AudioFlinger.cpp revision 77c1119ea0b5cb32287088ceeeb7e3b6bd14a85d
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#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51 52#include <media/EffectsFactoryApi.h> 53#include <audio_effects/effect_visualizer.h> 54#include <audio_effects/effect_ns.h> 55#include <audio_effects/effect_aec.h> 56 57#include <audio_utils/primitives.h> 58 59#include <cpustats/ThreadCpuUsage.h> 60#include <powermanager/PowerManager.h> 61// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 62 63// ---------------------------------------------------------------------------- 64 65 66namespace android { 67 68static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 69static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 70 71//static const nsecs_t kStandbyTimeInNsecs = seconds(3); 72static const float MAX_GAIN = 4096.0f; 73static const uint32_t MAX_GAIN_INT = 0x1000; 74 75// retry counts for buffer fill timeout 76// 50 * ~20msecs = 1 second 77static const int8_t kMaxTrackRetries = 50; 78static const int8_t kMaxTrackStartupRetries = 50; 79// allow less retry attempts on direct output thread. 80// direct outputs can be a scarce resource in audio hardware and should 81// be released as quickly as possible. 82static const int8_t kMaxTrackRetriesDirect = 2; 83 84static const int kDumpLockRetries = 50; 85static const int kDumpLockSleepUs = 20000; 86 87// don't warn about blocked writes or record buffer overflows more often than this 88static const nsecs_t kWarningThrottleNs = seconds(5); 89 90// RecordThread loop sleep time upon application overrun or audio HAL read error 91static const int kRecordThreadSleepUs = 5000; 92 93// maximum time to wait for setParameters to complete 94static const nsecs_t kSetParametersTimeoutNs = seconds(2); 95 96// minimum sleep time for the mixer thread loop when tracks are active but in underrun 97static const uint32_t kMinThreadSleepTimeUs = 5000; 98// maximum divider applied to the active sleep time in the mixer thread loop 99static const uint32_t kMaxThreadSleepTimeShift = 2; 100 101 102// ---------------------------------------------------------------------------- 103 104static bool recordingAllowed() { 105 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 106 bool ok = checkCallingPermission(String16("android.permission.RECORD_AUDIO")); 107 if (!ok) ALOGE("Request requires android.permission.RECORD_AUDIO"); 108 return ok; 109} 110 111static bool settingsAllowed() { 112 if (getpid() == IPCThreadState::self()->getCallingPid()) return true; 113 bool ok = checkCallingPermission(String16("android.permission.MODIFY_AUDIO_SETTINGS")); 114 if (!ok) ALOGE("Request requires android.permission.MODIFY_AUDIO_SETTINGS"); 115 return ok; 116} 117 118// To collect the amplifier usage 119static void addBatteryData(uint32_t params) { 120 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 121 if (service == NULL) { 122 // it already logged 123 return; 124 } 125 126 service->addBatteryData(params); 127} 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(NULL), mMasterVolume(1.0f), mMasterMute(false), mNextUniqueId(1), 164 mBtNrecIsOff(false) 165{ 166} 167 168void AudioFlinger::onFirstRef() 169{ 170 int rc = 0; 171 172 Mutex::Autolock _l(mLock); 173 174 /* TODO: move all this work into an Init() function */ 175 mHardwareStatus = AUDIO_HW_IDLE; 176 177 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 178 const hw_module_t *mod; 179 audio_hw_device_t *dev; 180 181 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 182 if (rc) 183 continue; 184 185 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 186 mod->name, mod->id); 187 mAudioHwDevs.push(dev); 188 189 if (!mPrimaryHardwareDev) { 190 mPrimaryHardwareDev = dev; 191 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 192 mod->name, mod->id, audio_interfaces[i]); 193 } 194 } 195 196 mHardwareStatus = AUDIO_HW_INIT; 197 198 if (!mPrimaryHardwareDev || mAudioHwDevs.size() == 0) { 199 ALOGE("Primary audio interface not found"); 200 return; 201 } 202 203 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 204 audio_hw_device_t *dev = mAudioHwDevs[i]; 205 206 mHardwareStatus = AUDIO_HW_INIT; 207 rc = dev->init_check(dev); 208 if (rc == 0) { 209 AutoMutex lock(mHardwareLock); 210 211 mMode = AUDIO_MODE_NORMAL; 212 mHardwareStatus = AUDIO_HW_SET_MODE; 213 dev->set_mode(dev, mMode); 214 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 215 dev->set_master_volume(dev, 1.0f); 216 mHardwareStatus = AUDIO_HW_IDLE; 217 } 218 } 219} 220 221status_t AudioFlinger::initCheck() const 222{ 223 Mutex::Autolock _l(mLock); 224 if (mPrimaryHardwareDev == NULL || mAudioHwDevs.size() == 0) 225 return NO_INIT; 226 return NO_ERROR; 227} 228 229AudioFlinger::~AudioFlinger() 230{ 231 int num_devs = mAudioHwDevs.size(); 232 233 while (!mRecordThreads.isEmpty()) { 234 // closeInput() will remove first entry from mRecordThreads 235 closeInput(mRecordThreads.keyAt(0)); 236 } 237 while (!mPlaybackThreads.isEmpty()) { 238 // closeOutput() will remove first entry from mPlaybackThreads 239 closeOutput(mPlaybackThreads.keyAt(0)); 240 } 241 242 for (int i = 0; i < num_devs; i++) { 243 audio_hw_device_t *dev = mAudioHwDevs[i]; 244 audio_hw_device_close(dev); 245 } 246 mAudioHwDevs.clear(); 247} 248 249audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 250{ 251 /* first matching HW device is returned */ 252 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 253 audio_hw_device_t *dev = mAudioHwDevs[i]; 254 if ((dev->get_supported_devices(dev) & devices) == devices) 255 return dev; 256 } 257 return NULL; 258} 259 260status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 261{ 262 const size_t SIZE = 256; 263 char buffer[SIZE]; 264 String8 result; 265 266 