540 while (!exitPending()) 1541 { 1542 processConfigEvents(); 1543 1544 mixerStatus = MIXER_IDLE; 1545 { // scope for mLock 1546 1547 Mutex::Autolock _l(mLock); 1548 1549 if (checkForNewParameters_l()) { 1550 mixBufferSize = mFrameCount * mFrameSize; 1551 // FIXME: Relaxed timing because of a certain device that can't meet latency 1552 // Should be reduced to 2x after the vendor fixes the driver issue 1553 maxPeriod = seconds(mFrameCount) / mSampleRate * 3; 1554 activeSleepTime = activeSleepTimeUs(); 1555 idleSleepTime = idleSleepTimeUs(); 1556 } 1557 1558 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1559 1560 // put audio hardware into standby after short delay 1561 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1562 mSuspended) { 1563 if (!mStandby) { 1564 LOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1565 mOutput->stream->common.standby(&mOutput->stream->common); 1566 mStandby = true; 1567 mBytesWritten = 0; 1568 } 1569 1570 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1571 // we're about to wait, flush the binder command buffer 1572 IPCThreadState::self()->flushCommands(); 1573 1574 if (exitPending()) break; 1575 1576 // wait until we have something to do... 1577 LOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1578 mWaitWorkCV.wait(mLock); 1579 LOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1580 1581 if (mMasterMute == false) { 1582 char value[PROPERTY_VALUE_MAX]; 1583 property_get("ro.audio.silent", value, "0"); 1584 if (atoi(value)) { 1585 LOGD("Silence is golden"); 1586 setMasterMute(true); 1587 } 1588 } 1589 1590 standbyTime = systemTime() + kStandbyTimeInNsecs; 1591 sleepTime = idleSleepTime; 1592 continue; 1593 } 1594 } 1595 1596 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1597 1598 // prevent any changes in effect chain list and in each effect chain 1599 // during mixing and effect process as the audio buffers could be deleted 1600 // or modified if an effect is created or deleted 1601 lockEffectChains_l(effectChains); 1602 } 1603 1604 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1605 // mix buffers... 1606 mAudioMixer->process(); 1607 sleepTime = 0; 1608 standbyTime = systemTime() + kStandbyTimeInNsecs; 1609 //TODO: delay standby when effects have a tail 1610 } else { 1611 // If no tracks are ready, sleep once for the duration of an output 1612 // buffer size, then write 0s to the output 1613 if (sleepTime == 0) { 1614 if (mixerStatus == MIXER_TRACKS_ENABLED) { 1615 sleepTime = activeSleepTime; 1616 } else { 1617 sleepTime = idleSleepTime; 1618 } 1619 } else if (mBytesWritten != 0 || 1620 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 1621 memset (mMixBuffer, 0, mixBufferSize); 1622 sleepTime = 0; 1623 LOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 1624 } 1625 // TODO add standby time extension fct of effect tail 1626 } 1627 1628 if (mSuspended) { 1629 sleepTime = suspendSleepTimeUs(); 1630 } 1631 // sleepTime == 0 means we must write to audio hardware 1632 if (sleepTime == 0) { 1633 for (size_t i = 0; i < effectChains.size(); i ++) { 1634 effectChains[i]->process_l(); 1635 } 1636 // enable changes in effect chain 1637 unlockEffectChains(effectChains); 1638 mLastWriteTime = systemTime(); 1639 mInWrite = true; 1640 mBytesWritten += mixBufferSize; 1641 1642 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 1643 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 1644 mNumWrites++; 1645 mInWrite = false; 1646 nsecs_t now = systemTime(); 1647 nsecs_t delta = now - mLastWriteTime; 1648 if (delta > maxPeriod) { 1649 mNumDelayedWrites++; 1650 if ((now - lastWarning) > kWarningThrottle) { 1651 LOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 1652 ns2ms(delta), mNumDelayedWrites, this); 1653 lastWarning = now; 1654 } 1655 if (mStandby) { 1656 longStandbyExit = true; 1657 } 1658 } 1659 mStandby = false; 1660 } else { 1661 // enable changes in effect chain 1662 unlockEffectChains(effectChains); 1663 usleep(sleepTime); 1664 } 1665 1666 // finally let go of all our tracks, without the lock held 1667 // since we can't guarantee the destructors won't acquire that 1668 // same lock. 1669 tracksToRemove.clear(); 1670 1671 // Effect chains will be actually deleted here if they were removed from 1672 // mEffectChains list during mixing or effects processing 1673 effectChains.clear(); 1674 } 1675 1676 if (!mStandby) { 1677 mOutput->stream->common.standby(&mOutput->stream->common); 1678 } 1679 1680 LOGV("MixerThread %p exiting", this); 1681 return false; 1682} 1683 1684// prepareTracks_l() must be called with ThreadBase::mLock held 1685uint32_t AudioFlinger::MixerThread::prepareTracks_l(const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 1686{ 1687 1688 uint32_t mixerStatus = MIXER_IDLE; 1689 // find out which tracks need to be processed 1690 size_t count = activeTracks.size(); 1691 size_t mixedTracks = 0; 1692 size_t tracksWithEffect = 0; 1693 1694 float masterVolume = mMasterVolume; 1695 bool masterMute = mMasterMute; 1696 1697 if (masterMute) { 1698 masterVolume = 0; 1699 } 1700 // Delegate master volume control to effect in output mix effect chain if needed 1701 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 1702 if (chain != 0) { 1703 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 1704 chain->setVolume_l(&v, &v); 1705 masterVolume = (float)((v + (1 << 23)) >> 24); 1706 chain.clear(); 1707 } 1708 1709 for (size_t i=0 ; i<count ; i++) { 1710 sp<Track> t = activeTracks[i].promote(); 1711 if (t == 0) continue; 1712 1713 Track* const track = t.get(); 1714 audio_track_cblk_t* cblk = track->cblk(); 1715 1716 // The first time a track is added we wait 1717 // for all its buffers to be filled before processing it 1718 mAudioMixer->setActiveTrack(track->name()); 1719 if (cblk->framesReady() && track->isReady() && 1720 !track->isPaused() && !track->isTerminated()) 1721 { 1722 //LOGV("track %d u=%08x, s=%08x [OK] on thread %p", track->name(), cblk->user, cblk->server, this); 1723 1724 mixedTracks++; 1725 1726 // track->mainBuffer() != mMixBuffer means there is an effect chain 1727 // connected to the track 1728 chain.clear(); 1729 if (track->mainBuffer() != mMixBuffer) { 1730 chain = getEffectChain_l(track->sessionId()); 1731 // Delegate volume control to effect in track effect chain if needed 1732 if (chain != 0) { 1733 tracksWithEffect++; 1734 } else { 1735 LOGW("prepareTracks_l(): track %08x attached to effect but no chain found on session %d", 1736 track->name(), track->sessionId()); 1737 } 1738 } 1739 1740 1741 int param = AudioMixer::VOLUME; 1742 if (track->mFillingUpStatus == Track::FS_FILLED) { 1743 // no ramp for the first volume setting 1744 track->mFillingUpStatus = Track::FS_ACTIVE; 1745 if (track->mState == TrackBase::RESUMING) { 1746 track->mState = TrackBase::ACTIVE; 1747 param = AudioMixer::RAMP_VOLUME; 1748 } 1749 mAudioMixer->setParameter(AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 1750 } else if (cblk->server != 0) { 1751 // If the track is stopped before the first frame was mixed, 1752 // do not apply ramp 1753 param = AudioMixer::RAMP_VOLUME; 1754 } 1755 1756 // compute volume for this track 1757 uint32_t vl, vr, va; 1758 if (track->isMuted() || track->isPausing() || 1759 mStreamTypes[track->type()].mute) { 1760 vl = vr = va = 0; 1761 if (track->isPausing()) { 1762 track->setPaused(); 1763 } 1764 } else { 1765 1766 // read original volumes with volume control 1767 float typeVolume = mStreamTypes[track->type()].volume; 1768 float v = masterVolume * typeVolume; 1769 vl = (uint32_t)(v * cblk->volume[0]) << 12; 1770 vr = (uint32_t)(v * cblk->volume[1]) << 12; 1771 1772 va = (uint32_t)(v * cblk->sendLevel); 1773 } 1774 // Delegate volume control to effect in track effect chain if needed 1775 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 1776 // Do not ramp volume if volume is controlled by effect 1777 param = AudioMixer::VOLUME; 1778 track->mHasVolumeController = true; 1779 } else { 1780 // force no volume ramp when volume controller was just disabled or removed 1781 // from effect chain to avoid volume spike 1782 if (track->mHasVolumeController) { 1783 param = AudioMixer::VOLUME; 1784 } 1785 track->mHasVolumeController = false; 1786 } 1787 1788 // Convert volumes from 8.24 to 4.12 format 1789 int16_t left, right, aux; 1790 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 1791 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1792 left = int16_t(v_clamped); 1793 v_clamped = (vr + (1 << 11)) >> 12; 1794 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 1795 right = int16_t(v_clamped); 1796 1797 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 1798 aux = int16_t(va); 1799 1800 // XXX: these things DON'T need to be done each time 1801 mAudioMixer->setBufferProvider(track); 1802 mAudioMixer->enable(AudioMixer::MIXING); 1803 1804 mAudioMixer->setParameter(param, AudioMixer::VOLUME0, (void *)left); 1805 mAudioMixer->setParameter(param, AudioMixer::VOLUME1, (void *)right); 1806 mAudioMixer->setParameter(param, AudioMixer::AUXLEVEL, (void *)aux); 1807 mAudioMixer->setParameter( 1808 AudioMixer::TRACK, 1809 AudioMixer::FORMAT, (void *)track->format()); 1810 mAudioMixer->setParameter( 1811 AudioMixer::TRACK, 1812 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 1813 mAudioMixer->setParameter( 1814 AudioMixer::RESAMPLE, 1815 AudioMixer::SAMPLE_RATE, 1816 (void *)(cblk->sampleRate)); 1817 mAudioMixer->setParameter( 1818 AudioMixer::TRACK, 1819 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 1820 mAudioMixer->setParameter( 1821 AudioMixer::TRACK, 1822 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 1823 1824 // reset retry count 1825 track->mRetryCount = kMaxTrackRetries; 1826 mixerStatus = MIXER_TRACKS_READY; 1827 } else { 1828 //LOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", track->name(), cblk->user, cblk->server, this); 1829 if (track->isStopped()) { 1830 track->reset(); 1831 } 1832 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 1833 // We have consumed all the buffers of this track. 1834 // Remove it from the list of active tracks. 1835 tracksToRemove->add(track); 1836 } else { 1837 // No buffers for this track. Give it a few chances to 1838 // fill a buffer, then remove it from active list. 1839 if (--(track->mRetryCount) <= 0) { 1840 LOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", track->name(), this); 1841 tracksToRemove->add(track); 1842 // indicate to client process that the track was disabled because of underrun 1843 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 1844 } else if (mixerStatus != MIXER_TRACKS_READY) { 1845 mixerStatus = MIXER_TRACKS_ENABLED; 1846 } 1847 } 1848 mAudioMixer->disable(AudioMixer::MIXING); 1849 } 1850 } 1851 1852 // remove all the tracks that need to be... 1853 count = tracksToRemove->size(); 1854 if (UNLIKELY(count)) { 1855 for (size_t i=0 ; i<count ; i++) { 1856 const sp<Track>& track = tracksToRemove->itemAt(i); 1857 mActiveTracks.remove(track); 1858 if (track->mainBuffer() != mMixBuffer) { 1859 chain = getEffectChain_l(track->sessionId()); 1860 if (chain != 0) { 1861 LOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 1862 chain->decActiveTrackCnt(); 1863 } 1864 } 1865 if (track->isTerminated()) { 1866 removeTrack_l(track); 1867 } 1868 } 1869 } 1870 1871 // mix buffer must be cleared if all tracks are connected to an 1872 // effect chain as in this case the mixer will not write to 1873 // mix buffer and track effects will accumulate into it 1874 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 1875 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 1876 } 1877 1878 return mixerStatus; 1879} 1880 1881void AudioFlinger::MixerThread::invalidateTracks(int streamType) 1882{ 1883 LOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 1884 this, streamType, mTracks.size()); 1885 Mutex::Autolock _l(mLock); 1886 1887 size_t size = mTracks.size(); 1888 for (size_t i = 0; i < size; i++) { 1889 sp<Track> t = mTracks[i]; 1890 if (t->type() == streamType) { 1891 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 1892 t->mCblk->cv.signal(); 1893 } 1894 } 1895} 1896 1897 1898// getTrackName_l() must be called with ThreadBase::mLock held 1899int AudioFlinger::MixerThread::getTrackName_l() 1900{ 1901 return mAudioMixer->getTrackName(); 1902} 1903 1904// deleteTrackName_l() must be called with ThreadBase::mLock held 1905void AudioFlinger::MixerThread::deleteTrackName_l(int name) 1906{ 1907 LOGV("remove track (%d) and delete from mixer", name); 1908 mAudioMixer->deleteTrackName(name); 1909} 1910 1911// checkForNewParameters_l() must be called with ThreadBase::mLock held 1912bool AudioFlinger::MixerThread::checkForNewParameters_l() 1913{ 1914 bool reconfig = false; 1915 1916 while (!mNewParameters.isEmpty()) { 1917 status_t status = NO_ERROR; 1918 String8 keyValuePair = mNewParameters[0]; 1919 AudioParameter param = AudioParameter(keyValuePair); 1920 int value; 1921 1922 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 1923 reconfig = true; 1924 } 1925 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 1926 if (value != AUDIO_FORMAT_PCM_16_BIT) { 1927 status = BAD_VALUE; 1928 } else { 1929 reconfig = true; 1930 } 1931 } 1932 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 1933 if (value != AUDIO_CHANNEL_OUT_STEREO) { 1934 status = BAD_VALUE; 1935 } else { 1936 reconfig = true; 1937 } 1938 } 1939 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 1940 // do not accept frame count changes if tracks are open as the track buffer 1941 // size depends on frame count and correct behavior would not be garantied 1942 // if frame count is changed after track creation 1943 if (!mTracks.isEmpty()) { 1944 status = INVALID_OPERATION; 1945 } else { 1946 reconfig = true; 1947 } 1948 } 1949 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 1950 // when changing the audio output device, call addBatteryData to notify 1951 // the change 1952 if ((int)mDevice != value) { 1953 uint32_t params = 0; 1954 // check whether speaker is on 1955 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 1956 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 1957 } 1958 1959 int deviceWithoutSpeaker 1960 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 1961 // check if any other device (except speaker) is on 1962 if (value & deviceWithoutSpeaker ) { 1963 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 1964 } 1965 1966 if (params != 0) { 1967 addBatteryData(params); 1968 } 1969 } 1970 1971 // forward device change to effects that have requested to be 1972 // aware of attached audio device. 1973 mDevice = (uint32_t)value; 1974 for (size_t i = 0; i < mEffectChains.size(); i++) { 1975 mEffectChains[i]->setDevice_l(mDevice); 1976 } 1977 } 1978 1979 if (status == NO_ERROR) { 1980 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 1981 keyValuePair.string()); 1982 if (!mStandby && status == INVALID_OPERATION) { 1983 mOutput->stream->common.standby(&mOutput->stream->common); 1984 mStandby = true; 1985 mBytesWritten = 0; 1986 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 1987 keyValuePair.string()); 1988 } 1989 if (status == NO_ERROR && reconfig) { 1990 delete mAudioMixer; 1991 readOutputParameters(); 1992 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1993 for (size_t i = 0; i < mTracks.size() ; i++) { 1994 int name = getTrackName_l(); 1995 if (name < 0) break; 1996 mTracks[i]->mName = name; 1997 // limit track sample rate to 2 x new output sample rate 1998 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 1999 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2000 } 2001 } 2002 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2003 } 2004 } 2005 2006 mNewParameters.removeAt(0); 2007 2008 mParamStatus = status; 2009 mParamCond.signal(); 2010 mWaitWorkCV.wait(mLock); 2011 } 2012 return reconfig; 2013} 2014 2015status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2016{ 2017 const size_t SIZE = 256; 2018 char buffer[SIZE]; 2019 String8 result; 2020 2021 PlaybackThread::dumpInternals(fd, args); 2022 2023 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2024 result.append(buffer); 2025 write(fd, result.string(), result.size()); 2026 return NO_ERROR; 2027} 2028 2029uint32_t AudioFlinger::MixerThread::activeSleepTimeUs() 2030{ 2031 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2032} 2033 2034uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2035{ 2036 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2037} 2038 2039uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2040{ 2041 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2042} 2043 2044// ---------------------------------------------------------------------------- 2045AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2046 : PlaybackThread(audioFlinger, output, id, device) 2047{ 2048 mType = PlaybackThread::DIRECT; 2049} 2050 2051AudioFlinger::DirectOutputThread::~DirectOutputThread() 2052{ 2053} 2054 2055 2056static inline int16_t clamp16(int32_t sample) 2057{ 2058 if ((sample>>15) ^ (sample>>31)) 2059 sample = 0x7FFF ^ (sample>>31); 2060 return sample; 2061} 2062 2063static inline 2064int32_t mul(int16_t in, int16_t v) 2065{ 2066#if defined(__arm__) && !defined(__thumb__) 2067 int32_t out; 2068 asm( "smulbb %[out], %[in], %[v] \n" 2069 : [out]"=r"(out) 2070 : [in]"%r"(in), [v]"r"(v) 2071 : ); 2072 return out; 2073#else 2074 return in * int32_t(v); 2075#endif 2076} 2077 2078void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2079{ 2080 // Do not apply volume on compressed audio 2081 if (!audio_is_linear_pcm(mFormat)) { 2082 return; 2083 } 2084 2085 // convert to signed 16 bit before volume calculation 2086 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2087 size_t count = mFrameCount * mChannelCount; 2088 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2089 int16_t *dst = mMixBuffer + count-1; 2090 while(count--) { 2091 *dst-- = (int16_t)(*src--^0x80) << 8; 2092 } 2093 } 2094 2095 size_t frameCount = mFrameCount; 2096 int16_t *out = mMixBuffer; 2097 if (ramp) { 2098 if (mChannelCount == 1) { 2099 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2100 int32_t vlInc = d / (int32_t)frameCount; 2101 int32_t vl = ((int32_t)mLeftVolShort << 16); 2102 do { 2103 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2104 out++; 2105 vl += vlInc; 2106 } while (--frameCount); 2107 2108 } else { 2109 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2110 int32_t vlInc = d / (int32_t)frameCount; 2111 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2112 int32_t vrInc = d / (int32_t)frameCount; 2113 int32_t vl = ((int32_t)mLeftVolShort << 16); 2114 int32_t vr = ((int32_t)mRightVolShort << 16); 2115 do { 2116 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2117 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2118 out += 2; 2119 vl += vlInc; 2120 vr += vrInc; 2121 } while (--frameCount); 2122 } 2123 } else { 2124 if (mChannelCount == 1) { 2125 do { 2126 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2127 out++; 2128 } while (--frameCount); 2129 } else { 2130 do { 2131 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2132 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2133 out += 2; 2134 } while (--frameCount); 2135 } 2136 } 2137 2138 // convert back to unsigned 8 bit after volume calculation 2139 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2140 size_t count = mFrameCount * mChannelCount; 2141 int16_t *src = mMixBuffer; 2142 uint8_t *dst = (uint8_t *)mMixBuffer; 2143 while(count--) { 2144 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2145 } 2146 } 2147 2148 mLeftVolShort = leftVol; 2149 mRightVolShort = rightVol; 2150} 2151 2152bool AudioFlinger::DirectOutputThread::threadLoop() 2153{ 2154 uint32_t mixerStatus = MIXER_IDLE; 2155 sp<Track> trackToRemove; 2156 sp<Track> activeTrack; 2157 nsecs_t standbyTime = systemTime(); 2158 int8_t *curBuf; 2159 size_t mixBufferSize = mFrameCount*mFrameSize; 2160 uint32_t activeSleepTime = activeSleepTimeUs(); 2161 uint32_t idleSleepTime = idleSleepTimeUs(); 2162 uint32_t sleepTime = idleSleepTime; 2163 // use shorter standby delay as on normal output to release 2164 // hardware resources as soon as possible 2165 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2166 2167 while (!exitPending()) 2168 { 2169 bool rampVolume; 2170 uint16_t leftVol; 2171 uint16_t rightVol; 2172 Vector< sp<EffectChain> > effectChains; 2173 2174 processConfigEvents(); 2175 2176 mixerStatus = MIXER_IDLE; 2177 2178 { // scope for the mLock 2179 2180 Mutex::Autolock _l(mLock); 2181 2182 if (checkForNewParameters_l()) { 2183 mixBufferSize = mFrameCount*mFrameSize; 2184 activeSleepTime = activeSleepTimeUs(); 2185 idleSleepTime = idleSleepTimeUs(); 2186 standbyDelay = microseconds(activeSleepTime*2); 2187 } 2188 2189 // put audio hardware into standby after short delay 2190 if UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2191 mSuspended) { 2192 // wait until we have something to do... 2193 if (!mStandby) { 2194 LOGV("Audio hardware entering standby, mixer %p\n", this); 2195 mOutput->stream->common.standby(&mOutput->stream->common); 2196 mStandby = true; 2197 mBytesWritten = 0; 2198 } 2199 2200 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2201 // we're about to wait, flush the binder command buffer 2202 IPCThreadState::self()->flushCommands(); 2203 2204 if (exitPending()) break; 2205 2206 LOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2207 mWaitWorkCV.wait(mLock); 2208 LOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2209 2210 if (mMasterMute == false) { 2211 char value[PROPERTY_VALUE_MAX]; 2212 property_get("ro.audio.silent", value, "0"); 2213 if (atoi(value)) { 2214 LOGD("Silence is golden"); 2215 setMasterMute(true); 2216 } 2217 } 2218 2219 standbyTime = systemTime() + standbyDelay; 2220 sleepTime = idleSleepTime; 2221 continue; 2222 } 2223 } 2224 2225 effectChains = mEffectChains; 2226 2227 // find out which tracks need to be processed 2228 if (mActiveTracks.size() != 0) { 2229 sp<Track> t = mActiveTracks[0].promote(); 2230 if (t == 0) continue; 2231 2232 Track* const track = t.get(); 2233 audio_track_cblk_t* cblk = track->cblk(); 2234 2235 // The first time a track is added we wait 2236 // for all its buffers to be filled before processing it 2237 if (cblk->framesReady() && track->isReady() && 2238 !track->isPaused() && !track->isTerminated()) 2239 { 2240 //LOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2241 2242 if (track->mFillingUpStatus == Track::FS_FILLED) { 2243 track->mFillingUpStatus = Track::FS_ACTIVE; 2244 mLeftVolFloat = mRightVolFloat = 0; 2245 mLeftVolShort = mRightVolShort = 0; 2246 if (track->mState == TrackBase::RESUMING) { 2247 track->mState = TrackBase::ACTIVE; 2248 rampVolume = true; 2249 } 2250 } else if (cblk->server != 0) { 2251 // If the track is stopped before the first frame was mixed, 2252 // do not apply ramp 2253 rampVolume = true; 2254 } 2255 // compute volume for this track 2256 float left, right; 2257 if (track->isMuted() || mMasterMute || track->isPausing() || 2258 mStreamTypes[track->type()].mute) { 2259 left = right = 0; 2260 if (track->isPausing()) { 2261 track->setPaused(); 2262 } 2263 } else { 2264 float typeVolume = mStreamTypes[track->type()].volume; 2265 float v = mMasterVolume * typeVolume; 2266 float v_clamped = v * cblk->volume[0]; 2267 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2268 left = v_clamped/MAX_GAIN; 2269 v_clamped = v * cblk->volume[1]; 2270 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2271 right = v_clamped/MAX_GAIN; 2272 } 2273 2274 if (left != mLeftVolFloat || right != mRightVolFloat) { 2275 mLeftVolFloat = left; 2276 mRightVolFloat = right; 2277 2278 // If audio HAL implements volume control, 2279 // force software volume to nominal value 2280 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2281 left = 1.0f; 2282 right = 1.0f; 2283 } 2284 2285 // Convert volumes from float to 8.24 2286 uint32_t vl = (uint32_t)(left * (1 << 24)); 2287 uint32_t vr = (uint32_t)(right * (1 << 24)); 2288 2289 // Delegate volume control to effect in track effect chain if needed 2290 // only one effect chain can be present on DirectOutputThread, so if 2291 // there is one, the track is connected to it 2292 if (!effectChains.isEmpty()) { 2293 // Do not ramp volume if volume is controlled by effect 2294 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2295 rampVolume = false; 2296 } 2297 } 2298 2299 // Convert volumes from 8.24 to 4.12 format 2300 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2301 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2302 leftVol = (uint16_t)v_clamped; 2303 v_clamped = (vr + (1 << 11)) >> 12; 2304 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2305 rightVol = (uint16_t)v_clamped; 2306 } else { 2307 leftVol = mLeftVolShort; 2308 rightVol = mRightVolShort; 2309 rampVolume = false; 2310 } 2311 2312 // reset retry count 2313 track->mRetryCount = kMaxTrackRetriesDirect; 2314 activeTrack = t; 2315 mixerStatus = MIXER_TRACKS_READY; 2316 } else { 2317 //LOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2318 if (track->isStopped()) { 2319 track->reset(); 2320 } 2321 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2322 // We have consumed all the buffers of this track. 2323 // Remove it from the list of active tracks. 2324 trackToRemove = track; 2325 } else { 2326 // No buffers for this track. Give it a few chances to 2327 // fill a buffer, then remove it from active list. 2328 if (--(track->mRetryCount) <= 0) { 2329 LOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2330 trackToRemove = track; 2331 } else { 2332 mixerStatus = MIXER_TRACKS_ENABLED; 2333 } 2334 } 2335 } 2336 } 2337 2338 // remove all the tracks that need to be... 2339 if (UNLIKELY(trackToRemove != 0)) { 2340 mActiveTracks.remove(trackToRemove); 2341 if (!effectChains.isEmpty()) { 2342 LOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2343 trackToRemove->sessionId()); 2344 effectChains[0]->decActiveTrackCnt(); 2345 } 2346 if (trackToRemove->isTerminated()) { 2347 removeTrack_l(trackToRemove); 2348 } 2349 } 2350 2351 lockEffectChains_l(effectChains); 2352 } 2353 2354 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2355 AudioBufferProvider::Buffer buffer; 2356 size_t frameCount = mFrameCount; 2357 curBuf = (int8_t *)mMixBuffer; 2358 // output audio to hardware 2359 while (frameCount) { 2360 buffer.frameCount = frameCount; 2361 activeTrack->getNextBuffer(&buffer); 2362 if (UNLIKELY(buffer.raw == 0)) { 2363 memset(curBuf, 0, frameCount * mFrameSize); 2364 break; 2365 } 2366 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2367 frameCount -= buffer.frameCount; 2368 curBuf += buffer.frameCount * mFrameSize; 2369 activeTrack->releaseBuffer(&buffer); 2370 } 2371 sleepTime = 0; 2372 standbyTime = systemTime() + standbyDelay; 2373 } else { 2374 if (sleepTime == 0) { 2375 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2376 sleepTime = activeSleepTime; 2377 } else { 2378 sleepTime = idleSleepTime; 2379 } 2380 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2381 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2382 sleepTime = 0; 2383 } 2384 } 2385 2386 if (mSuspended) { 2387 sleepTime = suspendSleepTimeUs(); 2388 } 2389 // sleepTime == 0 means we must write to audio hardware 2390 if (sleepTime == 0) { 2391 if (mixerStatus == MIXER_TRACKS_READY) { 2392 applyVolume(leftVol, rightVol, rampVolume); 2393 } 2394 for (size_t i = 0; i < effectChains.size(); i ++) { 2395 effectChains[i]->process_l(); 2396 } 2397 unlockEffectChains(effectChains); 2398 2399 mLastWriteTime = systemTime(); 2400 mInWrite = true; 2401 mBytesWritten += mixBufferSize; 2402 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2403 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2404 mNumWrites++; 2405 mInWrite = false; 2406 mStandby = false; 2407 } else { 2408 unlockEffectChains(effectChains); 2409 usleep(sleepTime); 2410 } 2411 2412 // finally let go of removed track, without the lock held 2413 // since we can't guarantee the destructors won't acquire that 2414 // same lock. 2415 trackToRemove.clear(); 2416 activeTrack.clear(); 2417 2418 // Effect chains will be actually deleted here if they were removed from 2419 // mEffectChains list during mixing or effects processing 2420 effectChains.clear(); 2421 } 2422 2423 if (!mStandby) { 2424 mOutput->stream->common.standby(&mOutput->stream->common); 2425 } 2426 2427 LOGV("DirectOutputThread %p exiting", this); 2428 return false; 2429} 2430 2431// getTrackName_l() must be called with ThreadBase::mLock held 2432int AudioFlinger::DirectOutputThread::getTrackName_l() 2433{ 2434 return 0; 2435} 2436 2437// deleteTrackName_l() must be called with ThreadBase::mLock held 2438void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2439{ 2440} 2441 2442// checkForNewParameters_l() must be called with ThreadBase::mLock held 2443bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2444{ 2445 bool reconfig = false; 2446 2447 while (!mNewParameters.isEmpty()) { 2448 status_t status = NO_ERROR; 2449 String8 keyValuePair = mNewParameters[0]; 2450 AudioParameter param = AudioParameter(keyValuePair); 2451 int value; 2452 2453 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2454 // do not accept frame count changes if tracks are open as the track buffer 2455 // size depends on frame count and correct behavior would not be garantied 2456 // if frame count is changed after track creation 2457 if (!mTracks.isEmpty()) { 2458 status = INVALID_OPERATION; 2459 } else { 2460 reconfig = true; 2461 } 2462 } 2463 if (status == NO_ERROR) { 2464 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2465 keyValuePair.string()); 2466 if (!mStandby && status == INVALID_OPERATION) { 2467 mOutput->stream->common.standby(&mOutput->stream->common); 2468 mStandby = true; 2469 mBytesWritten = 0; 2470 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2471 keyValuePair.string()); 2472 } 2473 if (status == NO_ERROR && reconfig) { 2474 readOutputParameters(); 2475 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2476 } 2477 } 2478 2479 mNewParameters.removeAt(0); 2480 2481 mParamStatus = status; 2482 mParamCond.signal(); 2483 mWaitWorkCV.wait(mLock); 2484 } 2485 return reconfig; 2486} 2487 2488uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2489{ 2490 uint32_t time; 2491 if (audio_is_linear_pcm(mFormat)) { 2492 time = (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 2493 } else { 2494 time = 10000; 2495 } 2496 return time; 2497} 2498 2499uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2500{ 2501 uint32_t time; 2502 if (audio_is_linear_pcm(mFormat)) { 2503 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2504 } else { 2505 time = 10000; 2506 } 2507 return time; 2508} 2509 2510uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2511{ 2512 uint32_t time; 2513 if (audio_is_linear_pcm(mFormat)) { 2514 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2515 } else { 2516 time = 10000; 2517 } 2518 return time; 2519} 2520 2521 2522// ---------------------------------------------------------------------------- 2523 2524AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2525 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2526{ 2527 mType = PlaybackThread::DUPLICATING; 2528 addOutputTrack(mainThread); 2529} 2530 2531AudioFlinger::DuplicatingThread::~DuplicatingThread() 2532{ 2533 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2534 mOutputTracks[i]->destroy(); 2535 } 2536 mOutputTracks.clear(); 2537} 2538 2539bool AudioFlinger::DuplicatingThread::threadLoop() 2540{ 2541 Vector< sp<Track> > tracksToRemove; 2542 uint32_t mixerStatus = MIXER_IDLE; 2543 nsecs_t standbyTime = systemTime(); 2544 size_t mixBufferSize = mFrameCount*mFrameSize; 2545 SortedVector< sp<OutputTrack> > outputTracks; 2546 uint32_t writeFrames = 0; 2547 uint32_t activeSleepTime = activeSleepTimeUs(); 2548 uint32_t idleSleepTime = idleSleepTimeUs(); 2549 uint32_t sleepTime = idleSleepTime; 2550 Vector< sp<EffectChain> > effectChains; 2551 2552 while (!exitPending()) 2553 { 2554 processConfigEvents(); 2555 2556 mixerStatus = MIXER_IDLE; 2557 { // scope for the mLock 2558 2559 Mutex::Autolock _l(mLock); 2560 2561 if (checkForNewParameters_l()) { 2562 mixBufferSize = mFrameCount*mFrameSize; 2563 updateWaitTime(); 2564 activeSleepTime = activeSleepTimeUs(); 2565 idleSleepTime = idleSleepTimeUs(); 2566 } 2567 2568 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2569 2570 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2571 outputTracks.add(mOutputTracks[i]); 2572 } 2573 2574 // put audio hardware into standby after short delay 2575 if UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2576 mSuspended) { 2577 if (!mStandby) { 2578 for (size_t i = 0; i < outputTracks.size(); i++) { 2579 outputTracks[i]->stop(); 2580 } 2581 mStandby = true; 2582 mBytesWritten = 0; 2583 } 2584 2585 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2586 // we're about to wait, flush the binder command buffer 2587 IPCThreadState::self()->flushCommands(); 2588 outputTracks.clear(); 2589 2590 if (exitPending()) break; 2591 2592 LOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 2593 mWaitWorkCV.wait(mLock); 2594 LOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 2595 if (mMasterMute == false) { 2596 char value[PROPERTY_VALUE_MAX]; 2597 property_get("ro.audio.silent", value, "0"); 2598 if (atoi(value)) { 2599 LOGD("Silence is golden"); 2600 setMasterMute(true); 2601 } 2602 } 2603 2604 standbyTime = systemTime() + kStandbyTimeInNsecs; 2605 sleepTime = idleSleepTime; 2606 continue; 2607 } 2608 } 2609 2610 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2611 2612 // prevent any changes in effect chain list and in each effect chain 2613 // during mixing and effect process as the audio buffers could be deleted 2614 // or modified if an effect is created or deleted 2615 lockEffectChains_l(effectChains); 2616 } 2617 2618 if (LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2619 // mix buffers... 2620 if (outputsReady(outputTracks)) { 2621 mAudioMixer->process(); 2622 } else { 2623 memset(mMixBuffer, 0, mixBufferSize); 2624 } 2625 sleepTime = 0; 2626 writeFrames = mFrameCount; 2627 } else { 2628 if (sleepTime == 0) { 2629 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2630 sleepTime = activeSleepTime; 2631 } else { 2632 sleepTime = idleSleepTime; 2633 } 2634 } else if (mBytesWritten != 0) { 2635 // flush remaining overflow buffers in output tracks 2636 for (size_t i = 0; i < outputTracks.size(); i++) { 2637 if (outputTracks[i]->isActive()) { 2638 sleepTime = 0; 2639 writeFrames = 0; 2640 memset(mMixBuffer, 0, mixBufferSize); 2641 break; 2642 } 2643 } 2644 } 2645 } 2646 2647 if (mSuspended) { 2648 sleepTime = suspendSleepTimeUs(); 2649 } 2650 // sleepTime == 0 means we must write to audio hardware 2651 if (sleepTime == 0) { 2652 for (size_t i = 0; i < effectChains.size(); i ++) { 2653 effectChains[i]->process_l(); 2654 } 2655 // enable changes in effect chain 2656 unlockEffectChains(effectChains); 2657 2658 standbyTime = systemTime() + kStandbyTimeInNsecs; 2659 for (size_t i = 0; i < outputTracks.size(); i++) { 2660 outputTracks[i]->write(mMixBuffer, writeFrames); 2661 } 2662 mStandby = false; 2663 mBytesWritten += mixBufferSize; 2664 } else { 2665 // enable changes in effect chain 2666 unlockEffectChains(effectChains); 2667 usleep(sleepTime); 2668 } 2669 2670 // finally let go of all our tracks, without the lock held 2671 // since we can't guarantee the destructors won't acquire that 2672 // same lock. 2673 tracksToRemove.clear(); 2674 outputTracks.clear(); 2675 2676 // Effect chains will be actually deleted here if they were removed from 2677 // mEffectChains list during mixing or effects processing 2678 effectChains.clear(); 2679 } 2680 2681 return false; 2682} 2683 2684void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 2685{ 2686 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 2687 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 2688 this, 2689 mSampleRate, 2690 mFormat, 2691 mChannelMask, 2692 frameCount); 2693 if (outputTrack->cblk() != NULL) { 2694 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 2695 mOutputTracks.add(outputTrack); 2696 LOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 2697 updateWaitTime(); 2698 } 2699} 2700 2701void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 2702{ 2703 Mutex::Autolock _l(mLock); 2704 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2705 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 2706 mOutputTracks[i]->destroy(); 2707 mOutputTracks.removeAt(i); 2708 updateWaitTime(); 2709 return; 2710 } 2711 } 2712 LOGV("removeOutputTrack(): unkonwn thread: %p", thread); 2713} 2714 2715void AudioFlinger::DuplicatingThread::updateWaitTime() 2716{ 2717 mWaitTimeMs = UINT_MAX; 2718 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2719 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 2720 if (strong != NULL) { 2721 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 2722 if (waitTimeMs < mWaitTimeMs) { 2723 mWaitTimeMs = waitTimeMs; 2724 } 2725 } 2726 } 2727} 2728 2729 2730bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 2731{ 2732 for (size_t i = 0; i < outputTracks.size(); i++) { 2733 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 2734 if (thread == 0) { 2735 LOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 2736 return false; 2737 } 2738 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2739 if (playbackThread->standby() && !playbackThread->isSuspended()) { 2740 LOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 2741 return false; 2742 } 2743 } 2744 return true; 2745} 2746 2747uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 2748{ 2749 return (mWaitTimeMs * 1000) / 2; 2750} 2751 2752// ---------------------------------------------------------------------------- 2753 2754// TrackBase constructor must be called with AudioFlinger::mLock held 2755AudioFlinger::ThreadBase::TrackBase::TrackBase( 2756 const wp<ThreadBase>& thread, 2757 const sp<Client>& client, 2758 uint32_t sampleRate, 2759 uint32_t format, 2760 uint32_t channelMask, 2761 int frameCount, 2762 uint32_t flags, 2763 const sp<IMemory>& sharedBuffer, 2764 int sessionId) 2765 : RefBase(), 2766 mThread(thread), 2767 mClient(client), 2768 mCblk(0), 2769 mFrameCount(0), 2770 mState(IDLE), 2771 mClientTid(-1), 2772 mFormat(format), 2773 mFlags(flags & ~SYSTEM_FLAGS_MASK), 2774 mSessionId(sessionId) 2775{ 2776 LOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 2777 2778 // LOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 2779 size_t size = sizeof(audio_track_cblk_t); 2780 uint8_t channelCount = popcount(channelMask); 2781 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 2782 if (sharedBuffer == 0) { 2783 size += bufferSize; 2784 } 2785 2786 if (client != NULL) { 2787 mCblkMemory = client->heap()->allocate(size); 2788 if (mCblkMemory != 0) { 2789 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 2790 if (mCblk) { // construct the shared structure in-place. 