result.append("Clients:\n"); 267 for (size_t i = 0; i < mClients.size(); ++i) { 268 sp<Client> client = mClients.valueAt(i).promote(); 269 if (client != 0) { 270 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 271 result.append(buffer); 272 } 273 } 274 275 result.append("Global session refs:\n"); 276 result.append(" session pid cnt\n"); 277 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 278 AudioSessionRef *r = mAudioSessionRefs[i]; 279 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 280 result.append(buffer); 281 } 282 write(fd, result.string(), result.size()); 283 return NO_ERROR; 284} 285 286 287status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 288{ 289 const size_t SIZE = 256; 290 char buffer[SIZE]; 291 String8 result; 292 hardware_call_state hardwareStatus = mHardwareStatus; 293 294 snprintf(buffer, SIZE, "Hardware status: %d\n", hardwareStatus); 295 result.append(buffer); 296 write(fd, result.string(), result.size()); 297 return NO_ERROR; 298} 299 300status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 301{ 302 const size_t SIZE = 256; 303 char buffer[SIZE]; 304 String8 result; 305 snprintf(buffer, SIZE, "Permission Denial: " 306 "can't dump AudioFlinger from pid=%d, uid=%d\n", 307 IPCThreadState::self()->getCallingPid(), 308 IPCThreadState::self()->getCallingUid()); 309 result.append(buffer); 310 write(fd, result.string(), result.size()); 311 return NO_ERROR; 312} 313 314static bool tryLock(Mutex& mutex) 315{ 316 bool locked = false; 317 for (int i = 0; i < kDumpLockRetries; ++i) { 318 if (mutex.tryLock() == NO_ERROR) { 319 locked = true; 320 break; 321 } 322 usleep(kDumpLockSleepUs); 323 } 324 return locked; 325} 326 327status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 328{ 329 if (!checkCallingPermission(String16("android.permission.DUMP"))) { 330 dumpPermissionDenial(fd, args); 331 } else { 332 // get state of hardware lock 333 bool hardwareLocked = tryLock(mHardwareLock); 334 if (!hardwareLocked) { 335 String8 result(kHardwareLockedString); 336 write(fd, result.string(), result.size()); 337 } else { 338 mHardwareLock.unlock(); 339 } 340 341 bool locked = tryLock(mLock); 342 343 // failed to lock - AudioFlinger is probably deadlocked 344 if (!locked) { 345 String8 result(kDeadlockedString); 346 write(fd, result.string(), result.size()); 347 } 348 349 dumpClients(fd, args); 350 dumpInternals(fd, args); 351 352 // dump playback threads 353 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 354 mPlaybackThreads.valueAt(i)->dump(fd, args); 355 } 356 357 // dump record threads 358 for (size_t i = 0; i < mRecordThreads.size(); i++) { 359 mRecordThreads.valueAt(i)->dump(fd, args); 360 } 361 362 // dump all hardware devs 363 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 364 audio_hw_device_t *dev = mAudioHwDevs[i]; 365 dev->dump(dev, fd); 366 } 367 if (locked) mLock.unlock(); 368 } 369 return NO_ERROR; 370} 371 372 373// IAudioFlinger interface 374 375 376sp<IAudioTrack> AudioFlinger::createTrack( 377 pid_t pid, 378 audio_stream_type_t streamType, 379 uint32_t sampleRate, 380 audio_format_t format, 381 uint32_t channelMask, 382 int frameCount, 383 uint32_t flags, 384 const sp<IMemory>& sharedBuffer, 385 int output, 386 int *sessionId, 387 status_t *status) 388{ 389 sp<PlaybackThread::Track> track; 390 sp<TrackHandle> trackHandle; 391 sp<Client> client; 392 wp<Client> wclient; 393 status_t lStatus; 394 int lSessionId; 395 396 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 397 // but if someone uses binder directly they could bypass that and cause us to crash 398 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 399 ALOGE("createTrack() invalid stream type %d", streamType); 400 lStatus = BAD_VALUE; 401 goto Exit; 402 } 403 404 { 405 Mutex::Autolock _l(mLock); 406 PlaybackThread *thread = checkPlaybackThread_l(output); 407 PlaybackThread *effectThread = NULL; 408 if (thread == NULL) { 409 ALOGE("unknown output thread"); 410 lStatus = BAD_VALUE; 411 goto Exit; 412 } 413 414 wclient = mClients.valueFor(pid); 415 416 if (wclient != NULL) { 417 client = wclient.promote(); 418 } else { 419 client = new Client(this, pid); 420 mClients.add(pid, client); 421 } 422 423 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 424 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 425 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 426 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 427 if (mPlaybackThreads.keyAt(i) != output) { 428 // prevent same audio session on different output threads 429 uint32_t sessions = t->hasAudioSession(*sessionId); 430 if (sessions & PlaybackThread::TRACK_SESSION) { 431 ALOGE("createTrack() session ID %d already in use", *sessionId); 432 lStatus = BAD_VALUE; 433 goto Exit; 434 } 435 // check if an effect with same session ID is waiting for a track to be created 436 if (sessions & PlaybackThread::EFFECT_SESSION) { 437 effectThread = t.get(); 438 } 439 } 440 } 441 lSessionId = *sessionId; 442 } else { 443 // if no audio session id is provided, create one here 444 lSessionId = nextUniqueId(); 445 if (sessionId != NULL) { 446 *sessionId = lSessionId; 447 } 448 } 449 ALOGV("createTrack() lSessionId: %d", lSessionId); 450 451 track = thread->createTrack_l(client, streamType, sampleRate, format, 452 channelMask, frameCount, sharedBuffer, lSessionId, &lStatus); 453 454 // move effect chain to this output thread if an effect on same session was waiting 455 // for a track to be created 456 if (lStatus == NO_ERROR && effectThread != NULL) { 457 Mutex::Autolock _dl(thread->mLock); 458 Mutex::Autolock _sl(effectThread->mLock); 459 moveEffectChain_l(lSessionId, effectThread, thread, true); 460 } 461 } 462 if (lStatus == NO_ERROR) { 463 trackHandle = new TrackHandle(track); 