2791 new(mCblk) audio_track_cblk_t(); 2792 // clear all buffers 2793 mCblk->frameCount = frameCount; 2794 mCblk->sampleRate = sampleRate; 2795 mChannelCount = channelCount; 2796 mChannelMask = channelMask; 2797 if (sharedBuffer == 0) { 2798 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2799 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2800 // Force underrun condition to avoid false underrun callback until first data is 2801 // written to buffer (other flags are cleared) 2802 mCblk->flags = CBLK_UNDERRUN_ON; 2803 } else { 2804 mBuffer = sharedBuffer->pointer(); 2805 } 2806 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2807 } 2808 } else { 2809 LOGE("not enough memory for AudioTrack size=%u", size); 2810 client->heap()->dump("AudioTrack"); 2811 return; 2812 } 2813 } else { 2814 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 2815 if (mCblk) { // construct the shared structure in-place. 2816 new(mCblk) audio_track_cblk_t(); 2817 // clear all buffers 2818 mCblk->frameCount = frameCount; 2819 mCblk->sampleRate = sampleRate; 2820 mChannelCount = channelCount; 2821 mChannelMask = channelMask; 2822 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 2823 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 2824 // Force underrun condition to avoid false underrun callback until first data is 2825 // written to buffer (other flags are cleared) 2826 mCblk->flags = CBLK_UNDERRUN_ON; 2827 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 2828 } 2829 } 2830} 2831 2832AudioFlinger::ThreadBase::TrackBase::~TrackBase() 2833{ 2834 if (mCblk) { 2835 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 2836 if (mClient == NULL) { 2837 delete mCblk; 2838 } 2839 } 2840 mCblkMemory.clear(); // and free the shared memory 2841 if (mClient != NULL) { 2842 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 2843 mClient.clear(); 2844 } 2845} 2846 2847void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 2848{ 2849 buffer->raw = 0; 2850 mFrameCount = buffer->frameCount; 2851 step(); 2852 buffer->frameCount = 0; 2853} 2854 2855bool AudioFlinger::ThreadBase::TrackBase::step() { 2856 bool result; 2857 audio_track_cblk_t* cblk = this->cblk(); 2858 2859 result = cblk->stepServer(mFrameCount); 2860 if (!result) { 2861 LOGV("stepServer failed acquiring cblk mutex"); 2862 mFlags |= STEPSERVER_FAILED; 2863 } 2864 return result; 2865} 2866 2867void AudioFlinger::ThreadBase::TrackBase::reset() { 2868 audio_track_cblk_t* cblk = this->cblk(); 2869 2870 cblk->user = 0; 2871 cblk->server = 0; 2872 cblk->userBase = 0; 2873 cblk->serverBase = 0; 2874 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 2875 LOGV("TrackBase::reset"); 2876} 2877 2878sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 2879{ 2880 return mCblkMemory; 2881} 2882 2883int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 2884 return (int)mCblk->sampleRate; 2885} 2886 2887int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 2888 return (const int)mChannelCount; 2889} 2890 2891uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 2892 return mChannelMask; 2893} 2894 2895void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 2896 audio_track_cblk_t* cblk = this->cblk(); 2897 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*cblk->frameSize; 2898 int8_t *bufferEnd = bufferStart + frames * cblk->frameSize; 2899 2900 // Check validity of returned pointer in case the track control block would have been corrupted. 2901 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 2902 ((unsigned long)bufferStart & (unsigned long)(cblk->frameSize - 1))) { 2903 LOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 2904 server %d, serverBase %d, user %d, userBase %d", 2905 bufferStart, bufferEnd, mBuffer, mBufferEnd, 2906 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 2907 return 0; 2908 } 2909 2910 return bufferStart; 2911} 2912 2913// ---------------------------------------------------------------------------- 2914 2915// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 2916AudioFlinger::PlaybackThread::Track::Track( 2917 const wp<ThreadBase>& thread, 2918 const sp<Client>& client, 2919 int streamType, 2920 uint32_t sampleRate, 2921 uint32_t format, 2922 uint32_t channelMask, 2923 int frameCount, 2924 const sp<IMemory>& sharedBuffer, 2925 int sessionId) 2926 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 2927 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 2928 mAuxEffectId(0), mHasVolumeController(false) 2929{ 2930 if (mCblk != NULL) { 2931 sp<ThreadBase> baseThread = thread.promote(); 2932 if (baseThread != 0) { 2933 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 2934 mName = playbackThread->getTrackName_l(); 2935 mMainBuffer = playbackThread->mixBuffer(); 2936 } 2937 LOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 2938 if (mName < 0) { 2939 LOGE("no more track names available"); 2940 } 2941 mVolume[0] = 1.0f; 2942 mVolume[1] = 1.0f; 2943 mStreamType = streamType; 2944 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 2945 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 2946 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(int8_t); 2947 } 2948} 2949 2950AudioFlinger::PlaybackThread::Track::~Track() 2951{ 2952 LOGV("PlaybackThread::Track destructor"); 2953 sp<ThreadBase> thread = mThread.promote(); 2954 if (thread != 0) { 2955 Mutex::Autolock _l(thread->mLock); 2956 mState = TERMINATED; 2957 } 2958} 2959 2960void AudioFlinger::PlaybackThread::Track::destroy() 2961{ 2962 // NOTE: destroyTrack_l() can remove a strong reference to this Track 2963 // by removing it from mTracks vector, so there is a risk that this Tracks's 2964 // desctructor is called. As the destructor needs to lock mLock, 2965 // we must acquire a strong reference on this Track before locking mLock 2966 // here so that the destructor is called only when exiting this function. 2967 // On the other hand, as long as Track::destroy() is only called by 2968 // TrackHandle destructor, the TrackHandle still holds a strong ref on 2969 // this Track with its member mTrack. 2970 sp<Track> keep(this); 2971 { // scope for mLock 2972 sp<ThreadBase> thread = mThread.promote(); 2973 if (thread != 0) { 2974 if (!isOutputTrack()) { 2975 if (mState == ACTIVE || mState == RESUMING) { 2976 AudioSystem::stopOutput(thread->id(), 2977 (audio_stream_type_t)mStreamType, 2978 mSessionId); 2979 2980 // to track the speaker usage 2981 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 2982 } 2983 AudioSystem::releaseOutput(thread->id()); 2984 } 2985 Mutex::Autolock _l(thread->mLock); 2986 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 2987 playbackThread->destroyTrack_l(this); 2988 } 2989 } 2990} 2991 2992void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 2993{ 2994 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", 2995 mName - AudioMixer::TRACK0, 2996 (mClient == NULL) ? getpid() : mClient->pid(), 2997 mStreamType, 2998 mFormat, 2999 mChannelMask, 3000 mSessionId, 3001 mFrameCount, 3002 mState, 3003 mMute, 3004 mFillingUpStatus, 3005 mCblk->sampleRate, 3006 mCblk->volume[0], 3007 mCblk->volume[1], 3008 mCblk->server, 3009 mCblk->user, 3010 (int)mMainBuffer, 3011 (int)mAuxBuffer); 3012} 3013 3014status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3015{ 3016 audio_track_cblk_t* cblk = this->cblk(); 3017 uint32_t framesReady; 3018 uint32_t framesReq = buffer->frameCount; 3019 3020 // Check if last stepServer failed, try to step now 3021 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3022 if (!step()) goto getNextBuffer_exit; 3023 LOGV("stepServer recovered"); 3024 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3025 } 3026 3027 framesReady = cblk->framesReady(); 3028 3029 if (LIKELY(framesReady)) { 3030 uint32_t s = cblk->server; 3031 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3032 3033 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3034 if (framesReq > framesReady) { 3035 framesReq = framesReady; 3036 } 3037 if (s + framesReq > bufferEnd) { 3038 framesReq = bufferEnd - s; 3039 } 3040 3041 buffer->raw = getBuffer(s, framesReq); 3042 if (buffer->raw == 0) goto getNextBuffer_exit; 3043 3044 buffer->frameCount = framesReq; 3045 return NO_ERROR; 3046 } 3047 3048getNextBuffer_exit: 3049 buffer->raw = 0; 3050 buffer->frameCount = 0; 3051 LOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3052 return NOT_ENOUGH_DATA; 3053} 3054 3055bool AudioFlinger::PlaybackThread::Track::isReady() const { 3056 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3057 3058 if (mCblk->framesReady() >= mCblk->frameCount || 3059 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3060 mFillingUpStatus = FS_FILLED; 3061 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3062 return true; 3063 } 3064 return false; 3065} 3066 3067status_t AudioFlinger::PlaybackThread::Track::start() 3068{ 3069 status_t status = NO_ERROR; 3070 LOGV("start(%d), calling thread %d session %d", 3071 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3072 sp<ThreadBase> thread = mThread.promote(); 3073 if (thread != 0) { 3074 Mutex::Autolock _l(thread->mLock); 3075 int state = mState; 3076 // here the track could be either new, or restarted 3077 // in both cases "unstop" the track 3078 if (mState == PAUSED) { 3079 mState = TrackBase::RESUMING; 3080 LOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3081 } else { 3082 mState = TrackBase::ACTIVE; 3083 LOGV("? => ACTIVE (%d) on thread %p", mName, this); 3084 } 3085 3086 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3087 thread->mLock.unlock(); 3088 status = AudioSystem::startOutput(thread->id(), 3089 (audio_stream_type_t)mStreamType, 3090 mSessionId); 3091 thread->mLock.lock(); 3092 3093 // to track the speaker usage 3094 if (status == NO_ERROR) { 3095 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3096 } 3097 } 3098 if (status == NO_ERROR) { 3099 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3100 playbackThread->addTrack_l(this); 3101 } else { 3102 mState = state; 3103 } 3104 } else { 3105 status = BAD_VALUE; 3106 } 3107 return status; 3108} 3109 3110void AudioFlinger::PlaybackThread::Track::stop() 3111{ 3112 LOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3113 sp<ThreadBase> thread = mThread.promote(); 3114 if (thread != 0) { 3115 Mutex::Autolock _l(thread->mLock); 3116 int state = mState; 3117 if (mState > STOPPED) { 3118 mState = STOPPED; 3119 // If the track is not active (PAUSED and buffers full), flush buffers 3120 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3121 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3122 reset(); 3123 } 3124 LOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3125 } 3126 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3127 thread->mLock.unlock(); 3128 AudioSystem::stopOutput(thread->id(), 3129 (audio_stream_type_t)mStreamType, 3130 mSessionId); 3131 thread->mLock.lock(); 3132 3133 // to track the speaker usage 3134 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3135 } 3136 } 3137} 3138 3139void AudioFlinger::PlaybackThread::Track::pause() 3140{ 3141 LOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3142 sp<ThreadBase> thread = mThread.promote(); 3143 if (thread != 0) { 3144 Mutex::Autolock _l(thread->mLock); 3145 if (mState == ACTIVE || mState == RESUMING) { 3146 mState = PAUSING; 3147 LOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3148 if (!isOutputTrack()) { 3149 thread->mLock.unlock(); 3150 AudioSystem::stopOutput(thread->id(), 3151 (audio_stream_type_t)mStreamType, 3152 mSessionId); 3153 thread->mLock.lock(); 3154 3155 // to track the speaker usage 3156 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3157 } 3158 } 3159 } 3160} 3161 3162void AudioFlinger::PlaybackThread::Track::flush() 3163{ 3164 LOGV("flush(%d)", mName); 3165 sp<ThreadBase> thread = mThread.promote(); 3166 if (thread != 0) { 3167 Mutex::Autolock _l(thread->mLock); 3168 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3169 return; 3170 } 3171 // No point remaining in PAUSED state after a flush => go to 3172 // STOPPED state 3173 mState = STOPPED; 3174 3175 // do not reset the track if it is still in the process of being stopped or paused. 3176 // this will be done by prepareTracks_l() when the track is stopped. 3177 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3178 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3179 reset(); 3180 } 3181 } 3182} 3183 3184void AudioFlinger::PlaybackThread::Track::reset() 3185{ 3186 // Do not reset twice to avoid discarding data written just after a flush and before 3187 // the audioflinger thread detects the track is stopped. 3188 if (!mResetDone) { 3189 TrackBase::reset(); 3190 // Force underrun condition to avoid false underrun callback until first data is 3191 // written to buffer 3192 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3193 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3194 mFillingUpStatus = FS_FILLING; 3195 mResetDone = true; 3196 } 3197} 3198 3199void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3200{ 3201 mMute = muted; 3202} 3203 3204void AudioFlinger::PlaybackThread::Track::setVolume(float left, float right) 3205{ 3206 mVolume[0] = left; 3207 mVolume[1] = right; 3208} 3209 3210status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3211{ 3212 status_t status = DEAD_OBJECT; 3213 sp<ThreadBase> thread = mThread.promote(); 3214 if (thread != 0) { 3215 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3216 status = playbackThread->attachAuxEffect(this, EffectId); 3217 } 3218 return status; 3219} 3220 3221void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3222{ 3223 mAuxEffectId = EffectId; 3224 mAuxBuffer = buffer; 3225} 3226 3227// ---------------------------------------------------------------------------- 3228 3229// RecordTrack constructor must be called with AudioFlinger::mLock held 3230AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3231 const wp<ThreadBase>& thread, 3232 const sp<Client>& client, 3233 uint32_t sampleRate, 3234 uint32_t format, 3235 uint32_t channelMask, 3236 int frameCount, 3237 uint32_t flags, 3238 int sessionId) 3239 : TrackBase(thread, client, sampleRate, format, 3240 channelMask, frameCount, flags, 0, sessionId), 3241 mOverflow(false) 3242{ 3243 if (mCblk != NULL) { 3244 LOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3245 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3246 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3247 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3248 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3249 } else { 3250 mCblk->frameSize = sizeof(int8_t); 3251 } 3252 } 3253} 3254 3255AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3256{ 3257 sp<ThreadBase> thread = mThread.promote(); 3258 if (thread != 0) { 3259 AudioSystem::releaseInput(thread->id()); 3260 } 3261} 3262 3263status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3264{ 3265 audio_track_cblk_t* cblk = this->cblk(); 3266 uint32_t framesAvail; 3267 uint32_t framesReq = buffer->frameCount; 3268 3269 // Check if last stepServer failed, try to step now 3270 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3271 if (!step()) goto getNextBuffer_exit; 3272 LOGV("stepServer recovered"); 3273 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3274 } 3275 3276 framesAvail = cblk->framesAvailable_l(); 3277 3278 if (LIKELY(framesAvail)) { 3279 uint32_t s = cblk->server; 3280 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3281 3282 if (framesReq > framesAvail) { 3283 framesReq = framesAvail; 3284 } 3285 if (s + framesReq > bufferEnd) { 3286 framesReq = bufferEnd - s; 3287 } 3288 3289 buffer->raw = getBuffer(s, framesReq); 3290 if (buffer->raw == 0) goto getNextBuffer_exit; 3291 3292 buffer->frameCount = framesReq; 3293 return NO_ERROR; 3294 } 3295 3296getNextBuffer_exit: 3297 buffer->raw = 0; 3298 buffer->frameCount = 0; 3299 return NOT_ENOUGH_DATA; 3300} 3301 3302status_t AudioFlinger::RecordThread::RecordTrack::start() 3303{ 3304 sp<ThreadBase> thread = mThread.promote(); 3305 if (thread != 0) { 3306 RecordThread *recordThread = (RecordThread *)thread.get(); 3307 return recordThread->start(this); 3308 } else { 3309 return BAD_VALUE; 3310 } 3311} 3312 3313void AudioFlinger::RecordThread::RecordTrack::stop() 3314{ 3315 sp<ThreadBase> thread = mThread.promote(); 3316 if (thread != 0) { 3317 RecordThread *recordThread = (RecordThread *)thread.get(); 3318 recordThread->stop(this); 3319 TrackBase::reset(); 3320 // Force overerrun condition to avoid false overrun callback until first data is 3321 // read from buffer 3322 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3323 } 3324} 3325 3326void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3327{ 3328 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3329 (mClient == NULL) ? getpid() : mClient->pid(), 3330 mFormat, 3331 mChannelMask, 3332 mSessionId, 