464 } else { 465 // remove local strong reference to Client before deleting the Track so that the Client 466 // destructor is called by the TrackBase destructor with mLock held 467 client.clear(); 468 track.clear(); 469 } 470 471Exit: 472 if(status) { 473 *status = lStatus; 474 } 475 return trackHandle; 476} 477 478uint32_t AudioFlinger::sampleRate(int output) const 479{ 480 Mutex::Autolock _l(mLock); 481 PlaybackThread *thread = checkPlaybackThread_l(output); 482 if (thread == NULL) { 483 ALOGW("sampleRate() unknown thread %d", output); 484 return 0; 485 } 486 return thread->sampleRate(); 487} 488 489int AudioFlinger::channelCount(int output) const 490{ 491 Mutex::Autolock _l(mLock); 492 PlaybackThread *thread = checkPlaybackThread_l(output); 493 if (thread == NULL) { 494 ALOGW("channelCount() unknown thread %d", output); 495 return 0; 496 } 497 return thread->channelCount(); 498} 499 500audio_format_t AudioFlinger::format(int output) const 501{ 502 Mutex::Autolock _l(mLock); 503 PlaybackThread *thread = checkPlaybackThread_l(output); 504 if (thread == NULL) { 505 ALOGW("format() unknown thread %d", output); 506 return AUDIO_FORMAT_INVALID; 507 } 508 return thread->format(); 509} 510 511size_t AudioFlinger::frameCount(int output) const 512{ 513 Mutex::Autolock _l(mLock); 514 PlaybackThread *thread = checkPlaybackThread_l(output); 515 if (thread == NULL) { 516 ALOGW("frameCount() unknown thread %d", output); 517 return 0; 518 } 519 return thread->frameCount(); 520} 521 522uint32_t AudioFlinger::latency(int output) const 523{ 524 Mutex::Autolock _l(mLock); 525 PlaybackThread *thread = checkPlaybackThread_l(output); 526 if (thread == NULL) { 527 ALOGW("latency() unknown thread %d", output); 528 return 0; 529 } 530 return thread->latency(); 531} 532 533status_t AudioFlinger::setMasterVolume(float value) 534{ 535 status_t ret = initCheck(); 536 if (ret != NO_ERROR) { 537 return ret; 538 } 539 540 // check calling permissions 541 if (!settingsAllowed()) { 542 return PERMISSION_DENIED; 543 } 544 545 // when hw supports master volume, don't scale in sw mixer 546 { // scope for the lock 547 AutoMutex lock(mHardwareLock); 548 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 549 if (mPrimaryHardwareDev->set_master_volume(mPrimaryHardwareDev, value) == NO_ERROR) { 550 value = 1.0f; 551 } 552 mHardwareStatus = AUDIO_HW_IDLE; 553 } 554 555 Mutex::Autolock _l(mLock); 556 mMasterVolume = value; 557 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 558 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 559 560 return NO_ERROR; 561} 562 563status_t AudioFlinger::setMode(audio_mode_t mode) 564{ 565 status_t ret = initCheck(); 566 if (ret != NO_ERROR) { 567 return ret; 568 } 569 570 // check calling permissions 571 if (!settingsAllowed()) { 572 return PERMISSION_DENIED; 573 } 574 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 575 ALOGW("Illegal value: setMode(%d)", mode); 576 return BAD_VALUE; 577 } 578 579 { // scope for the lock 580 AutoMutex lock(mHardwareLock); 581 mHardwareStatus = AUDIO_HW_SET_MODE; 582 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 583 mHardwareStatus = AUDIO_HW_IDLE; 584 } 585 586 if (NO_ERROR == ret) { 587 Mutex::Autolock _l(mLock); 588 mMode = mode; 589 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 590 mPlaybackThreads.valueAt(i)->setMode(mode); 591 } 592 593 return ret; 594} 595 596status_t AudioFlinger::setMicMute(bool state) 597{ 598 status_t ret = initCheck(); 599 if (ret != NO_ERROR) { 600 return ret; 601 } 602 603 // check calling permissions 604 if (!settingsAllowed()) { 605 return PERMISSION_DENIED; 606 } 607 608 AutoMutex lock(mHardwareLock); 609 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 610 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 611 mHardwareStatus = AUDIO_HW_IDLE; 612 return ret; 613} 614 615bool AudioFlinger::getMicMute() const 616{ 617 status_t ret = initCheck(); 618 if (ret != NO_ERROR) { 619 return false; 620 } 621 622 bool state = AUDIO_MODE_INVALID; 623 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 624 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 625 mHardwareStatus = AUDIO_HW_IDLE; 626 return state; 627} 628 629status_t AudioFlinger::setMasterMute(bool muted) 630{ 631 // check calling permissions 632 if (!settingsAllowed()) { 633 return PERMISSION_DENIED; 634 } 635 636 Mutex::Autolock _l(mLock); 637 mMasterMute = muted; 638 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 639 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 640 641 return NO_ERROR; 642} 643 644float AudioFlinger::masterVolume() const 645{ 646 Mutex::Autolock _l(mLock); 647 return masterVolume_l(); 648} 649 650bool AudioFlinger::masterMute() const 651{ 652 Mutex::Autolock _l(mLock); 653 return masterMute_l(); 654} 655 656status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, int output) 657{ 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 664 ALOGE("setStreamVolume() invalid stream %d", stream); 665 return BAD_VALUE; 666 } 667 668 AutoMutex lock(mLock); 669 PlaybackThread *thread = NULL; 670 if (output) { 671 thread = checkPlaybackThread_l(output); 672 if (thread == NULL) { 673 return BAD_VALUE; 674 } 675 } 676 677 mStreamTypes[stream].volume = value; 678 679 if (thread == NULL) { 680 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) { 681 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 682 } 683 } else { 684 thread->setStreamVolume(stream, value); 685 } 686 687 return NO_ERROR; 688} 689 690status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 691{ 692 // check calling permissions 693 if (!settingsAllowed()) { 694 return PERMISSION_DENIED; 695 } 696 697 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 698 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 