3333 mFrameCount, 3334 mState, 3335 mCblk->sampleRate, 3336 mCblk->server, 3337 mCblk->user); 3338} 3339 3340 3341// ---------------------------------------------------------------------------- 3342 3343AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3344 const wp<ThreadBase>& thread, 3345 DuplicatingThread *sourceThread, 3346 uint32_t sampleRate, 3347 uint32_t format, 3348 uint32_t channelMask, 3349 int frameCount) 3350 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3351 mActive(false), mSourceThread(sourceThread) 3352{ 3353 3354 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3355 if (mCblk != NULL) { 3356 mCblk->flags |= CBLK_DIRECTION_OUT; 3357 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3358 mCblk->volume[0] = mCblk->volume[1] = 0x1000; 3359 mOutBuffer.frameCount = 0; 3360 playbackThread->mTracks.add(this); 3361 LOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3362 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3363 mCblk, mBuffer, mCblk->buffers, 3364 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3365 } else { 3366 LOGW("Error creating output track on thread %p", playbackThread); 3367 } 3368} 3369 3370AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3371{ 3372 clearBufferQueue(); 3373} 3374 3375status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3376{ 3377 status_t status = Track::start(); 3378 if (status != NO_ERROR) { 3379 return status; 3380 } 3381 3382 mActive = true; 3383 mRetryCount = 127; 3384 return status; 3385} 3386 3387void AudioFlinger::PlaybackThread::OutputTrack::stop() 3388{ 3389 Track::stop(); 3390 clearBufferQueue(); 3391 mOutBuffer.frameCount = 0; 3392 mActive = false; 3393} 3394 3395bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3396{ 3397 Buffer *pInBuffer; 3398 Buffer inBuffer; 3399 uint32_t channelCount = mChannelCount; 3400 bool outputBufferFull = false; 3401 inBuffer.frameCount = frames; 3402 inBuffer.i16 = data; 3403 3404 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3405 3406 if (!mActive && frames != 0) { 3407 start(); 3408 sp<ThreadBase> thread = mThread.promote(); 3409 if (thread != 0) { 3410 MixerThread *mixerThread = (MixerThread *)thread.get(); 3411 if (mCblk->frameCount > frames){ 3412 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3413 uint32_t startFrames = (mCblk->frameCount - frames); 3414 pInBuffer = new Buffer; 3415 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3416 pInBuffer->frameCount = startFrames; 3417 pInBuffer->i16 = pInBuffer->mBuffer; 3418 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3419 mBufferQueue.add(pInBuffer); 3420 } else { 3421 LOGW ("OutputTrack::write() %p no more buffers in queue", this); 3422 } 3423 } 3424 } 3425 } 3426 3427 while (waitTimeLeftMs) { 3428 // First write pending buffers, then new data 3429 if (mBufferQueue.size()) { 3430 pInBuffer = mBufferQueue.itemAt(0); 3431 } else { 3432 pInBuffer = &inBuffer; 3433 } 3434 3435 if (pInBuffer->frameCount == 0) { 3436 break; 3437 } 3438 3439 if (mOutBuffer.frameCount == 0) { 3440 mOutBuffer.frameCount = pInBuffer->frameCount; 3441 nsecs_t startTime = systemTime(); 3442 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)AudioTrack::NO_MORE_BUFFERS) { 3443 LOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3444 outputBufferFull = true; 3445 break; 3446 } 3447 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3448 if (waitTimeLeftMs >= waitTimeMs) { 3449 waitTimeLeftMs -= waitTimeMs; 3450 } else { 3451 waitTimeLeftMs = 0; 3452 } 3453 } 3454 3455 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3456 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3457 mCblk->stepUser(outFrames); 3458 pInBuffer->frameCount -= outFrames; 3459 pInBuffer->i16 += outFrames * channelCount; 3460 mOutBuffer.frameCount -= outFrames; 3461 mOutBuffer.i16 += outFrames * channelCount; 3462 3463 if (pInBuffer->frameCount == 0) { 3464 if (mBufferQueue.size()) { 3465 mBufferQueue.removeAt(0); 3466 delete [] pInBuffer->mBuffer; 3467 delete pInBuffer; 3468 LOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3469 } else { 3470 break; 3471 } 3472 } 3473 } 3474 3475 // If we could not write all frames, allocate a buffer and queue it for next time. 3476 if (inBuffer.frameCount) { 3477 sp<ThreadBase> thread = mThread.promote(); 3478 if (thread != 0 && !thread->standby()) { 3479 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3480 pInBuffer = new Buffer; 3481 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3482 pInBuffer->frameCount = inBuffer.frameCount; 3483 pInBuffer->i16 = pInBuffer->mBuffer; 3484 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3485 mBufferQueue.add(pInBuffer); 3486 LOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3487 } else { 3488 LOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3489 } 3490 } 3491 } 3492 3493 // Calling write() with a 0 length buffer, means that no more data will be written: 3494 // If no more buffers are pending, fill output track buffer to make sure it is started 3495 // by output mixer. 3496 if (frames == 0 && mBufferQueue.size() == 0) { 3497 if (mCblk->user < mCblk->frameCount) { 3498 frames = mCblk->frameCount - mCblk->user; 3499 pInBuffer = new Buffer; 3500 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3501 pInBuffer->frameCount = frames; 3502 pInBuffer->i16 = pInBuffer->mBuffer; 3503 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3504 mBufferQueue.add(pInBuffer); 3505 } else if (mActive) { 3506 stop(); 3507 } 3508 } 3509 3510 return outputBufferFull; 3511} 3512 3513status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3514{ 3515 int active; 3516 status_t result; 3517 audio_track_cblk_t* cblk = mCblk; 3518 uint32_t framesReq = buffer->frameCount; 3519 3520// LOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3521 buffer->frameCount = 0; 3522 3523 uint32_t framesAvail = cblk->framesAvailable(); 3524 3525 3526 if (framesAvail == 0) { 3527 Mutex::Autolock _l(cblk->lock); 3528 goto start_loop_here; 3529 while (framesAvail == 0) { 3530 active = mActive; 3531 if (UNLIKELY(!active)) { 3532 LOGV("Not active and NO_MORE_BUFFERS"); 3533 return AudioTrack::NO_MORE_BUFFERS; 3534 } 3535 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 3536 if (result != NO_ERROR) { 3537 return AudioTrack::NO_MORE_BUFFERS; 3538 } 3539 // read the server count again 3540 start_loop_here: 3541 framesAvail = cblk->framesAvailable_l(); 3542 } 3543 } 3544 3545// if (framesAvail < framesReq) { 3546// return AudioTrack::NO_MORE_BUFFERS; 3547// } 3548 3549 if (framesReq > framesAvail) { 3550 framesReq = framesAvail; 3551 } 3552 3553 uint32_t u = cblk->user; 3554 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 3555 3556 if (u + framesReq > bufferEnd) { 3557 framesReq = bufferEnd - u; 3558 } 3559 3560 buffer->frameCount = framesReq; 3561 buffer->raw = (void *)cblk->buffer(u); 3562 return NO_ERROR; 3563} 3564 3565 3566void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 3567{ 3568 size_t size = mBufferQueue.size(); 3569 Buffer *pBuffer; 3570 3571 for (size_t i = 0; i < size; i++) { 3572 pBuffer = mBufferQueue.itemAt(i); 3573 delete [] pBuffer->mBuffer; 3574 delete pBuffer; 3575 } 3576 mBufferQueue.clear(); 3577} 3578 3579// ---------------------------------------------------------------------------- 3580 3581AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 3582 : RefBase(), 3583 mAudioFlinger(audioFlinger), 3584 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 3585 mPid(pid) 3586{ 3587 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 3588} 3589 3590// Client destructor must be called with AudioFlinger::mLock held 3591AudioFlinger::Client::~Client() 3592{ 3593 mAudioFlinger->removeClient_l(mPid); 3594} 3595 3596const sp<MemoryDealer>& AudioFlinger::Client::heap() const 3597{ 3598 return mMemoryDealer; 3599} 3600 3601// ---------------------------------------------------------------------------- 3602 3603AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 3604 const sp<IAudioFlingerClient>& client, 3605 pid_t pid) 3606 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 3607{ 3608} 3609 3610AudioFlinger::NotificationClient::~NotificationClient() 3611{ 3612 mClient.clear(); 3613} 3614 3615void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 3616{ 3617 sp<NotificationClient> keep(this); 3618 { 3619 mAudioFlinger->removeNotificationClient(mPid); 3620 } 3621} 3622 3623// ---------------------------------------------------------------------------- 3624 3625AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 3626 : BnAudioTrack(), 3627 mTrack(track) 3628{ 3629} 3630 3631AudioFlinger::TrackHandle::~TrackHandle() { 3632 // just stop the track on deletion, associated resources 3633 // will be freed from the main thread once all pending buffers have 3634 // been played. Unless it's not in the active track list, in which 3635 // case we free everything now... 3636 mTrack->destroy(); 3637} 3638 3639status_t AudioFlinger::TrackHandle::start() { 3640 return mTrack->start(); 3641} 3642 3643void AudioFlinger::TrackHandle::stop() { 3644 mTrack->stop(); 3645} 3646 3647void AudioFlinger::TrackHandle::flush() { 3648 mTrack->flush(); 3649} 3650 3651void AudioFlinger::TrackHandle::mute(bool e) { 3652 mTrack->mute(e); 3653} 3654 3655void AudioFlinger::TrackHandle::pause() { 3656 mTrack->pause(); 3657} 3658 3659void AudioFlinger::TrackHandle::setVolume(float left, float right) { 3660 mTrack->setVolume(left, right); 3661} 3662 3663sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 3664 return mTrack->getCblk(); 3665} 3666 3667status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 3668{ 3669 return mTrack->attachAuxEffect(EffectId); 3670} 3671 3672status_t AudioFlinger::TrackHandle::onTransact( 3673 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3674{ 3675 return BnAudioTrack::onTransact(code, data, reply, flags); 3676} 3677 3678// ---------------------------------------------------------------------------- 3679 3680sp<IAudioRecord> AudioFlinger::openRecord( 3681 pid_t pid, 3682 int input, 3683 uint32_t sampleRate, 3684 uint32_t format, 3685 uint32_t channelMask, 3686 int frameCount, 3687 uint32_t flags, 3688 int *sessionId, 3689 status_t *status) 3690{ 3691 sp<RecordThread::RecordTrack> recordTrack; 3692 sp<RecordHandle> recordHandle; 3693 sp<Client> client; 3694 wp<Client> wclient; 3695 status_t lStatus; 3696 RecordThread *thread; 3697 size_t inFrameCount; 3698 int lSessionId; 3699 3700 // check calling permissions 3701 if (!recordingAllowed()) { 3702 lStatus = PERMISSION_DENIED; 3703 goto Exit; 3704 } 3705 3706 // add client to list 3707 { // scope for mLock 3708 Mutex::Autolock _l(mLock); 3709 thread = checkRecordThread_l(input); 3710 if (thread == NULL) { 3711 lStatus = BAD_VALUE; 3712 goto Exit; 3713 } 3714 3715 wclient = mClients.valueFor(pid); 3716 if (wclient != NULL) { 3717 client = wclient.promote(); 3718 } else { 3719 client = new Client(this, pid); 3720 mClients.add(pid, client); 3721 } 3722 3723 // If no audio session id is provided, create one here 3724 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 3725 lSessionId = *sessionId; 3726 } else { 3727 lSessionId = nextUniqueId_l(); 3728 if (sessionId != NULL) { 3729 *sessionId = lSessionId; 3730 } 3731 } 3732 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 3733 recordTrack = new RecordThread::RecordTrack(thread, client, sampleRate, 3734 format, channelMask, frameCount, flags, lSessionId); 3735 } 3736 if (recordTrack->getCblk() == NULL) { 3737 // remove local strong reference to Client before deleting the RecordTrack so that the Client 3738 // destructor is called by the TrackBase destructor with mLock held 3739 client.clear(); 3740 recordTrack.clear(); 3741 lStatus = NO_MEMORY; 3742 goto Exit; 3743 } 3744 3745 // return to handle to client 3746 recordHandle = new RecordHandle(recordTrack); 3747 lStatus = NO_ERROR; 3748 3749Exit: 3750 if (status) { 3751 *status = lStatus; 3752 } 3753 return recordHandle; 3754} 3755 3756// ---------------------------------------------------------------------------- 3757 3758AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 3759 : BnAudioRecord(), 3760 mRecordTrack(recordTrack) 3761{ 3762} 3763 3764AudioFlinger::RecordHandle::~RecordHandle() { 3765 stop(); 3766} 3767 3768status_t AudioFlinger::RecordHandle::start() { 3769 LOGV("RecordHandle::start()"); 3770 return mRecordTrack->start(); 3771} 3772 3773void AudioFlinger::RecordHandle::stop() { 3774 LOGV("RecordHandle::stop()"); 3775 mRecordTrack->stop(); 3776} 3777 3778sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 3779 return mRecordTrack->getCblk(); 3780} 3781 3782status_t AudioFlinger::RecordHandle::onTransact( 3783 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3784{ 3785 return BnAudioRecord::onTransact(code, data, reply, flags); 3786} 3787 3788// ---------------------------------------------------------------------------- 3789 3790AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, AudioStreamIn *input, uint32_t sampleRate, uint32_t channels, int id) : 3791 ThreadBase(audioFlinger, id), 3792 mInput(input), mResampler(0), mRsmpOutBuffer(0), mRsmpInBuffer(0) 3793{ 3794 mReqChannelCount = popcount(channels); 3795 mReqSampleRate = sampleRate; 3796 readInputParameters(); 3797} 3798 3799 3800AudioFlinger::RecordThread::~RecordThread() 3801{ 3802 delete[] mRsmpInBuffer; 3803 if (mResampler != 0) { 3804 delete mResampler; 3805 delete[] mRsmpOutBuffer; 3806 } 3807} 3808 3809void AudioFlinger::RecordThread::onFirstRef() 3810{ 3811 const size_t SIZE = 256; 3812 char buffer[SIZE]; 3813 3814 snprintf(buffer, SIZE, "Record Thread %p", this); 3815 3816 run(buffer, PRIORITY_URGENT_AUDIO); 3817} 3818 3819bool AudioFlinger::RecordThread::threadLoop() 3820{ 3821 AudioBufferProvider::Buffer buffer; 3822 sp<RecordTrack> activeTrack; 3823 3824 nsecs_t lastWarning = 0; 3825 3826 // start recording 3827 while (!exitPending()) { 3828 3829 processConfigEvents(); 3830 3831 { // scope for mLock 3832 Mutex::Autolock _l(mLock); 3833 checkForNewParameters_l(); 3834 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 3835 if (!mStandby) { 3836 mInput->stream->common.standby(&mInput->stream->common); 3837 mStandby = true; 3838 } 3839 3840 if (exitPending()) break; 3841 3842 LOGV("RecordThread: loop stopping"); 3843 // go to sleep 3844 mWaitWorkCV.wait(mLock); 3845 LOGV("RecordThread: loop starting"); 3846 continue; 3847 } 3848 if (mActiveTrack != 0) { 3849 if (mActiveTrack->mState == TrackBase::PAUSING) { 3850 if (!mStandby) { 3851 mInput->stream->common.standby(&mInput->stream->common); 3852 mStandby = true; 3853 } 3854 mActiveTrack.clear(); 3855 mStartStopCond.broadcast(); 3856 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 3857 if (mReqChannelCount != mActiveTrack->channelCount()) { 3858 mActiveTrack.clear(); 3859 mStartStopCond.broadcast(); 3860 } else if (mBytesRead != 0) { 3861 // record start succeeds only if first read from audio input 3862 // succeeds 3863 if (mBytesRead > 0) { 3864 mActiveTrack->mState = TrackBase::ACTIVE; 3865 } else { 3866 mActiveTrack.clear(); 3867 } 3868 mStartStopCond.broadcast(); 3869 } 3870 mStandby = false; 3871 } 3872 } 3873 } 3874 3875 if (mActiveTrack != 0) { 3876 if (mActiveTrack->mState != TrackBase::ACTIVE && 3877 mActiveTrack->mState != TrackBase::RESUMING) { 3878 usleep(5000); 3879 continue; 3880 } 3881 buffer.frameCount = mFrameCount; 3882 if (LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 3883 size_t framesOut = buffer.frameCount; 3884 if (mResampler == 0) { 3885 // no resampling 3886 while (framesOut) { 3887 size_t framesIn = mFrameCount - mRsmpInIndex; 3888 if (framesIn) { 3889 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 3890 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 3891 if (framesIn > framesOut) 3892 framesIn = framesOut; 3893 mRsmpInIndex += framesIn; 3894 framesOut -= framesIn; 3895 if ((int)mChannelCount == mReqChannelCount || 3896 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 3897 memcpy(dst, src, framesIn * mFrameSize); 3898 } else { 3899 int16_t *src16 = (int16_t *)src; 3900 int16_t *dst16 = (int16_t *)dst; 3901 if (mChannelCount == 1) { 3902 while (framesIn--) { 3903 *dst16++ = *src16; 3904 *dst16++ = *src16++; 3905 } 3906 } else { 3907 while (framesIn--) { 3908 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 3909 src16 += 2; 3910 } 3911 } 3912 } 3913 } 3914 if (framesOut && mFrameCount == mRsmpInIndex) { 3915 if (framesOut == mFrameCount && 3916 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 3917 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 3918 framesOut = 0; 3919 } else { 3920 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 3921 mRsmpInIndex = 0; 3922 } 3923 if (mBytesRead < 0) { 3924 LOGE("Error reading audio input"); 3925 if (mActiveTrack->mState == TrackBase::ACTIVE) { 3926 // Force input into standby so that it tries to 3927 // recover at next read attempt 3928 mInput->stream->common.standby(&mInput->stream->common); 3929 usleep(5000); 3930 } 3931 mRsmpInIndex = mFrameCount; 3932 framesOut = 0; 3933 buffer.frameCount = 0; 3934 } 3935 } 3936 } 3937 } else { 3938 // resampling 3939 3940 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 3941 // alter output frame count as if we were expecting stereo samples 3942 if (mChannelCount == 1 && mReqChannelCount == 1) { 3943 framesOut >>= 1; 3944 } 3945 mResampler->resample(mRsmpOutBuffer, framesOut, this); 3946 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 3947 // are 32 bit aligned which should be always true. 3948 if (mChannelCount == 2 && mReqChannelCount == 1) { 3949 AudioMixer::ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 3950 // the resampler always outputs stereo samples: do post stereo to mono conversion 3951 int16_t *src = (int16_t *)mRsmpOutBuffer; 3952 int16_t *dst = buffer.i16; 3953 while (framesOut--) { 3954 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 3955 src += 2; 3956 } 3957 } else { 3958 AudioMixer::ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 3959 } 3960 3961 } 3962 mActiveTrack->releaseBuffer(&buffer); 3963 mActiveTrack->overflow(); 3964 } 3965 // client isn't retrieving buffers fast enough 3966 else { 3967 if (!mActiveTrack->setOverflow()) { 3968 nsecs_t now = systemTime(); 3969 if ((now - lastWarning) > kWarningThrottle) { 3970 LOGW("RecordThread: buffer overflow"); 3971 lastWarning = now; 3972 } 3973 } 3974 // Release the processor for a while before asking for a new buffer. 3975 // This will give the application more chance to read from the buffer and 3976 // clear the overflow. 3977 usleep(5000); 3978 } 3979 } 3980 } 3981 3982 if (!mStandby) { 3983 mInput->stream->common.standby(&mInput->stream->common); 3984 } 3985 mActiveTrack.clear(); 3986 3987 mStartStopCond.broadcast(); 3988 3989 LOGV("RecordThread %p exiting", this); 3990 return false; 3991} 3992 3993status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 3994{ 3995 LOGV("RecordThread::start"); 3996 sp <ThreadBase> strongMe = this; 3997 status_t status = NO_ERROR; 3998 { 3999 AutoMutex lock(&mLock); 4000 if (mActiveTrack != 0) { 4001 if (recordTrack != mActiveTrack.get()) { 4002 status = -EBUSY; 4003 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4004 mActiveTrack->mState = TrackBase::ACTIVE; 4005 } 4006 return status; 4007 } 4008 4009 recordTrack->mState = TrackBase::IDLE; 4010 mActiveTrack = recordTrack; 4011 mLock.unlock(); 4012 status_t status = AudioSystem::startInput(mId); 4013 mLock.lock(); 4014 if (status != NO_ERROR) { 4015 mActiveTrack.clear(); 4016 return status; 4017 } 4018 mRsmpInIndex = mFrameCount; 4019 mBytesRead = 0; 4020 if (mResampler != NULL) { 4021 mResampler->reset(); 4022 } 4023 mActiveTrack->mState = TrackBase::RESUMING; 4024 // signal thread to start 4025 LOGV("Signal record thread"); 4026 mWaitWorkCV.signal(); 4027 // do not wait for mStartStopCond if exiting 4028 if (mExiting) { 4029 mActiveTrack.clear(); 4030 status = INVALID_OPERATION; 4031 goto startError; 4032 } 4033 mStartStopCond.wait(mLock); 4034 if (mActiveTrack == 0) { 4035 LOGV("Record failed to start"); 4036 status = BAD_VALUE; 4037 goto startError; 4038 } 4039 LOGV("Record started OK"); 4040 return status; 4041 } 4042startError: 4043 AudioSystem::stopInput(mId); 4044 return status; 4045} 4046 4047void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4048 LOGV("RecordThread::stop"); 4049 sp <ThreadBase> strongMe = this; 4050 { 4051 AutoMutex lock(&mLock); 4052 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4053 mActiveTrack->mState = TrackBase::PAUSING; 4054 // do not wait for mStartStopCond if exiting 4055 if (mExiting) { 4056 return; 4057 } 4058 mStartStopCond.wait(mLock); 4059 // if we have been restarted, recordTrack == mActiveTrack.get() here 4060 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4061 mLock.unlock(); 4062 AudioSystem::stopInput(mId); 4063 mLock.lock(); 4064 LOGV("Record stopped OK"); 4065 } 4066 } 4067 } 4068} 4069 4070status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4071{ 4072 const size_t SIZE = 256; 4073 char buffer[SIZE]; 4074 String8 result; 4075 pid_t pid = 0; 4076 4077 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4078 result.append(buffer); 4079 4080 if (mActiveTrack != 0) { 4081 result.append("Active Track:\n"); 4082 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4083 mActiveTrack->dump(buffer, SIZE); 4084 result.append(buffer); 4085 4086 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4087 result.append(buffer); 4088 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4089 result.append(buffer); 4090 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != 0)); 4091 result.append(buffer); 4092 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4093 result.append(buffer); 4094 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4095 result.append(buffer); 4096 4097 4098 } else { 4099 result.append("No record client\n"); 4100 } 4101 write(fd, result.string(), result.size()); 4102 4103 dumpBase(fd, args); 4104 4105 return NO_ERROR; 4106} 4107 4108status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4109{ 4110 size_t framesReq = buffer->frameCount; 4111 size_t framesReady = mFrameCount - mRsmpInIndex; 4112 int channelCount; 4113 4114 if (framesReady == 0) { 4115 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4116 if (mBytesRead < 0) { 4117 LOGE("RecordThread::getNextBuffer() Error reading audio input"); 4118 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4119 // Force input into standby so that it tries to 4120 // recover at next read attempt 4121 mInput->stream->common.standby(&mInput->stream->common); 4122 usleep(5000); 4123 } 4124 buffer->raw = 0; 4125 buffer->frameCount = 0; 4126 return NOT_ENOUGH_DATA; 4127 } 4128 mRsmpInIndex = 0; 4129 framesReady = mFrameCount; 4130 } 4131 4132 if (framesReq > framesReady) { 4133 framesReq = framesReady; 4134 } 4135 4136 if (mChannelCount == 1 && mReqChannelCount == 2) { 4137 channelCount = 1; 4138 } else { 4139 channelCount = 2; 4140 } 4141 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4142 buffer->frameCount = framesReq; 4143 return NO_ERROR; 4144} 4145 4146void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4147{ 4148 mRsmpInIndex += buffer->frameCount; 4149 buffer->frameCount = 0; 4150} 4151 4152bool AudioFlinger::RecordThread::checkForNewParameters_l() 4153{ 4154 bool reconfig = false; 4155 4156 while (!mNewParameters.isEmpty()) { 4157 status_t status = NO_ERROR; 4158 String8 keyValuePair = mNewParameters[0]; 4159 AudioParameter param = AudioParameter(keyValuePair); 4160 int value; 4161 int reqFormat = mFormat; 4162 int reqSamplingRate = mReqSampleRate; 4163 int reqChannelCount = mReqChannelCount; 4164 4165 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4166 reqSamplingRate = value; 4167 reconfig = true; 4168 } 4169 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4170 reqFormat = value; 4171 reconfig = true; 4172 } 4173 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4174 reqChannelCount = popcount(value); 4175 reconfig = true; 4176 } 4177 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4178 // do not accept frame count changes if tracks are open as the track buffer 4179 // size depends on frame count and correct behavior would not be garantied 4180 // if frame count is changed after track creation 4181 if (mActiveTrack != 0) { 4182 status = INVALID_OPERATION; 4183 } else { 4184 reconfig = true; 4185 } 4186 } 4187 if (status == NO_ERROR) { 4188 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4189 if (status == INVALID_OPERATION) { 4190 mInput->stream->common.standby(&mInput->stream->common); 4191 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4192 } 4193 if (reconfig) { 4194 if (status == BAD_VALUE && 4195 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4196 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4197 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4198 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4199 (reqChannelCount < 3)) { 4200 status = NO_ERROR; 4201 } 4202 if (status == NO_ERROR) { 4203 readInputParameters(); 4204 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4205 } 4206 } 4207 } 4208 4209 mNewParameters.removeAt(0); 4210 4211 mParamStatus = status; 4212 mParamCond.signal(); 4213 mWaitWorkCV.wait(mLock); 4214 } 4215 return reconfig; 4216} 4217 4218String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4219{ 4220 char *s; 4221 String8 out_s8; 4222 4223 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4224 out_s8 = String8(s); 4225 free(s); 4226 return out_s8; 4227} 4228 4229void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4230 AudioSystem::OutputDescriptor desc; 4231 void *param2 = 0; 4232 4233 switch (event) { 4234 case AudioSystem::INPUT_OPENED: 4235 case AudioSystem::INPUT_CONFIG_CHANGED: 4236 desc.channels = mChannelMask; 4237 desc.samplingRate = mSampleRate; 4238 desc.format = mFormat; 4239 desc.frameCount = mFrameCount; 4240 desc.latency = 0; 4241 param2 = &desc; 4242 break; 4243 4244 case AudioSystem::INPUT_CLOSED: 4245 default: 4246 break; 4247 } 4248 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4249} 4250 4251void AudioFlinger::RecordThread::readInputParameters() 4252{ 4253 if (mRsmpInBuffer) delete mRsmpInBuffer; 4254 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4255 if (mResampler) delete mResampler; 4256 mResampler = 0; 4257 4258 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4259 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4260 mChannelCount = (uint16_t)popcount(mChannelMask); 4261 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4262 mFrameSize = (uint16_t)audio_stream_frame_size(&mInput->stream->common); 4263 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4264 mFrameCount = mInputBytes / mFrameSize; 4265 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4266 4267 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4268 { 4269 int channelCount; 4270 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4271 // stereo to mono post process as the resampler always outputs stereo. 4272 if (mChannelCount == 1 && mReqChannelCount == 2) { 4273 channelCount = 1; 4274 } else { 4275 channelCount = 2; 4276 } 4277 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4278 mResampler->setSampleRate(mSampleRate); 4279 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4280 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4281 4282 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4283 if (mChannelCount == 1 && mReqChannelCount == 1) { 4284 mFrameCount >>= 1; 4285 } 4286 4287 } 4288 mRsmpInIndex = mFrameCount; 4289} 4290 4291unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4292{ 4293 return mInput->stream->get_input_frames_lost(mInput->stream); 4294} 4295 4296// ---------------------------------------------------------------------------- 4297 4298int AudioFlinger::openOutput(uint32_t *pDevices, 4299 uint32_t *pSamplingRate, 4300 uint32_t *pFormat, 4301 uint32_t *pChannels, 4302 uint32_t *pLatencyMs, 4303 uint32_t flags) 4304{ 4305 status_t status; 4306 PlaybackThread *thread = NULL; 4307 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4308 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4309 uint32_t format = pFormat ? *pFormat : 0; 4310 uint32_t channels = pChannels ? *pChannels : 0; 4311 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4312 audio_stream_out_t *outStream; 4313 audio_hw_device_t *outHwDev; 4314 4315 LOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4316 pDevices ? *pDevices : 0, 4317 samplingRate, 4318 format, 4319 channels, 4320 flags); 4321 4322 if (pDevices == NULL || *pDevices == 0) { 4323 return 0; 4324 } 4325 4326 Mutex::Autolock _l(mLock); 4327 4328 outHwDev = findSuitableHwDev_l(*pDevices); 4329 if (outHwDev == NULL) 4330 return 0; 4331 4332 status = outHwDev->open_output_stream(outHwDev, *pDevices, (int *)&format, 4333 &channels, &samplingRate, &outStream); 4334 LOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4335 outStream, 4336 samplingRate, 4337 format, 4338 channels, 4339 status); 4340 4341 mHardwareStatus = AUDIO_HW_IDLE; 4342 if (outStream != NULL) { 4343 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4344 int id = nextUniqueId_l(); 4345 4346 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4347 (format != AUDIO_FORMAT_PCM_16_BIT) || 4348 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4349 thread = new DirectOutputThread(this, output, id, *pDevices); 4350 LOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4351 } else { 4352 thread = new MixerThread(this, output, id, *pDevices); 4353 LOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4354 } 4355 mPlaybackThreads.add(id, thread); 4356 4357 if (pSamplingRate) *pSamplingRate = samplingRate; 4358 if (pFormat) *pFormat = format; 4359 if (pChannels) *pChannels = channels; 4360 if (pLatencyMs) *pLatencyMs = thread->latency(); 4361 4362 // notify client processes of the new output creation 4363 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4364 return id; 4365 } 4366 4367 return 0; 4368} 4369 4370int AudioFlinger::openDuplicateOutput(int output1, int output2) 4371{ 4372 Mutex::Autolock _l(mLock); 4373 MixerThread *thread1 = checkMixerThread_l(output1); 4374 MixerThread *thread2 = checkMixerThread_l(output2); 4375 4376 if (thread1 == NULL || thread2 == NULL) { 4377 LOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 4378 return 0; 4379 } 4380 4381 int id = nextUniqueId_l(); 4382 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 4383 thread->addOutputTrack(thread2); 4384 mPlaybackThreads.add(id, thread); 4385 // notify client processes of the new output creation 4386 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4387 return id; 4388} 4389 4390status_t AudioFlinger::closeOutput(int output) 4391{ 4392 // keep strong reference on the playback thread so that 4393 // it is not destroyed while exit() is executed 4394 sp <PlaybackThread> thread; 4395 { 4396 Mutex::Autolock _l(mLock); 4397 thread = checkPlaybackThread_l(output); 4398 if (thread == NULL) { 4399 return BAD_VALUE; 4400 } 4401 4402 LOGV("closeOutput() %d", output); 4403 4404 if (thread->type() == PlaybackThread::MIXER) { 4405 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4406 if (mPlaybackThreads.valueAt(i)->type() == PlaybackThread::DUPLICATING) { 4407 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 4408 dupThread->removeOutputTrack((MixerThread *)thread.get()); 4409 } 4410 } 4411 } 4412 void *param2 = 0; 4413 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 4414 mPlaybackThreads.removeItem(output); 4415 } 4416 thread->exit(); 4417 4418 if (thread->type() != PlaybackThread::DUPLICATING) { 4419 AudioStreamOut *out = thread->getOutput(); 4420 out->hwDev->close_output_stream(out->hwDev, out->stream); 4421 delete out; 4422 } 4423 return NO_ERROR; 4424} 4425 4426status_t AudioFlinger::suspendOutput(int output) 4427{ 4428 Mutex::Autolock _l(mLock); 4429 PlaybackThread *thread = checkPlaybackThread_l(output); 4430 4431 if (thread == NULL) { 4432 return BAD_VALUE; 4433 } 4434 4435 LOGV("suspendOutput() %d", output); 4436 thread->suspend(); 4437 4438 return NO_ERROR; 4439} 4440 4441status_t AudioFlinger::restoreOutput(int output) 4442{ 4443 Mutex::Autolock _l(mLock); 4444 PlaybackThread *thread = checkPlaybackThread_l(output); 4445 4446 if (thread == NULL) { 4447 return BAD_VALUE; 4448 } 4449 4450 LOGV("restoreOutput() %d", output); 4451 4452 thread->restore(); 4453 4454 return NO_ERROR; 4455} 4456 4457int AudioFlinger::openInput(uint32_t *pDevices, 4458 uint32_t *pSamplingRate, 4459 uint32_t *pFormat, 4460 uint32_t *pChannels, 4461 uint32_t acoustics) 4462{ 4463 status_t status; 4464 RecordThread *thread = NULL; 4465 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4466 uint32_t format = pFormat ? *pFormat : 0; 4467 uint32_t channels = pChannels ? *pChannels : 0; 4468 uint32_t reqSamplingRate = samplingRate; 4469 uint32_t reqFormat = format; 4470 uint32_t reqChannels = channels; 4471 audio_stream_in_t *inStream; 4472 audio_hw_device_t *inHwDev; 4473 4474 if (pDevices == NULL || *pDevices == 0) { 4475 return 0; 4476 } 4477 4478 Mutex::Autolock _l(mLock); 4479 4480 inHwDev = findSuitableHwDev_l(*pDevices); 4481 if (inHwDev == NULL) 4482 return 0; 4483 4484 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 4485 &channels, &samplingRate, 4486 (audio_in_acoustics_t)acoustics, 4487 &inStream); 4488 LOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 4489 inStream, 4490 samplingRate, 4491 format, 4492 channels, 4493 acoustics, 4494 status); 4495 4496 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 4497 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 4498 // or stereo to mono conversions on 16 bit PCM inputs. 4499 if (inStream == NULL && status == BAD_VALUE && 4500 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 4501 (samplingRate <= 2 * reqSamplingRate) && 4502 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 4503 LOGV("openInput() reopening with proposed sampling rate and channels"); 4504 status = inHwDev->open_input_stream(inHwDev, *pDevices, (int *)&format, 4505 &channels, &samplingRate, 4506 (audio_in_acoustics_t)acoustics, 4507 &inStream); 4508 } 4509 4510 if (inStream != NULL) { 4511 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 4512 4513 int id = nextUniqueId_l(); 4514 // Start record thread 4515 thread = new RecordThread(this, input, reqSamplingRate, reqChannels, id); 4516 mRecordThreads.add(id, thread); 4517 LOGV("openInput() created record thread: ID %d thread %p", id, thread); 4518 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 4519 if (pFormat) *pFormat = format; 4520 if (pChannels) *pChannels = reqChannels; 4521 4522 input->stream->common.standby(&input->stream->common); 4523 4524 // notify client processes of the new input creation 4525 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 4526 return id; 4527 } 4528 4529 return 0; 4530} 4531 4532status_t AudioFlinger::closeInput(int input) 4533{ 4534 // keep strong reference on the record thread so that 4535 // it is not destroyed while exit() is executed 4536 sp <RecordThread> thread; 4537 { 4538 Mutex::Autolock _l(mLock); 4539 thread = checkRecordThread_l(input); 4540 if (thread == NULL) { 4541 return BAD_VALUE; 4542 } 4543 4544 LOGV("closeInput() %d", input); 4545 void *param2 = 0; 4546 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 4547 mRecordThreads.removeItem(input); 4548 } 4549 thread->exit(); 4550 4551 AudioStreamIn *in = thread->getInput(); 4552 in->hwDev->close_input_stream(in->hwDev, in->stream); 4553 delete in; 4554 4555 return NO_ERROR; 4556} 4557 4558status_t AudioFlinger::setStreamOutput(uint32_t stream, int output) 4559{ 4560 Mutex::Autolock _l(mLock); 4561 MixerThread *dstThread = checkMixerThread_l(output); 4562 if (dstThread == NULL) { 4563 LOGW("setStreamOutput() bad output id %d", output); 4564 return BAD_VALUE; 4565 } 4566 4567 LOGV("setStreamOutput() stream %d to output %d", stream, output); 4568 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 4569 4570 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4571 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 4572 if (thread != dstThread && 4573 thread->type() != PlaybackThread::DIRECT) { 4574 MixerThread *srcThread = (MixerThread *)thread; 4575 srcThread->invalidateTracks(stream); 4576 } 4577 } 4578 4579 return NO_ERROR; 4580} 4581 4582 4583int AudioFlinger::newAudioSessionId() 4584{ 4585 AutoMutex _l(mLock); 4586 return nextUniqueId_l(); 4587} 4588 4589// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 4590AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 4591{ 4592 PlaybackThread *thread = NULL; 4593 if (mPlaybackThreads.indexOfKey(output) >= 0) { 4594 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 4595 } 4596 return thread; 4597} 4598 4599// checkMixerThread_l() must be called with AudioFlinger::mLock held 4600AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 4601{ 4602 PlaybackThread *thread = checkPlaybackThread_l(output); 4603 if (thread != NULL) { 4604 if (thread->type() == PlaybackThread::DIRECT) { 4605 thread = NULL; 4606 } 4607 } 4608 return (MixerThread *)thread; 4609} 4610 4611// checkRecordThread_l() must be called with AudioFlinger::mLock held 