699 ALOGE("setStreamMute() invalid stream %d", stream); 700 return BAD_VALUE; 701 } 702 703 AutoMutex lock(mLock); 704 mStreamTypes[stream].mute = muted; 705 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 706 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 707 708 return NO_ERROR; 709} 710 711float AudioFlinger::streamVolume(audio_stream_type_t stream, int output) const 712{ 713 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 714 return 0.0f; 715 } 716 717 AutoMutex lock(mLock); 718 float volume; 719 if (output) { 720 PlaybackThread *thread = checkPlaybackThread_l(output); 721 if (thread == NULL) { 722 return 0.0f; 723 } 724 volume = thread->streamVolume(stream); 725 } else { 726 volume = mStreamTypes[stream].volume; 727 } 728 729 return volume; 730} 731 732bool AudioFlinger::streamMute(audio_stream_type_t stream) const 733{ 734 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 735 return true; 736 } 737 738 return mStreamTypes[stream].mute; 739} 740 741status_t AudioFlinger::setParameters(int ioHandle, const String8& keyValuePairs) 742{ 743 status_t result; 744 745 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling tid %d", 746 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 747 // check calling permissions 748 if (!settingsAllowed()) { 749 return PERMISSION_DENIED; 750 } 751 752 // ioHandle == 0 means the parameters are global to the audio hardware interface 753 if (ioHandle == 0) { 754 AutoMutex lock(mHardwareLock); 755 mHardwareStatus = AUDIO_SET_PARAMETER; 756 status_t final_result = NO_ERROR; 757 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 758 audio_hw_device_t *dev = mAudioHwDevs[i]; 759 result = dev->set_parameters(dev, keyValuePairs.string()); 760 final_result = result ?: final_result; 761 } 762 mHardwareStatus = AUDIO_HW_IDLE; 763 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 764 AudioParameter param = AudioParameter(keyValuePairs); 765 String8 value; 766 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 767 Mutex::Autolock _l(mLock); 768 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 769 if (mBtNrecIsOff != btNrecIsOff) { 770 for (size_t i = 0; i < mRecordThreads.size(); i++) { 771 sp<RecordThread> thread = mRecordThreads.valueAt(i); 772 RecordThread::RecordTrack *track = thread->track(); 773 if (track != NULL) { 774 audio_devices_t device = (audio_devices_t)( 775 thread->device() & AUDIO_DEVICE_IN_ALL); 776 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 777 thread->setEffectSuspended(FX_IID_AEC, 778 suspend, 779 track->sessionId()); 780 thread->setEffectSuspended(FX_IID_NS, 781 suspend, 782 track->sessionId()); 783 } 784 } 785 mBtNrecIsOff = btNrecIsOff; 786 } 787 } 788 return final_result; 789 } 790 791 // hold a strong ref on thread in case closeOutput() or closeInput() is called 792 // and the thread is exited once the lock is released 793 sp<ThreadBase> thread; 794 { 795 Mutex::Autolock _l(mLock); 796 thread = checkPlaybackThread_l(ioHandle); 797 if (thread == NULL) { 798 thread = checkRecordThread_l(ioHandle); 799 } else if (thread == primaryPlaybackThread_l()) { 800 // indicate output device change to all input threads for pre processing 801 AudioParameter param = AudioParameter(keyValuePairs); 802 int value; 803 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 804 for (size_t i = 0; i < mRecordThreads.size(); i++) { 805 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 806 } 807 } 808 } 809 } 810 if (thread != NULL) { 811 result = thread->setParameters(keyValuePairs); 812 return result; 813 } 814 return BAD_VALUE; 815} 816 817String8 AudioFlinger::getParameters(int ioHandle, const String8& keys) 818{ 819// ALOGV("getParameters() io %d, keys %s, tid %d, calling tid %d", 820// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 821 822 if (ioHandle == 0) { 823 String8 out_s8; 824 825 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 826 audio_hw_device_t *dev = mAudioHwDevs[i]; 827 char *s = dev->get_parameters(dev, keys.string()); 828 out_s8 += String8(s); 829 free(s); 830 } 831 return out_s8; 832 } 833 834 Mutex::Autolock _l(mLock); 835 836 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 837 if (playbackThread != NULL) { 838 return playbackThread->getParameters(keys); 839 } 840 RecordThread *recordThread = checkRecordThread_l(ioHandle); 841 if (recordThread != NULL) { 842 return recordThread->getParameters(keys); 843 } 844 return String8(""); 845} 846 847size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) 848{ 849 status_t ret = initCheck(); 850 if (ret != NO_ERROR) { 851 return 0; 852 } 853 854 return mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 855} 856 857unsigned int AudioFlinger::getInputFramesLost(int ioHandle) 858{ 859 if (ioHandle == 0) { 860 return 0; 861 } 862 863 Mutex::Autolock _l(mLock); 864 865 RecordThread *recordThread = checkRecordThread_l(ioHandle); 866 if (recordThread != NULL) { 867 return recordThread->getInputFramesLost(); 868 } 869 return 0; 870} 871 872status_t AudioFlinger::setVoiceVolume(float value) 873{ 874 status_t ret = initCheck(); 875 if (ret != NO_ERROR) { 876 return ret; 877 } 878 879 // check calling permissions 880 if (!settingsAllowed()) { 881 return PERMISSION_DENIED; 882 } 883 884 AutoMutex lock(mHardwareLock); 885 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 886 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 887 mHardwareStatus = AUDIO_HW_IDLE; 888 889 return ret; 890} 891 892status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int output) 893{ 894 status_t status; 895 896 Mutex::Autolock _l(mLock); 897 898 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 899 if (playbackThread != NULL) { 900 return playbackThread->getRenderPosition(halFrames, dspFrames); 