4612AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 4613{ 4614 RecordThread *thread = NULL; 4615 if (mRecordThreads.indexOfKey(input) >= 0) { 4616 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 4617 } 4618 return thread; 4619} 4620 4621// nextUniqueId_l() must be called with AudioFlinger::mLock held 4622int AudioFlinger::nextUniqueId_l() 4623{ 4624 return mNextUniqueId++; 4625} 4626 4627// ---------------------------------------------------------------------------- 4628// Effect management 4629// ---------------------------------------------------------------------------- 4630 4631 4632status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 4633{ 4634 Mutex::Autolock _l(mLock); 4635 return EffectQueryNumberEffects(numEffects); 4636} 4637 4638status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 4639{ 4640 Mutex::Autolock _l(mLock); 4641 return EffectQueryEffect(index, descriptor); 4642} 4643 4644status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 4645{ 4646 Mutex::Autolock _l(mLock); 4647 return EffectGetDescriptor(pUuid, descriptor); 4648} 4649 4650 4651// this UUID must match the one defined in media/libeffects/EffectVisualizer.cpp 4652static const effect_uuid_t VISUALIZATION_UUID_ = 4653 {0xd069d9e0, 0x8329, 0x11df, 0x9168, {0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b}}; 4654 4655sp<IEffect> AudioFlinger::createEffect(pid_t pid, 4656 effect_descriptor_t *pDesc, 4657 const sp<IEffectClient>& effectClient, 4658 int32_t priority, 4659 int output, 4660 int sessionId, 4661 status_t *status, 4662 int *id, 4663 int *enabled) 4664{ 4665 status_t lStatus = NO_ERROR; 4666 sp<EffectHandle> handle; 4667 effect_descriptor_t desc; 4668 sp<Client> client; 4669 wp<Client> wclient; 4670 4671 LOGV("createEffect pid %d, client %p, priority %d, sessionId %d, output %d", 4672 pid, effectClient.get(), priority, sessionId, output); 4673 4674 if (pDesc == NULL) { 4675 lStatus = BAD_VALUE; 4676 goto Exit; 4677 } 4678 4679 // check audio settings permission for global effects 4680 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 4681 lStatus = PERMISSION_DENIED; 4682 goto Exit; 4683 } 4684 4685 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 4686 // that can only be created by audio policy manager (running in same process) 4687 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 4688 lStatus = PERMISSION_DENIED; 4689 goto Exit; 4690 } 4691 4692 // check recording permission for visualizer 4693 if ((memcmp(&pDesc->type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0 || 4694 memcmp(&pDesc->uuid, &VISUALIZATION_UUID_, sizeof(effect_uuid_t)) == 0) && 4695 !recordingAllowed()) { 4696 lStatus = PERMISSION_DENIED; 4697 goto Exit; 4698 } 4699 4700 if (output == 0) { 4701 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 4702 // output must be specified by AudioPolicyManager when using session 4703 // AUDIO_SESSION_OUTPUT_STAGE 4704 lStatus = BAD_VALUE; 4705 goto Exit; 4706 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 4707 // if the output returned by getOutputForEffect() is removed before we lock the 4708 // mutex below, the call to checkPlaybackThread_l(output) below will detect it 4709 // and we will exit safely 4710 output = AudioSystem::getOutputForEffect(&desc); 4711 } 4712 } 4713 4714 { 4715 Mutex::Autolock _l(mLock); 4716 4717 4718 if (!EffectIsNullUuid(&pDesc->uuid)) { 4719 // if uuid is specified, request effect descriptor 4720 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 4721 if (lStatus < 0) { 4722 LOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 4723 goto Exit; 4724 } 4725 } else { 4726 // if uuid is not specified, look for an available implementation 4727 // of the required type in effect factory 4728 if (EffectIsNullUuid(&pDesc->type)) { 4729 LOGW("createEffect() no effect type"); 4730 lStatus = BAD_VALUE; 4731 goto Exit; 4732 } 4733 uint32_t numEffects = 0; 4734 effect_descriptor_t d; 4735 bool found = false; 4736 4737 lStatus = EffectQueryNumberEffects(&numEffects); 4738 if (lStatus < 0) { 4739 LOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 4740 goto Exit; 4741 } 4742 for (uint32_t i = 0; i < numEffects; i++) { 4743 lStatus = EffectQueryEffect(i, &desc); 4744 if (lStatus < 0) { 4745 LOGW("createEffect() error %d from EffectQueryEffect", lStatus); 4746 continue; 4747 } 4748 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 4749 // If matching type found save effect descriptor. If the session is 4750 // 0 and the effect is not auxiliary, continue enumeration in case 4751 // an auxiliary version of this effect type is available 4752 found = true; 4753 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 4754 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 4755 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4756 break; 4757 } 4758 } 4759 } 4760 if (!found) { 4761 lStatus = BAD_VALUE; 4762 LOGW("createEffect() effect not found"); 4763 goto Exit; 4764 } 4765 // For same effect type, chose auxiliary version over insert version if 4766 // connect to output mix (Compliance to OpenSL ES) 4767 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 4768 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 4769 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 4770 } 4771 } 4772 4773 // Do not allow auxiliary effects on a session different from 0 (output mix) 4774 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 4775 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 4776 lStatus = INVALID_OPERATION; 4777 goto Exit; 4778 } 4779 4780 // return effect descriptor 4781 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 4782 4783 // If output is not specified try to find a matching audio session ID in one of the 4784 // output threads. 4785 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 4786 // because of code checking output when entering the function. 4787 if (output == 0) { 4788 // look for the thread where the specified audio session is present 4789 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 4790 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 4791 output = mPlaybackThreads.keyAt(i); 4792 break; 4793 } 4794 } 4795 // If no output thread contains the requested session ID, default to 4796 // first output. The effect chain will be moved to the correct output 4797 // thread when a track with the same session ID is created 4798 if (output == 0 && mPlaybackThreads.size()) { 4799 output = mPlaybackThreads.keyAt(0); 4800 } 4801 } 4802 LOGV("createEffect() got output %d for effect %s", output, desc.name); 4803 PlaybackThread *thread = checkPlaybackThread_l(output); 4804 if (thread == NULL) { 4805 LOGE("createEffect() unknown output thread"); 4806 lStatus = BAD_VALUE; 4807 goto Exit; 4808 } 4809 4810 // TODO: allow attachment of effect to inputs 4811 4812 wclient = mClients.valueFor(pid); 4813 4814 if (wclient != NULL) { 4815 client = wclient.promote(); 4816 } else { 4817 client = new Client(this, pid); 4818 mClients.add(pid, client); 4819 } 4820 4821 // create effect on selected output trhead 4822 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 4823 &desc, enabled, &lStatus); 4824 if (handle != 0 && id != NULL) { 4825 *id = handle->id(); 4826 } 4827 } 4828 4829Exit: 4830 if(status) { 4831 *status = lStatus; 4832 } 4833 return handle; 4834} 4835 4836status_t AudioFlinger::moveEffects(int session, int srcOutput, int dstOutput) 4837{ 4838 LOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 4839 session, srcOutput, dstOutput); 4840 Mutex::Autolock _l(mLock); 4841 if (srcOutput == dstOutput) { 4842 LOGW("moveEffects() same dst and src outputs %d", dstOutput); 4843 return NO_ERROR; 4844 } 4845 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 4846 if (srcThread == NULL) { 4847 LOGW("moveEffects() bad srcOutput %d", srcOutput); 4848 return BAD_VALUE; 4849 } 4850 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 4851 if (dstThread == NULL) { 4852 LOGW("moveEffects() bad dstOutput %d", dstOutput); 4853 return BAD_VALUE; 4854 } 4855 4856 Mutex::Autolock _dl(dstThread->mLock); 4857 Mutex::Autolock _sl(srcThread->mLock); 4858 moveEffectChain_l(session, srcThread, dstThread, false); 4859 4860 return NO_ERROR; 4861} 4862 4863// moveEffectChain_l mustbe called with both srcThread and dstThread mLocks held 4864status_t AudioFlinger::moveEffectChain_l(int session, 4865 AudioFlinger::PlaybackThread *srcThread, 4866 AudioFlinger::PlaybackThread *dstThread, 4867 bool reRegister) 4868{ 4869 LOGV("moveEffectChain_l() session %d from thread %p to thread %p", 4870 session, srcThread, dstThread); 4871 4872 sp<EffectChain> chain = srcThread->getEffectChain_l(session); 4873 if (chain == 0) { 4874 LOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 4875 session, srcThread); 4876 return INVALID_OPERATION; 4877 } 4878 4879 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 4880 // so that a new chain is created with correct parameters when first effect is added. This is 4881 // otherwise unecessary as removeEffect_l() will remove the chain when last effect is 4882 // removed. 4883 srcThread->removeEffectChain_l(chain); 4884 4885 // transfer all effects one by one so that new effect chain is created on new thread with 4886 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 4887 int dstOutput = dstThread->id(); 4888 sp<EffectChain> dstChain; 4889 uint32_t strategy; 4890 sp<EffectModule> effect = chain->getEffectFromId_l(0); 4891 while (effect != 0) { 4892 srcThread->removeEffect_l(effect); 4893 dstThread->addEffect_l(effect); 4894 // if the move request is not received from audio policy manager, the effect must be 4895 // re-registered with the new strategy and output 4896 if (dstChain == 0) { 4897 dstChain = effect->chain().promote(); 4898 if (dstChain == 0) { 4899 LOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 4900 srcThread->addEffect_l(effect); 4901 return NO_INIT; 4902 } 4903 strategy = dstChain->strategy(); 4904 } 4905 if (reRegister) { 4906 AudioSystem::unregisterEffect(effect->id()); 4907 AudioSystem::registerEffect(&effect->desc(), 4908 dstOutput, 4909 strategy, 4910 session, 4911 effect->id()); 4912 } 4913 effect = chain->getEffectFromId_l(0); 4914 } 4915 4916 return NO_ERROR; 4917} 4918 4919// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 4920sp<AudioFlinger::EffectHandle> AudioFlinger::PlaybackThread::createEffect_l( 4921 const sp<AudioFlinger::Client>& client, 4922 const sp<IEffectClient>& effectClient, 4923 int32_t priority, 4924 int sessionId, 4925 effect_descriptor_t *desc, 4926 int *enabled, 4927 status_t *status 4928 ) 4929{ 4930 sp<EffectModule> effect; 4931 sp<EffectHandle> handle; 4932 status_t lStatus; 4933 sp<Track> track; 4934 sp<EffectChain> chain; 4935 bool chainCreated = false; 4936 bool effectCreated = false; 4937 bool effectRegistered = false; 4938 4939 if (mOutput == 0) { 4940 LOGW("createEffect_l() Audio driver not initialized."); 4941 lStatus = NO_INIT; 4942 goto Exit; 4943 } 4944 4945 // Do not allow auxiliary effect on session other than 0 4946 if ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY && 4947 sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4948 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4949 desc->name, sessionId); 4950 lStatus = BAD_VALUE; 4951 goto Exit; 4952 } 4953 4954 // Do not allow effects with session ID 0 on direct output or duplicating threads 4955 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 4956 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 4957 LOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 4958 desc->name, sessionId); 4959 lStatus = BAD_VALUE; 4960 goto Exit; 4961 } 4962 4963 LOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 4964 4965 { // scope for mLock 4966 Mutex::Autolock _l(mLock); 4967 4968 // check for existing effect chain with the requested audio session 4969 chain = getEffectChain_l(sessionId); 4970 if (chain == 0) { 4971 // create a new chain for this session 4972 LOGV("createEffect_l() new effect chain for session %d", sessionId); 4973 chain = new EffectChain(this, sessionId); 4974 addEffectChain_l(chain); 4975 chain->setStrategy(getStrategyForSession_l(sessionId)); 4976 chainCreated = true; 4977 } else { 4978 effect = chain->getEffectFromDesc_l(desc); 4979 } 4980 4981 LOGV("createEffect_l() got effect %p on chain %p", effect == 0 ? 0 : effect.get(), chain.get()); 4982 4983 if (effect == 0) { 4984 int id = mAudioFlinger->nextUniqueId_l(); 4985 // Check CPU and memory usage 4986 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 4987 if (lStatus != NO_ERROR) { 4988 goto Exit; 4989 } 4990 effectRegistered = true; 4991 // create a new effect module if none present in the chain 4992 effect = new EffectModule(this, chain, desc, id, sessionId); 4993 lStatus = effect->status(); 4994 if (lStatus != NO_ERROR) { 4995 goto Exit; 4996 } 4997 lStatus = chain->addEffect_l(effect); 4998 if (lStatus != NO_ERROR) { 4999 goto Exit; 5000 } 5001 effectCreated = true; 5002 5003 effect->setDevice(mDevice); 5004 effect->setMode(mAudioFlinger->getMode()); 5005 } 5006 // create effect handle and connect it to effect module 5007 handle = new EffectHandle(effect, client, effectClient, priority); 5008 lStatus = effect->addHandle(handle); 5009 if (enabled) { 5010 *enabled = (int)effect->isEnabled(); 5011 } 5012 } 5013 5014Exit: 5015 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5016 Mutex::Autolock _l(mLock); 5017 if (effectCreated) { 5018 chain->removeEffect_l(effect); 5019 } 5020 if (effectRegistered) { 5021 AudioSystem::unregisterEffect(effect->id()); 5022 } 5023 if (chainCreated) { 5024 removeEffectChain_l(chain); 5025 } 5026 handle.clear(); 5027 } 5028 5029 if(status) { 5030 *status = lStatus; 5031 } 5032 return handle; 5033} 5034 5035// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5036// PlaybackThread::mLock held 5037status_t AudioFlinger::PlaybackThread::addEffect_l(const sp<EffectModule>& effect) 5038{ 5039 // check for existing effect chain with the requested audio session 5040 int sessionId = effect->sessionId(); 5041 sp<EffectChain> chain = getEffectChain_l(sessionId); 5042 bool chainCreated = false; 5043 5044 if (chain == 0) { 5045 // create a new chain for this session 5046 LOGV("addEffect_l() new effect chain for session %d", sessionId); 5047 chain = new EffectChain(this, sessionId); 5048 addEffectChain_l(chain); 5049 chain->setStrategy(getStrategyForSession_l(sessionId)); 5050 chainCreated = true; 5051 } 5052 LOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5053 5054 if (chain->getEffectFromId_l(effect->id()) != 0) { 5055 LOGW("addEffect_l() %p effect %s already present in chain %p", 5056 this, effect->desc().name, chain.get()); 5057 return BAD_VALUE; 5058 } 5059 5060 status_t status = chain->addEffect_l(effect); 5061 if (status != NO_ERROR) { 5062 if (chainCreated) { 5063 removeEffectChain_l(chain); 5064 } 5065 return status; 5066 } 5067 5068 effect->setDevice(mDevice); 5069 effect->setMode(mAudioFlinger->getMode()); 5070 return NO_ERROR; 5071} 5072 5073void AudioFlinger::PlaybackThread::removeEffect_l(const sp<EffectModule>& effect) { 5074 5075 LOGV("removeEffect_l() %p effect %p", this, effect.get()); 5076 effect_descriptor_t desc = effect->desc(); 5077 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5078 detachAuxEffect_l(effect->id()); 5079 } 5080 5081 sp<EffectChain> chain = effect->chain().promote(); 5082 if (chain != 0) { 5083 // remove effect chain if removing last effect 5084 if (chain->removeEffect_l(effect) == 0) { 5085 removeEffectChain_l(chain); 5086 } 5087 } else { 5088 LOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5089 } 5090} 5091 5092void AudioFlinger::PlaybackThread::disconnectEffect(const sp<EffectModule>& effect, 5093 const wp<EffectHandle>& handle) { 5094 Mutex::Autolock _l(mLock); 5095 LOGV("disconnectEffect() %p effect %p", this, effect.get()); 5096 // delete the effect module if removing last handle on it 5097 if (effect->removeHandle(handle) == 0) { 5098 removeEffect_l(effect); 5099 AudioSystem::unregisterEffect(effect->id()); 5100 } 5101} 5102 5103status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5104{ 5105 int session = chain->sessionId(); 5106 int16_t *buffer = mMixBuffer; 5107 bool ownsBuffer = false; 5108 5109 LOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5110 if (session > 0) { 5111 // Only one effect chain can be present in direct output thread and it uses 5112 // the mix buffer as input 5113 if (mType != DIRECT) { 5114 size_t numSamples = mFrameCount * mChannelCount; 5115 buffer = new int16_t[numSamples]; 5116 memset(buffer, 0, numSamples * sizeof(int16_t)); 5117 LOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5118 ownsBuffer = true; 5119 } 5120 5121 // Attach all tracks with same session ID to this chain. 5122 for (size_t i = 0; i < mTracks.size(); ++i) { 5123 sp<Track> track = mTracks[i]; 5124 if (session == track->sessionId()) { 5125 LOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5126 track->setMainBuffer(buffer); 5127 chain->incTrackCnt(); 5128 } 5129 } 5130 5131 // indicate all active tracks in the chain 5132 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5133 sp<Track> track = mActiveTracks[i].promote(); 5134 if (track == 0) continue; 5135 if (session == track->sessionId()) { 5136 LOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5137 chain->incActiveTrackCnt(); 5138 } 5139 } 5140 } 5141 5142 chain->setInBuffer(buffer, ownsBuffer); 5143 chain->setOutBuffer(mMixBuffer); 5144 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5145 // chains list in order to be processed last as it contains output stage effects 5146 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5147 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5148 // after track specific effects and before output stage 5149 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5150 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5151 // Effect chain for other sessions are inserted at beginning of effect 5152 // chains list to be processed before output mix effects. Relative order between other 5153 // sessions is not important 5154 size_t size = mEffectChains.size(); 5155 size_t i = 0; 5156 for (i = 0; i < size; i++) { 5157 if (mEffectChains[i]->sessionId() < session) break; 5158 } 5159 mEffectChains.insertAt(chain, i); 5160 5161 return NO_ERROR; 5162} 5163 5164size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 5165{ 5166 int session = chain->sessionId(); 5167 5168 LOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 5169 5170 for (size_t i = 0; i < mEffectChains.size(); i++) { 5171 if (chain == mEffectChains[i]) { 5172 mEffectChains.removeAt(i); 5173 // detach all active tracks from the chain 5174 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5175 sp<Track> track = mActiveTracks[i].promote(); 5176 if (track == 0) continue; 5177 if (session == track->sessionId()) { 5178 LOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 5179 chain.get(), session); 5180 chain->decActiveTrackCnt(); 5181 } 5182 } 5183 5184 // detach all tracks with same session ID from this chain 5185 for (size_t i = 0; i < mTracks.size(); ++i) { 5186 sp<Track> track = mTracks[i]; 5187 if (session == track->sessionId()) { 5188 track->setMainBuffer(mMixBuffer); 5189 chain->decTrackCnt(); 5190 } 5191 } 5192 break; 5193 } 5194 } 5195 return mEffectChains.size(); 5196} 5197 5198void AudioFlinger::PlaybackThread::lockEffectChains_l( 5199 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5200{ 5201 effectChains = mEffectChains; 5202 for (size_t i = 0; i < mEffectChains.size(); i++) { 5203 mEffectChains[i]->lock(); 5204 } 5205} 5206 5207void AudioFlinger::PlaybackThread::unlockEffectChains( 5208 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5209{ 5210 for (size_t i = 0; i < effectChains.size(); i++) { 5211 effectChains[i]->unlock(); 5212 } 5213} 5214 5215 5216sp<AudioFlinger::EffectModule> AudioFlinger::PlaybackThread::getEffect_l(int sessionId, int effectId) 5217{ 5218 sp<EffectModule> effect; 5219 5220 sp<EffectChain> chain = getEffectChain_l(sessionId); 5221 if (chain != 0) { 5222 effect = chain->getEffectFromId_l(effectId); 5223 } 5224 return effect; 5225} 5226 5227status_t AudioFlinger::PlaybackThread::attachAuxEffect( 5228 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5229{ 5230 Mutex::Autolock _l(mLock); 5231 return attachAuxEffect_l(track, EffectId); 5232} 5233 5234status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 5235 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 5236{ 5237 status_t status = NO_ERROR; 5238 5239 if (EffectId == 0) { 5240 track->setAuxBuffer(0, NULL); 5241 } else { 5242 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 5243 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 5244 if (effect != 0) { 5245 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5246 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 5247 } else { 5248 status = INVALID_OPERATION; 5249 } 5250 } else { 5251 status = BAD_VALUE; 5252 } 5253 } 5254 return status; 5255} 5256 5257void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 5258{ 5259 for (size_t i = 0; i < mTracks.size(); ++i) { 5260 sp<Track> track = mTracks[i]; 5261 if (track->auxEffectId() == effectId) { 5262 attachAuxEffect_l(track, 0); 5263 } 5264 } 5265} 5266 5267// ---------------------------------------------------------------------------- 5268// EffectModule implementation 5269// ---------------------------------------------------------------------------- 5270 5271#undef LOG_TAG 5272#define LOG_TAG "AudioFlinger::EffectModule" 5273 5274AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 5275 const wp<AudioFlinger::EffectChain>& chain, 5276 effect_descriptor_t *desc, 5277 int id, 5278 int sessionId) 5279 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 5280 mStatus(NO_INIT), mState(IDLE) 5281{ 5282 LOGV("Constructor %p", this); 5283 int lStatus; 5284 sp<ThreadBase> thread = mThread.promote(); 5285 if (thread == 0) { 5286 return; 5287 } 5288 PlaybackThread *p = (PlaybackThread *)thread.get(); 5289 5290 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 5291 5292 // create effect engine from effect factory 5293 mStatus = EffectCreate(&desc->uuid, sessionId, p->id(), &mEffectInterface); 5294 5295 if (mStatus != NO_ERROR) { 5296 return; 5297 } 5298 lStatus = init(); 5299 if (lStatus < 0) { 5300 mStatus = lStatus; 5301 goto Error; 5302 } 5303 5304 LOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 5305 return; 5306Error: 5307 EffectRelease(mEffectInterface); 5308 mEffectInterface = NULL; 5309 LOGV("Constructor Error %d", mStatus); 5310} 5311 5312AudioFlinger::EffectModule::~EffectModule() 5313{ 5314 LOGV("Destructor %p", this); 5315 if (mEffectInterface != NULL) { 5316 // release effect engine 5317 EffectRelease(mEffectInterface); 5318 } 5319} 5320 5321status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 5322{ 5323 status_t status; 5324 5325 Mutex::Autolock _l(mLock); 5326 // First handle in mHandles has highest priority and controls the effect module 5327 int priority = handle->priority(); 5328 size_t size = mHandles.size(); 5329 sp<EffectHandle> h; 5330 size_t i; 5331 for (i = 0; i < size; i++) { 5332 h = mHandles[i].promote(); 5333 if (h == 0) continue; 5334 if (h->priority() <= priority) break; 5335 } 5336 // if inserted in first place, move effect control from previous owner to this handle 5337 if (i == 0) { 5338 if (h != 0) { 5339 h->setControl(false, true); 5340 } 5341 handle->setControl(true, false); 5342 status = NO_ERROR; 5343 } else { 5344 status = ALREADY_EXISTS; 5345 } 5346 mHandles.insertAt(handle, i); 5347 return status; 5348} 5349 5350size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 5351{ 5352 Mutex::Autolock _l(mLock); 5353 size_t size = mHandles.size(); 5354 size_t i; 5355 for (i = 0; i < size; i++) { 5356 if (mHandles[i] == handle) break; 5357 } 5358 if (i == size) { 5359 return size; 5360 } 5361 mHandles.removeAt(i); 5362 size = mHandles.size(); 5363 // if removed from first place, move effect control from this handle to next in line 5364 if (i == 0 && size != 0) { 5365 sp<EffectHandle> h = mHandles[0].promote(); 5366 if (h != 0) { 5367 h->setControl(true, true); 5368 } 5369 } 5370 5371 // Release effect engine here so that it is done immediately. Otherwise it will be released 5372 // by the destructor when the last strong reference on the this object is released which can 5373 // happen after next process is called on this effect. 5374 if (size == 0 && mEffectInterface != NULL) { 5375 // release effect engine 5376 EffectRelease(mEffectInterface); 5377 mEffectInterface = NULL; 5378 } 5379 5380 return size; 5381} 5382 5383void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle) 5384{ 5385 // keep a strong reference on this EffectModule to avoid calling the 5386 // destructor before we exit 5387 sp<EffectModule> keep(this); 5388 { 5389 sp<ThreadBase> thread = mThread.promote(); 5390 if (thread != 0) { 5391 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 5392 playbackThread->disconnectEffect(keep, handle); 5393 } 5394 } 5395} 5396 5397void AudioFlinger::EffectModule::updateState() { 5398 Mutex::Autolock _l(mLock); 5399 5400 switch (mState) { 5401 case RESTART: 5402 reset_l(); 5403 // FALL THROUGH 5404 5405 case STARTING: 5406 // clear auxiliary effect input buffer for next accumulation 5407 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5408 memset(mConfig.inputCfg.buffer.raw, 5409 0, 5410 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5411 } 5412 start_l(); 5413 mState = ACTIVE; 5414 break; 5415 case STOPPING: 5416 stop_l(); 5417 mDisableWaitCnt = mMaxDisableWaitCnt; 5418 mState = STOPPED; 5419 break; 5420 case STOPPED: 5421 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 5422 // turn off sequence. 5423 if (--mDisableWaitCnt == 0) { 5424 reset_l(); 5425 mState = IDLE; 5426 } 5427 break; 5428 default: //IDLE , ACTIVE 5429 break; 5430 } 5431} 5432 5433void AudioFlinger::EffectModule::process() 5434{ 5435 Mutex::Autolock _l(mLock); 5436 5437 if (mEffectInterface == NULL || 5438 mConfig.inputCfg.buffer.raw == NULL || 5439 mConfig.outputCfg.buffer.raw == NULL) { 5440 return; 5441 } 5442 5443 if (isProcessEnabled()) { 5444 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 5445 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5446 AudioMixer::ditherAndClamp(mConfig.inputCfg.buffer.s32, 5447 mConfig.inputCfg.buffer.s32, 5448 mConfig.inputCfg.buffer.frameCount/2); 5449 } 5450 5451 // do the actual processing in the effect engine 5452 int ret = (*mEffectInterface)->process(mEffectInterface, 5453 &mConfig.inputCfg.buffer, 5454 &mConfig.outputCfg.buffer); 5455 5456 // force transition to IDLE state when engine is ready 5457 if (mState == STOPPED && ret == -ENODATA) { 5458 mDisableWaitCnt = 1; 5459 } 5460 5461 // clear auxiliary effect input buffer for next accumulation 5462 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5463 memset(mConfig.inputCfg.buffer.raw, 0, 5464 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 5465 } 5466 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 5467 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5468 // If an insert effect is idle and input buffer is different from output buffer, 5469 // accumulate input onto output 5470 sp<EffectChain> chain = mChain.promote(); 5471 if (chain != 0 && chain->activeTrackCnt() != 0) { 5472 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 5473 int16_t *in = mConfig.inputCfg.buffer.s16; 5474 int16_t *out = mConfig.outputCfg.buffer.s16; 5475 for (size_t i = 0; i < frameCnt; i++) { 5476 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 5477 } 5478 } 5479 } 5480} 5481 5482void AudioFlinger::EffectModule::reset_l() 5483{ 5484 if (mEffectInterface == NULL) { 5485 return; 5486 } 5487 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 5488} 5489 5490status_t AudioFlinger::EffectModule::configure() 5491{ 5492 uint32_t channels; 5493 if (mEffectInterface == NULL) { 5494 return NO_INIT; 5495 } 5496 5497 sp<ThreadBase> thread = mThread.promote(); 5498 if (thread == 0) { 5499 return DEAD_OBJECT; 5500 } 5501 5502 // TODO: handle configuration of effects replacing track process 5503 if (thread->channelCount() == 1) { 5504 channels = AUDIO_CHANNEL_OUT_MONO; 5505 } else { 5506 channels = AUDIO_CHANNEL_OUT_STEREO; 5507 } 5508 5509 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5510 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 5511 } else { 5512 mConfig.inputCfg.channels = channels; 5513 } 5514 mConfig.outputCfg.channels = channels; 5515 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 5516 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 5517 mConfig.inputCfg.samplingRate = thread->sampleRate(); 5518 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 5519 mConfig.inputCfg.bufferProvider.cookie = NULL; 5520 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 5521 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 5522 mConfig.outputCfg.bufferProvider.cookie = NULL; 5523 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 5524 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 5525 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 5526 // Insert effect: 5527 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 5528 // always overwrites output buffer: input buffer == output buffer 5529 // - in other sessions: 5530 // last effect in the chain accumulates in output buffer: input buffer != output buffer 5531 // other effect: overwrites output buffer: input buffer == output buffer 5532 // Auxiliary effect: 5533 // accumulates in output buffer: input buffer != output buffer 5534 // Therefore: accumulate <=> input buffer != output buffer 5535 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 5536 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 5537 } else { 5538 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 5539 } 5540 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 5541 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 5542 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 5543 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 5544 5545 LOGV("configure() %p thread %p buffer %p framecount %d", 5546 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 5547 5548 status_t cmdStatus; 5549 uint32_t size = sizeof(int); 5550 status_t status = (*mEffectInterface)->command(mEffectInterface, 5551 EFFECT_CMD_CONFIGURE, 5552 sizeof(effect_config_t), 5553 &mConfig, 5554 &size, 5555 &cmdStatus); 5556 if (status == 0) { 5557 status = cmdStatus; 5558 } 5559 5560 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 5561 (1000 * mConfig.outputCfg.buffer.frameCount); 5562 5563 return status; 5564} 5565 5566status_t AudioFlinger::EffectModule::init() 5567{ 5568 Mutex::Autolock _l(mLock); 5569 if (mEffectInterface == NULL) { 5570 return NO_INIT; 5571 } 5572 status_t cmdStatus; 5573 uint32_t size = sizeof(status_t); 5574 status_t status = (*mEffectInterface)->command(mEffectInterface, 5575 EFFECT_CMD_INIT, 5576 0, 5577 NULL, 5578 &size, 5579 &cmdStatus); 5580 if (status == 0) { 5581 status = cmdStatus; 5582 } 5583 return status; 5584} 5585 5586status_t AudioFlinger::EffectModule::start_l() 5587{ 5588 if (mEffectInterface == NULL) { 5589 return NO_INIT; 5590 } 5591 status_t cmdStatus; 5592 uint32_t size = sizeof(status_t); 5593 status_t status = (*mEffectInterface)->command(mEffectInterface, 5594 EFFECT_CMD_ENABLE, 5595 0, 5596 NULL, 5597 &size, 5598 &cmdStatus); 5599 if (status == 0) { 5600 status = cmdStatus; 5601 } 5602 return status; 5603} 5604 5605status_t AudioFlinger::EffectModule::stop_l() 5606{ 5607 if (mEffectInterface == NULL) { 5608 return NO_INIT; 5609 } 5610 status_t cmdStatus; 5611 uint32_t size = sizeof(status_t); 5612 status_t status = (*mEffectInterface)->command(mEffectInterface, 5613 EFFECT_CMD_DISABLE, 5614 0, 5615 NULL, 5616 &size, 5617 &cmdStatus); 5618 if (status == 0) { 5619 status = cmdStatus; 5620 } 5621 return status; 5622} 5623 5624status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 5625 uint32_t cmdSize, 5626 void *pCmdData, 5627 uint32_t *replySize, 5628 void *pReplyData) 5629{ 5630 Mutex::Autolock _l(mLock); 5631// LOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 5632 5633 if (mEffectInterface == NULL) { 5634 return NO_INIT; 5635 } 5636 status_t status = (*mEffectInterface)->command(mEffectInterface, 5637 cmdCode, 5638 cmdSize, 5639 pCmdData, 5640 replySize, 5641 pReplyData); 5642 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 5643 uint32_t size = (replySize == NULL) ? 0 : *replySize; 5644 for (size_t i = 1; i < mHandles.size(); i++) { 5645 sp<EffectHandle> h = mHandles[i].promote(); 5646 if (h != 0) { 5647 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 5648 } 5649 } 5650 } 5651 return status; 5652} 5653 5654status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 5655{ 5656 Mutex::Autolock _l(mLock); 5657 LOGV("setEnabled %p enabled %d", this, enabled); 5658 5659 if (enabled != isEnabled()) { 5660 switch (mState) { 5661 // going from disabled to enabled 5662 case IDLE: 5663 mState = STARTING; 5664 break; 5665 case STOPPED: 5666 mState = RESTART; 5667 break; 5668 case STOPPING: 5669 mState = ACTIVE; 5670 break; 5671 5672 // going from enabled to disabled 5673 case RESTART: 5674 mState = STOPPED; 5675 break; 5676 case STARTING: 5677 mState = IDLE; 5678 break; 5679 case ACTIVE: 5680 mState = STOPPING; 5681 break; 5682 } 5683 for (size_t i = 1; i < mHandles.size(); i++) { 5684 sp<EffectHandle> h = mHandles[i].promote(); 5685 if (h != 0) { 5686 h->setEnabled(enabled); 5687 } 5688 } 5689 } 5690 return NO_ERROR; 5691} 5692 5693bool AudioFlinger::EffectModule::isEnabled() 5694{ 5695 switch (mState) { 5696 case RESTART: 5697 case STARTING: 5698 case ACTIVE: 5699 return true; 5700 case IDLE: 5701 case STOPPING: 5702 case STOPPED: 5703 default: 5704 return false; 5705 } 5706} 5707 5708bool AudioFlinger::EffectModule::isProcessEnabled() 5709{ 5710 switch (mState) { 5711 case RESTART: 5712 case ACTIVE: 5713 case STOPPING: 5714 case STOPPED: 5715 return true; 5716 case IDLE: 5717 case STARTING: 5718 default: 5719 return false; 5720 } 5721} 5722 5723status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 5724{ 5725 Mutex::Autolock _l(mLock); 5726 status_t status = NO_ERROR; 5727 5728 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 5729 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 5730 if (isProcessEnabled() && 5731 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 5732 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 5733 status_t cmdStatus; 5734 uint32_t volume[2]; 5735 uint32_t *pVolume = NULL; 5736 uint32_t size = sizeof(volume); 5737 volume[0] = *left; 5738 volume[1] = *right; 5739 if (controller) { 5740 pVolume = volume; 5741 } 5742 status = (*mEffectInterface)->command(mEffectInterface, 5743 EFFECT_CMD_SET_VOLUME, 5744 size, 5745 volume, 5746 &size, 5747 pVolume); 5748 if (controller && status == NO_ERROR && size == sizeof(volume)) { 5749 *left = volume[0]; 5750 *right = volume[1]; 5751 } 5752 } 5753 return status; 5754} 5755 5756status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 5757{ 5758 Mutex::Autolock _l(mLock); 5759 status_t status = NO_ERROR; 5760 if ((mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 5761 status_t cmdStatus; 5762 uint32_t size = sizeof(status_t); 5763 status = (*mEffectInterface)->command(mEffectInterface, 5764 EFFECT_CMD_SET_DEVICE, 5765 sizeof(uint32_t), 5766 &device, 5767 &size, 5768 &cmdStatus); 5769 if (status == NO_ERROR) { 5770 status = cmdStatus; 5771 } 5772 } 5773 return status; 5774} 5775 5776status_t AudioFlinger::EffectModule::setMode(uint32_t mode) 5777{ 5778 Mutex::Autolock _l(mLock); 5779 status_t status = NO_ERROR; 5780 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 5781 status_t cmdStatus; 5782 uint32_t size = sizeof(status_t); 5783 status = (*mEffectInterface)->command(mEffectInterface, 5784 EFFECT_CMD_SET_AUDIO_MODE, 5785 sizeof(int), 5786 &mode, 5787 &size, 5788 &cmdStatus); 5789 if (status == NO_ERROR) { 5790 status = cmdStatus; 5791 } 5792 } 5793 return status; 5794} 5795 5796status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 5797{ 5798 const size_t SIZE = 256; 5799 char buffer[SIZE]; 5800 String8 result; 5801 5802 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 5803 result.append(buffer); 5804 5805 bool locked = tryLock(mLock); 5806 // failed to lock - AudioFlinger is probably deadlocked 5807 if (!locked) { 5808 result.append("\t\tCould not lock Fx mutex:\n"); 5809 } 5810 5811 result.append("\t\tSession Status State Engine:\n"); 5812 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 5813 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 5814 result.append(buffer); 5815 5816 result.append("\t\tDescriptor:\n"); 5817 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5818 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 5819 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 5820 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 5821 result.append(buffer); 5822 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 5823 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 5824 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 5825 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 5826 result.append(buffer); 5827 