901 } 902 903 return BAD_VALUE; 904} 905 906void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 907{ 908 909 Mutex::Autolock _l(mLock); 910 911 int pid = IPCThreadState::self()->getCallingPid(); 912 if (mNotificationClients.indexOfKey(pid) < 0) { 913 sp<NotificationClient> notificationClient = new NotificationClient(this, 914 client, 915 pid); 916 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 917 918 mNotificationClients.add(pid, notificationClient); 919 920 sp<IBinder> binder = client->asBinder(); 921 binder->linkToDeath(notificationClient); 922 923 // the config change is always sent from playback or record threads to avoid deadlock 924 // with AudioSystem::gLock 925 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 926 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 927 } 928 929 for (size_t i = 0; i < mRecordThreads.size(); i++) { 930 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 931 } 932 } 933} 934 935void AudioFlinger::removeNotificationClient(pid_t pid) 936{ 937 Mutex::Autolock _l(mLock); 938 939 int index = mNotificationClients.indexOfKey(pid); 940 if (index >= 0) { 941 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 942 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 943 mNotificationClients.removeItem(pid); 944 } 945 946 ALOGV("%d died, releasing its sessions", pid); 947 int num = mAudioSessionRefs.size(); 948 bool removed = false; 949 for (int i = 0; i< num; i++) { 950 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 951 ALOGV(" pid %d @ %d", ref->pid, i); 952 if (ref->pid == pid) { 953 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 954 mAudioSessionRefs.removeAt(i); 955 delete ref; 956 removed = true; 957 i--; 958 num--; 959 } 960 } 961 if (removed) { 962 purgeStaleEffects_l(); 963 } 964} 965 966// audioConfigChanged_l() must be called with AudioFlinger::mLock held 967void AudioFlinger::audioConfigChanged_l(int event, int ioHandle, void *param2) 968{ 969 size_t size = mNotificationClients.size(); 970 for (size_t i = 0; i < size; i++) { 971 mNotificationClients.valueAt(i)->client()->ioConfigChanged(event, ioHandle, param2); 972 } 973} 974 975// removeClient_l() must be called with AudioFlinger::mLock held 976void AudioFlinger::removeClient_l(pid_t pid) 977{ 978 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 979 mClients.removeItem(pid); 980} 981 982 983// ---------------------------------------------------------------------------- 984 985AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, int id, uint32_t device) 986 : Thread(false), 987 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), mChannelCount(0), 988 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), mStandby(false), mId(id), mExiting(false), 989 mDevice(device) 990{ 991 mDeathRecipient = new PMDeathRecipient(this); 992} 993 994AudioFlinger::ThreadBase::~ThreadBase() 995{ 996 mParamCond.broadcast(); 997 // do not lock the mutex in destructor 998 releaseWakeLock_l(); 999 if (mPowerManager != 0) { 1000 sp<IBinder> binder = mPowerManager->asBinder(); 1001 binder->unlinkToDeath(mDeathRecipient); 1002 } 1003} 1004 1005void AudioFlinger::ThreadBase::exit() 1006{ 1007 // keep a strong ref on ourself so that we won't get 1008 // destroyed in the middle of requestExitAndWait() 1009 sp <ThreadBase> strongMe = this; 1010 1011 ALOGV("ThreadBase::exit"); 1012 { 1013 AutoMutex lock(mLock); 1014 mExiting = true; 1015 requestExit(); 1016 mWaitWorkCV.signal(); 1017 } 1018 requestExitAndWait(); 1019} 1020 1021uint32_t AudioFlinger::ThreadBase::sampleRate() const 1022{ 1023 return mSampleRate; 1024} 1025 1026int AudioFlinger::ThreadBase::channelCount() const 1027{ 1028 return (int)mChannelCount; 1029} 1030 1031audio_format_t AudioFlinger::ThreadBase::format() const 1032{ 1033 return mFormat; 1034} 1035 1036size_t AudioFlinger::ThreadBase::frameCount() const 1037{ 1038 return mFrameCount; 1039} 1040 1041status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1042{ 1043 status_t status; 1044 1045 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1046 Mutex::Autolock _l(mLock); 1047 1048 mNewParameters.add(keyValuePairs); 1049 mWaitWorkCV.signal(); 1050 // wait condition with timeout in case the thread loop has exited 1051 // before the request could be processed 1052 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1053 status = mParamStatus; 1054 mWaitWorkCV.signal(); 1055 } else { 1056 status = TIMED_OUT; 1057 } 1058 return status; 1059} 1060 1061void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1062{ 1063 Mutex::Autolock _l(mLock); 1064 sendConfigEvent_l(event, param); 1065} 1066 1067// sendConfigEvent_l() must be called with ThreadBase::mLock held 1068void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1069{ 1070 ConfigEvent configEvent; 1071 configEvent.mEvent = event; 1072 configEvent.mParam = param; 1073 mConfigEvents.add(configEvent); 1074 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1075 mWaitWorkCV.signal(); 1076} 1077 1078void AudioFlinger::ThreadBase::processConfigEvents() 1079{ 1080 mLock.lock(); 1081 while(!mConfigEvents.isEmpty()) { 1082 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1083 ConfigEvent configEvent = mConfigEvents[0]; 1084 mConfigEvents.removeAt(0); 1085 // release mLock before locking AudioFlinger mLock: lock order is always 1086 // AudioFlinger then ThreadBase to avoid cross deadlock 1087 mLock.unlock(); 1088 mAudioFlinger->mLock.lock(); 1089 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1090 mAudioFlinger->mLock.unlock(); 1091 mLock.lock(); 1092 } 1093 mLock.unlock(); 1094} 1095 1096status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1097{ 1098 const size_t SIZE = 256; 1099 char buffer[SIZE]; 1100 String8 result; 1101 1102 bool locked = tryLock(mLock); 1103 if (!locked) { 1104 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1105 write(fd, buffer, strlen(buffer)); 1106 } 1107 1108 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1109 result.append(buffer); 1110 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1111 result.append(buffer); 1112 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1113 result.append(buffer); 1114 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1115 result.append(buffer); 1116 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1117 result.append(buffer); 1118 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1119 result.append(buffer); 1120 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1121 result.append(buffer); 1122 1123 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1124 result.append(buffer); 1125 result.append(" Index Command"); 1126 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1127 snprintf(buffer, SIZE, "\n %02d ", i); 1128 result.append(buffer); 1129 result.append(mNewParameters[i]); 1130 } 1131 1132 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1133 result.append(buffer); 1134 snprintf(buffer, SIZE, " Index event param\n"); 1135 result.append(buffer); 1136 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1137 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1138 result.append(buffer); 1139 } 1140 result.append("\n"); 1141 1142 write(fd, result.string(), result.size()); 1143 1144 if (locked) { 1145 mLock.unlock(); 1146 } 1147 return NO_ERROR; 1148} 1149 1150status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1151{ 1152 const size_t SIZE = 256; 1153 char buffer[SIZE]; 1154 String8 result; 1155 1156 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1157 write(fd, buffer, strlen(buffer)); 1158 1159 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1160 sp<EffectChain> chain = mEffectChains[i]; 1161 if (chain != 0) { 1162 chain->dump(fd, args); 1163 } 1164 } 1165 return NO_ERROR; 1166} 1167 1168void AudioFlinger::ThreadBase::acquireWakeLock() 1169{ 1170 Mutex::Autolock _l(mLock); 1171 acquireWakeLock_l(); 1172} 1173 1174void AudioFlinger::ThreadBase::acquireWakeLock_l() 1175{ 1176 if (mPowerManager == 0) { 1177 // use checkService() to avoid blocking if power service is not up yet 1178 sp<IBinder> binder = 1179 defaultServiceManager()->checkService(String16("power")); 1180 if (binder == 0) { 1181 ALOGW("Thread %s cannot connect to the power manager service", mName); 1182 } else { 1183 mPowerManager = interface_cast<IPowerManager>(binder); 1184 binder->linkToDeath(mDeathRecipient); 1185 } 1186 } 1187 if (mPowerManager != 0) { 1188 sp<IBinder> binder = new BBinder(); 1189 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1190 binder, 1191 String16(mName)); 1192 if (status == NO_ERROR) { 1193 mWakeLockToken = binder; 1194 } 1195 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1196 } 1197} 1198 1199void AudioFlinger::ThreadBase::releaseWakeLock() 1200{ 1201 Mutex::Autolock _l(mLock); 1202 releaseWakeLock_l(); 1203} 1204 1205void AudioFlinger::ThreadBase::releaseWakeLock_l() 1206{ 1207 if (mWakeLockToken != 0) { 1208 ALOGV("releaseWakeLock_l() %s", mName); 1209 if (mPowerManager != 0) { 1210 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1211 } 1212 mWakeLockToken.clear(); 1213 } 1214} 1215 1216void AudioFlinger::ThreadBase::clearPowerManager() 1217{ 1218 Mutex::Autolock _l(mLock); 1219 releaseWakeLock_l(); 1220 mPowerManager.clear(); 1221} 1222 1223void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1224{ 1225 sp<ThreadBase> thread = mThread.promote(); 1226 if (thread != 0) { 1227 thread->clearPowerManager(); 1228 } 1229 ALOGW("power manager service died !!!"); 1230} 1231 1232void AudioFlinger::ThreadBase::setEffectSuspended( 1233 const effect_uuid_t *type, bool suspend, int sessionId) 1234{ 1235 Mutex::Autolock _l(mLock); 1236 setEffectSuspended_l(type, suspend, sessionId); 1237} 1238 1239void AudioFlinger::ThreadBase::setEffectSuspended_l( 1240 const effect_uuid_t *type, bool suspend, int sessionId) 1241{ 1242 sp<EffectChain> chain; 1243 chain = getEffectChain_l(sessionId); 1244 if (chain != 0) { 1245 if (type != NULL) { 1246 chain->setEffectSuspended_l(type, suspend); 1247 } else { 1248 chain->setEffectSuspendedAll_l(suspend); 1249 } 1250 } 1251 1252 updateSuspendedSessions_l(type, suspend, sessionId); 1253} 1254 1255void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1256{ 1257 int index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1258 if (index < 0) { 1259 return; 1260 } 1261 1262 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1263 mSuspendedSessions.editValueAt(index); 1264 1265 for (size_t i = 0; i < sessionEffects.size(); i++) { 1266 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1267 for (int j = 0; j < desc->mRefCount; j++) { 1268 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1269 chain->setEffectSuspendedAll_l(true); 1270 } else { 1271 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1272 desc->mType.timeLow); 1273 chain->setEffectSuspended_l(&desc->mType, true); 1274 } 1275 } 1276 } 1277} 1278 1279void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1280 bool suspend, 1281 int sessionId) 1282{ 1283 int index = mSuspendedSessions.indexOfKey(sessionId); 1284 1285 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1286 1287 if (suspend) { 1288 if (index >= 0) { 1289 sessionEffects = mSuspendedSessions.editValueAt(index); 1290 } else { 1291 mSuspendedSessions.add(sessionId, sessionEffects); 1292 } 1293 } else { 1294 if (index < 0) { 1295 return; 1296 } 1297 sessionEffects = mSuspendedSessions.editValueAt(index); 1298 } 1299 1300 1301 int key = EffectChain::kKeyForSuspendAll; 1302 if (type != NULL) { 1303 key = type->timeLow; 1304 } 1305 index = sessionEffects.indexOfKey(key); 1306 1307 sp <SuspendedSessionDesc> desc; 1308 if (suspend) { 1309 if (index >= 0) { 1310 desc = sessionEffects.valueAt(index); 1311 } else { 1312 desc = new SuspendedSessionDesc(); 1313 if (type != NULL) { 1314 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1315 } 1316 sessionEffects.add(key, desc); 1317 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1318 } 1319 desc->mRefCount++; 1320 } else { 1321 if (index < 0) { 1322 return; 1323 } 1324 desc = sessionEffects.valueAt(index); 1325 if (--desc->mRefCount == 0) { 1326 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1327 sessionEffects.removeItemsAt(index); 1328 if (sessionEffects.isEmpty()) { 1329 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1330 sessionId); 1331 mSuspendedSessions.removeItem(sessionId); 1332 } 1333 } 1334 } 1335 if (!sessionEffects.isEmpty()) { 1336 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1337 } 1338} 1339 1340void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1341 bool enabled, 1342 int sessionId) 1343{ 1344 Mutex::Autolock _l(mLock); 1345 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1346} 1347 1348void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1349 bool enabled, 1350 int sessionId) 1351{ 1352 if (mType != RECORD) { 1353 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1354 // another session. This gives the priority to well behaved effect control panels 1355 // and applications not using global effects. 1356 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1357 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1358 } 1359 } 1360 1361 sp<EffectChain> chain = getEffectChain_l(sessionId); 1362 if (chain != 0) { 1363 chain->checkSuspendOnEffectEnabled(effect, enabled); 1364 } 1365} 1366 1367// ---------------------------------------------------------------------------- 1368 1369AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1370 AudioStreamOut* output, 1371 int id, 1372 uint32_t device) 1373 : ThreadBase(audioFlinger, id, device), 1374 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), mOutput(output), 1375 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1376{ 1377 snprintf(mName, kNameLength, "AudioOut_%d", id); 1378 1379 readOutputParameters(); 1380 1381 // Assumes constructor is called by AudioFlinger with it's mLock held, 1382 // but it would be safer to explicitly pass these as parameters 1383 mMasterVolume = mAudioFlinger->masterVolume_l(); 1384 mMasterMute = mAudioFlinger->masterMute_l(); 1385 1386 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1387 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1388 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1389 stream = (audio_stream_type_t) (stream + 1)) { 1390 mStreamTypes[stream].volume = mAudioFlinger->streamVolumeInternal(stream); 1391 mStreamTypes[stream].mute = mAudioFlinger->streamMute(stream); 1392 // initialized by stream_type_t default constructor 1393 // mStreamTypes[stream].valid = true; 1394 } 1395} 1396 1397AudioFlinger::PlaybackThread::~PlaybackThread() 1398{ 1399 delete [] mMixBuffer; 1400} 1401 1402status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1403{ 1404 dumpInternals(fd, args); 1405 dumpTracks(fd, args); 1406 dumpEffectChains(fd, args); 1407 return NO_ERROR; 1408} 1409 1410status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1411{ 1412 const size_t SIZE = 256; 1413 char buffer[SIZE]; 1414 String8 result; 1415 1416 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1417 result.append(buffer); 1418 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1419 for (size_t i = 0; i < mTracks.size(); ++i) { 1420 sp<Track> track = mTracks[i]; 1421 if (track != 0) { 1422 track->dump(buffer, SIZE); 1423 result.append(buffer); 1424 } 1425 } 1426 1427 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1428 result.append(buffer); 1429 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1430 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1431 sp<Track> track = mActiveTracks[i].promote(); 1432 if (track != 0) { 1433 track->dump(buffer, SIZE); 1434 result.append(buffer); 1435 } 1436 } 1437 write(fd, result.string(), result.size()); 1438 return NO_ERROR; 1439} 1440 1441status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1442{ 1443 const size_t SIZE = 256; 1444 char buffer[SIZE]; 1445 String8 result; 1446 1447 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1448 result.append(buffer); 1449 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1450 result.append(buffer); 1451 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1452 result.append(buffer); 1453 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1454 result.append(buffer); 1455 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1456 result.append(buffer); 1457 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1458 result.append(buffer); 1459 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1460 result.append(buffer); 1461 write(fd, result.string(), result.size()); 1462 1463 dumpBase(fd, args); 1464 1465 return NO_ERROR; 1466} 1467 1468// Thread virtuals 1469status_t AudioFlinger::PlaybackThread::readyToRun() 1470{ 1471 status_t status = initCheck(); 1472 if (status == NO_ERROR) { 1473 ALOGI("AudioFlinger's thread %p ready to run", this); 1474 } else { 1475 ALOGE("No working audio driver found."); 1476 } 1477 return status; 1478} 1479 1480void AudioFlinger::PlaybackThread::onFirstRef() 1481{ 1482 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1483} 1484 1485// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1486sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1487 const sp<AudioFlinger::Client>& client, 1488 audio_stream_type_t streamType, 