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 5828 mDescriptor.apiVersion, 5829 mDescriptor.flags); 5830 result.append(buffer); 5831 snprintf(buffer, SIZE, "\t\t- name: %s\n", 5832 mDescriptor.name); 5833 result.append(buffer); 5834 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 5835 mDescriptor.implementor); 5836 result.append(buffer); 5837 5838 result.append("\t\t- Input configuration:\n"); 5839 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5840 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5841 (uint32_t)mConfig.inputCfg.buffer.raw, 5842 mConfig.inputCfg.buffer.frameCount, 5843 mConfig.inputCfg.samplingRate, 5844 mConfig.inputCfg.channels, 5845 mConfig.inputCfg.format); 5846 result.append(buffer); 5847 5848 result.append("\t\t- Output configuration:\n"); 5849 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 5850 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 5851 (uint32_t)mConfig.outputCfg.buffer.raw, 5852 mConfig.outputCfg.buffer.frameCount, 5853 mConfig.outputCfg.samplingRate, 5854 mConfig.outputCfg.channels, 5855 mConfig.outputCfg.format); 5856 result.append(buffer); 5857 5858 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 5859 result.append(buffer); 5860 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 5861 for (size_t i = 0; i < mHandles.size(); ++i) { 5862 sp<EffectHandle> handle = mHandles[i].promote(); 5863 if (handle != 0) { 5864 handle->dump(buffer, SIZE); 5865 result.append(buffer); 5866 } 5867 } 5868 5869 result.append("\n"); 5870 5871 write(fd, result.string(), result.length()); 5872 5873 if (locked) { 5874 mLock.unlock(); 5875 } 5876 5877 return NO_ERROR; 5878} 5879 5880// ---------------------------------------------------------------------------- 5881// EffectHandle implementation 5882// ---------------------------------------------------------------------------- 5883 5884#undef LOG_TAG 5885#define LOG_TAG "AudioFlinger::EffectHandle" 5886 5887AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 5888 const sp<AudioFlinger::Client>& client, 5889 const sp<IEffectClient>& effectClient, 5890 int32_t priority) 5891 : BnEffect(), 5892 mEffect(effect), mEffectClient(effectClient), mClient(client), mPriority(priority), mHasControl(false) 5893{ 5894 LOGV("constructor %p", this); 5895 5896 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 5897 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 5898 if (mCblkMemory != 0) { 5899 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 5900 5901 if (mCblk) { 5902 new(mCblk) effect_param_cblk_t(); 5903 mBuffer = (uint8_t *)mCblk + bufOffset; 5904 } 5905 } else { 5906 LOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 5907 return; 5908 } 5909} 5910 5911AudioFlinger::EffectHandle::~EffectHandle() 5912{ 5913 LOGV("Destructor %p", this); 5914 disconnect(); 5915} 5916 5917status_t AudioFlinger::EffectHandle::enable() 5918{ 5919 if (!mHasControl) return INVALID_OPERATION; 5920 if (mEffect == 0) return DEAD_OBJECT; 5921 5922 return mEffect->setEnabled(true); 5923} 5924 5925status_t AudioFlinger::EffectHandle::disable() 5926{ 5927 if (!mHasControl) return INVALID_OPERATION; 5928 if (mEffect == NULL) return DEAD_OBJECT; 5929 5930 return mEffect->setEnabled(false); 5931} 5932 5933void AudioFlinger::EffectHandle::disconnect() 5934{ 5935 if (mEffect == 0) { 5936 return; 5937 } 5938 mEffect->disconnect(this); 5939 // release sp on module => module destructor can be called now 5940 mEffect.clear(); 5941 if (mCblk) { 5942 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 5943 } 5944 mCblkMemory.clear(); // and free the shared memory 5945 if (mClient != 0) { 5946 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 5947 mClient.clear(); 5948 } 5949} 5950 5951status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 5952 uint32_t cmdSize, 5953 void *pCmdData, 5954 uint32_t *replySize, 5955 void *pReplyData) 5956{ 5957// LOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 5958// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 5959 5960 // only get parameter command is permitted for applications not controlling the effect 5961 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 5962 return INVALID_OPERATION; 5963 } 5964 if (mEffect == 0) return DEAD_OBJECT; 5965 5966 // handle commands that are not forwarded transparently to effect engine 5967 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 5968 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 5969 // no risk to block the whole media server process or mixer threads is we are stuck here 5970 Mutex::Autolock _l(mCblk->lock); 5971 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 5972 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 5973 mCblk->serverIndex = 0; 5974 mCblk->clientIndex = 0; 5975 return BAD_VALUE; 5976 } 5977 status_t status = NO_ERROR; 5978 while (mCblk->serverIndex < mCblk->clientIndex) { 5979 int reply; 5980 uint32_t rsize = sizeof(int); 5981 int *p = (int *)(mBuffer + mCblk->serverIndex); 5982 int size = *p++; 5983 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 5984 LOGW("command(): invalid parameter block size"); 5985 break; 5986 } 5987 effect_param_t *param = (effect_param_t *)p; 5988 if (param->psize == 0 || param->vsize == 0) { 5989 LOGW("command(): null parameter or value size"); 5990 mCblk->serverIndex += size; 5991 continue; 5992 } 5993 uint32_t psize = sizeof(effect_param_t) + 5994 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 5995 param->vsize; 5996 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 5997 psize, 5998 p, 5999 &rsize, 6000 &reply); 6001 // stop at first error encountered 6002 if (ret != NO_ERROR) { 6003 status = ret; 6004 *(int *)pReplyData = reply; 6005 break; 6006 } else if (reply != NO_ERROR) { 6007 *(int *)pReplyData = reply; 6008 break; 6009 } 6010 mCblk->serverIndex += size; 6011 } 6012 mCblk->serverIndex = 0; 6013 mCblk->clientIndex = 0; 6014 return status; 6015 } else if (cmdCode == EFFECT_CMD_ENABLE) { 6016 *(int *)pReplyData = NO_ERROR; 6017 return enable(); 6018 } else if (cmdCode == EFFECT_CMD_DISABLE) { 6019 *(int *)pReplyData = NO_ERROR; 6020 return disable(); 6021 } 6022 6023 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6024} 6025 6026sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 6027 return mCblkMemory; 6028} 6029 6030void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal) 6031{ 6032 LOGV("setControl %p control %d", this, hasControl); 6033 6034 mHasControl = hasControl; 6035 if (signal && mEffectClient != 0) { 6036 mEffectClient->controlStatusChanged(hasControl); 6037 } 6038} 6039 6040void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 6041 uint32_t cmdSize, 6042 void *pCmdData, 6043 uint32_t replySize, 6044 void *pReplyData) 6045{ 6046 if (mEffectClient != 0) { 6047 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 6048 } 6049} 6050 6051 6052 6053void AudioFlinger::EffectHandle::setEnabled(bool enabled) 6054{ 6055 if (mEffectClient != 0) { 6056 mEffectClient->enableStatusChanged(enabled); 6057 } 6058} 6059 6060status_t AudioFlinger::EffectHandle::onTransact( 6061 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6062{ 6063 return BnEffect::onTransact(code, data, reply, flags); 6064} 6065 6066 6067void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 6068{ 6069 bool locked = tryLock(mCblk->lock); 6070 6071 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 6072 (mClient == NULL) ? getpid() : mClient->pid(), 6073 mPriority, 6074 mHasControl, 6075 !locked, 6076 mCblk->clientIndex, 6077 mCblk->serverIndex 6078 ); 6079 6080 if (locked) { 6081 mCblk->lock.unlock(); 6082 } 6083} 6084 6085#undef LOG_TAG 6086#define LOG_TAG "AudioFlinger::EffectChain" 6087 6088AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 6089 int sessionId) 6090 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), 6091 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 6092 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 6093{ 6094 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 6095} 6096 6097AudioFlinger::EffectChain::~EffectChain() 6098{ 6099 if (mOwnInBuffer) { 6100 delete mInBuffer; 6101 } 6102 6103} 6104 6105// getEffectFromDesc_l() must be called with PlaybackThread::mLock held 6106sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 6107{ 6108 sp<EffectModule> effect; 6109 size_t size = mEffects.size(); 6110 6111 for (size_t i = 0; i < size; i++) { 6112 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 6113 effect = mEffects[i]; 6114 break; 6115 } 6116 } 6117 return effect; 6118} 6119 6120// getEffectFromId_l() must be called with PlaybackThread::mLock held 6121sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 6122{ 6123 sp<EffectModule> effect; 6124 size_t size = mEffects.size(); 6125 6126 for (size_t i = 0; i < size; i++) { 6127 // by convention, return first effect if id provided is 0 (0 is never a valid id) 6128 if (id == 0 || mEffects[i]->id() == id) { 6129 effect = mEffects[i]; 6130 break; 6131 } 6132 } 6133 return effect; 6134} 6135 6136// Must be called with EffectChain::mLock locked 6137void AudioFlinger::EffectChain::process_l() 6138{ 6139 sp<ThreadBase> thread = mThread.promote(); 6140 if (thread == 0) { 6141 LOGW("process_l(): cannot promote mixer thread"); 6142 return; 6143 } 6144 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 6145 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 6146 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 6147 bool tracksOnSession = false; 6148 if (!isGlobalSession) { 6149 tracksOnSession = (trackCnt() != 0); 6150 } 6151 6152 // if no track is active, input buffer must be cleared here as the mixer process 6153 // will not do it 6154 if (tracksOnSession && 6155 activeTrackCnt() == 0) { 6156 size_t numSamples = playbackThread->frameCount() * playbackThread->channelCount(); 6157 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 6158 } 6159 6160 size_t size = mEffects.size(); 6161 // do not process effect if no track is present in same audio session 6162 if (isGlobalSession || tracksOnSession) { 6163 for (size_t i = 0; i < size; i++) { 6164 mEffects[i]->process(); 6165 } 6166 } 6167 for (size_t i = 0; i < size; i++) { 6168 mEffects[i]->updateState(); 6169 } 6170} 6171 6172// addEffect_l() must be called with PlaybackThread::mLock held 6173status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 6174{ 6175 effect_descriptor_t desc = effect->desc(); 6176 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 6177 6178 Mutex::Autolock _l(mLock); 6179 effect->setChain(this); 6180 sp<ThreadBase> thread = mThread.promote(); 6181 if (thread == 0) { 6182 return NO_INIT; 6183 } 6184 effect->setThread(thread); 6185 6186 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6187 // Auxiliary effects are inserted at the beginning of mEffects vector as 6188 // they are processed first and accumulated in chain input buffer 6189 mEffects.insertAt(effect, 0); 6190 6191 // the input buffer for auxiliary effect contains mono samples in 6192 // 32 bit format. This is to avoid saturation in AudoMixer 6193 // accumulation stage. Saturation is done in EffectModule::process() before 6194 // calling the process in effect engine 6195 size_t numSamples = thread->frameCount(); 6196 int32_t *buffer = new int32_t[numSamples]; 6197 memset(buffer, 0, numSamples * sizeof(int32_t)); 6198 effect->setInBuffer((int16_t *)buffer); 6199 // auxiliary effects output samples to chain input buffer for further processing 6200 // by insert effects 6201 effect->setOutBuffer(mInBuffer); 6202 } else { 6203 // Insert effects are inserted at the end of mEffects vector as they are processed 6204 // after track and auxiliary effects. 6205 // Insert effect order as a function of indicated preference: 6206 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 6207 // another effect is present 6208 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 6209 // last effect claiming first position 6210 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 6211 // first effect claiming last position 6212 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 6213 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 6214 // already present 6215 6216 int size = (int)mEffects.size(); 6217 int idx_insert = size; 6218 int idx_insert_first = -1; 6219 int idx_insert_last = -1; 6220 6221 for (int i = 0; i < size; i++) { 6222 effect_descriptor_t d = mEffects[i]->desc(); 6223 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 6224 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 6225 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 6226 // check invalid effect chaining combinations 6227 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 6228 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 6229 LOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 6230 return INVALID_OPERATION; 6231 } 6232 // remember position of first insert effect and by default 6233 // select this as insert position for new effect 6234 if (idx_insert == size) { 6235 idx_insert = i; 6236 } 6237 // remember position of last insert effect claiming 6238 // first position 6239 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 6240 idx_insert_first = i; 6241 } 6242 // remember position of first insert effect claiming 6243 // last position 6244 if (iPref == EFFECT_FLAG_INSERT_LAST && 6245 idx_insert_last == -1) { 6246 idx_insert_last = i; 6247 } 6248 } 6249 } 6250 6251 // modify idx_insert from first position if needed 6252 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 6253 if (idx_insert_last != -1) { 6254 idx_insert = idx_insert_last; 6255 } else { 6256 idx_insert = size; 6257 } 6258 } else { 6259 if (idx_insert_first != -1) { 6260 idx_insert = idx_insert_first + 1; 6261 } 6262 } 6263 6264 // always read samples from chain input buffer 6265 effect->setInBuffer(mInBuffer); 6266 6267 // if last effect in the chain, output samples to chain 6268 // output buffer, otherwise to chain input buffer 6269 if (idx_insert == size) { 6270 if (idx_insert != 0) { 6271 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 6272 mEffects[idx_insert-1]->configure(); 6273 } 6274 effect->setOutBuffer(mOutBuffer); 6275 } else { 6276 effect->setOutBuffer(mInBuffer); 6277 } 6278 mEffects.insertAt(effect, idx_insert); 6279 6280 LOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 6281 } 6282 effect->configure(); 6283 return NO_ERROR; 6284} 6285 6286// removeEffect_l() must be called with PlaybackThread::mLock held 6287size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 6288{ 6289 Mutex::Autolock _l(mLock); 6290 int size = (int)mEffects.size(); 6291 int i; 6292 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 6293 6294 for (i = 0; i < size; i++) { 6295 if (effect == mEffects[i]) { 6296 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 6297 delete[] effect->inBuffer(); 6298 } else { 6299 if (i == size - 1 && i != 0) { 6300 mEffects[i - 1]->setOutBuffer(mOutBuffer); 6301 mEffects[i - 1]->configure(); 6302 } 6303 } 6304 mEffects.removeAt(i); 6305 LOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 6306 break; 6307 } 6308 } 6309 6310 return mEffects.size(); 6311} 6312 6313// setDevice_l() must be called with PlaybackThread::mLock held 6314void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 6315{ 6316 size_t size = mEffects.size(); 6317 for (size_t i = 0; i < size; i++) { 6318 mEffects[i]->setDevice(device); 6319 } 6320} 6321 6322// setMode_l() must be called with PlaybackThread::mLock held 6323void AudioFlinger::EffectChain::setMode_l(uint32_t mode) 6324{ 6325 size_t size = mEffects.size(); 6326 for (size_t i = 0; i < size; i++) { 6327 mEffects[i]->setMode(mode); 6328 } 6329} 6330 6331// setVolume_l() must be called with PlaybackThread::mLock held 6332bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 6333{ 6334 uint32_t newLeft = *left; 6335 uint32_t newRight = *right; 6336 bool hasControl = false; 6337 int ctrlIdx = -1; 6338 size_t size = mEffects.size(); 6339 6340 // first update volume controller 6341 for (size_t i = size; i > 0; i--) { 6342 if (mEffects[i - 1]->isProcessEnabled() && 6343 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 6344 ctrlIdx = i - 1; 6345 hasControl = true; 6346 break; 6347 } 6348 } 6349 6350 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 6351 if (hasControl) { 6352 *left = mNewLeftVolume; 6353 *right = mNewRightVolume; 6354 } 6355 return hasControl; 6356 } 6357 6358 mVolumeCtrlIdx = ctrlIdx; 6359 mLeftVolume = newLeft; 6360 mRightVolume = newRight; 6361 6362 // second get volume update from volume controller 6363 if (ctrlIdx >= 0) { 6364 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 6365 mNewLeftVolume = newLeft; 6366 mNewRightVolume = newRight; 6367 } 6368 // then indicate volume to all other effects in chain. 6369 // Pass altered volume to effects before volume controller 6370 // and requested volume to effects after controller 6371 uint32_t lVol = newLeft; 6372 uint32_t rVol = newRight; 6373 6374 for (size_t i = 0; i < size; i++) { 6375 if ((int)i == ctrlIdx) continue; 6376 // this also works for ctrlIdx == -1 when there is no volume controller 6377 if ((int)i > ctrlIdx) { 6378 lVol = *left; 6379 rVol = *right; 6380 } 6381 mEffects[i]->setVolume(&lVol, &rVol, false); 6382 } 6383 *left = newLeft; 6384 *right = newRight; 6385 6386 return hasControl; 6387} 6388 6389status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 6390{ 6391 const size_t SIZE = 256; 6392 char buffer[SIZE]; 6393 String8 result; 6394 6395 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 6396 result.append(buffer); 6397 6398 bool locked = tryLock(mLock); 6399 // failed to lock - AudioFlinger is probably deadlocked 6400 if (!locked) { 6401 result.append("\tCould not lock mutex:\n"); 6402 } 6403 6404 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 6405 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 6406 mEffects.size(), 6407 (uint32_t)mInBuffer, 6408 (uint32_t)mOutBuffer, 6409 mActiveTrackCnt); 6410 result.append(buffer); 6411 write(fd, result.string(), result.size()); 6412 6413 for (size_t i = 0; i < mEffects.size(); ++i) { 6414 sp<EffectModule> effect = mEffects[i]; 6415 if (effect != 0) { 6416 effect->dump(fd, args); 6417 } 6418 } 6419 6420 if (locked) { 6421 mLock.unlock(); 6422 } 6423 6424 return NO_ERROR; 6425} 6426 6427#undef LOG_TAG 6428#define LOG_TAG "AudioFlinger" 6429 6430// ---------------------------------------------------------------------------- 6431 6432status_t AudioFlinger::onTransact( 6433 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 6434{ 6435 return BnAudioFlinger::onTransact(code, data, reply, flags); 6436} 6437 6438}; // namespace android 6439