1489 uint32_t sampleRate, 1490 audio_format_t format, 1491 uint32_t channelMask, 1492 int frameCount, 1493 const sp<IMemory>& sharedBuffer, 1494 int sessionId, 1495 status_t *status) 1496{ 1497 sp<Track> track; 1498 status_t lStatus; 1499 1500 if (mType == DIRECT) { 1501 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1502 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1503 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1504 "for output %p with format %d", 1505 sampleRate, format, channelMask, mOutput, mFormat); 1506 lStatus = BAD_VALUE; 1507 goto Exit; 1508 } 1509 } 1510 } else { 1511 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1512 if (sampleRate > mSampleRate*2) { 1513 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1514 lStatus = BAD_VALUE; 1515 goto Exit; 1516 } 1517 } 1518 1519 lStatus = initCheck(); 1520 if (lStatus != NO_ERROR) { 1521 ALOGE("Audio driver not initialized."); 1522 goto Exit; 1523 } 1524 1525 { // scope for mLock 1526 Mutex::Autolock _l(mLock); 1527 1528 // all tracks in same audio session must share the same routing strategy otherwise 1529 // conflicts will happen when tracks are moved from one output to another by audio policy 1530 // manager 1531 uint32_t strategy = 1532 AudioSystem::getStrategyForStream((audio_stream_type_t)streamType); 1533 for (size_t i = 0; i < mTracks.size(); ++i) { 1534 sp<Track> t = mTracks[i]; 1535 if (t != 0) { 1536 uint32_t actual = AudioSystem::getStrategyForStream((audio_stream_type_t)t->type()); 1537 if (sessionId == t->sessionId() && strategy != actual) { 1538 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1539 strategy, actual); 1540 lStatus = BAD_VALUE; 1541 goto Exit; 1542 } 1543 } 1544 } 1545 1546 track = new Track(this, client, streamType, sampleRate, format, 1547 channelMask, frameCount, sharedBuffer, sessionId); 1548 if (track->getCblk() == NULL || track->name() < 0) { 1549 lStatus = NO_MEMORY; 1550 goto Exit; 1551 } 1552 mTracks.add(track); 1553 1554 sp<EffectChain> chain = getEffectChain_l(sessionId); 1555 if (chain != 0) { 1556 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1557 track->setMainBuffer(chain->inBuffer()); 1558 chain->setStrategy(AudioSystem::getStrategyForStream((audio_stream_type_t)track->type())); 1559 chain->incTrackCnt(); 1560 } 1561 1562 // invalidate track immediately if the stream type was moved to another thread since 1563 // createTrack() was called by the client process. 1564 if (!mStreamTypes[streamType].valid) { 1565 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1566 this, streamType); 1567 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1568 } 1569 } 1570 lStatus = NO_ERROR; 1571 1572Exit: 1573 if(status) { 1574 *status = lStatus; 1575 } 1576 return track; 1577} 1578 1579uint32_t AudioFlinger::PlaybackThread::latency() const 1580{ 1581 Mutex::Autolock _l(mLock); 1582 if (initCheck() == NO_ERROR) { 1583 return mOutput->stream->get_latency(mOutput->stream); 1584 } else { 1585 return 0; 1586 } 1587} 1588 1589status_t AudioFlinger::PlaybackThread::setMasterVolume(float value) 1590{ 1591 mMasterVolume = value; 1592 return NO_ERROR; 1593} 1594 1595status_t AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1596{ 1597 mMasterMute = muted; 1598 return NO_ERROR; 1599} 1600 1601float AudioFlinger::PlaybackThread::masterVolume() const 1602{ 1603 return mMasterVolume; 1604} 1605 1606bool AudioFlinger::PlaybackThread::masterMute() const 1607{ 1608 return mMasterMute; 1609} 1610 1611status_t AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1612{ 1613 mStreamTypes[stream].volume = value; 1614 return NO_ERROR; 1615} 1616 1617status_t AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1618{ 1619 mStreamTypes[stream].mute = muted; 1620 return NO_ERROR; 1621} 1622 1623float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1624{ 1625 return mStreamTypes[stream].volume; 1626} 1627 1628bool AudioFlinger::PlaybackThread::streamMute(audio_stream_type_t stream) const 1629{ 1630 return mStreamTypes[stream].mute; 1631} 1632 1633// addTrack_l() must be called with ThreadBase::mLock held 1634status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1635{ 1636 status_t status = ALREADY_EXISTS; 1637 1638 // set retry count for buffer fill 1639 track->mRetryCount = kMaxTrackStartupRetries; 1640 if (mActiveTracks.indexOf(track) < 0) { 1641 // the track is newly added, make sure it fills up all its 1642 // buffers before playing. This is to ensure the client will 1643 // effectively get the latency it requested. 1644 track->mFillingUpStatus = Track::FS_FILLING; 1645 track->mResetDone = false; 1646 mActiveTracks.add(track); 1647 if (track->mainBuffer() != mMixBuffer) { 1648 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1649 if (chain != 0) { 1650 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1651 chain->incActiveTrackCnt(); 1652 } 1653 } 1654 1655 status = NO_ERROR; 1656 } 1657 1658 ALOGV("mWaitWorkCV.broadcast"); 1659 mWaitWorkCV.broadcast(); 1660 1661 return status; 1662} 1663 1664// destroyTrack_l() must be called with ThreadBase::mLock held 1665void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1666{ 1667 track->mState = TrackBase::TERMINATED; 1668 if (mActiveTracks.indexOf(track) < 0) { 1669 removeTrack_l(track); 1670 } 1671} 1672 1673void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1674{ 1675 mTracks.remove(track); 1676 deleteTrackName_l(track->name()); 1677 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1678 if (chain != 0) { 1679 chain->decTrackCnt(); 1680 } 1681} 1682 1683String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1684{ 1685 String8 out_s8 = String8(""); 1686 char *s; 1687 1688 Mutex::Autolock _l(mLock); 1689 if (initCheck() != NO_ERROR) { 1690 return out_s8; 1691 } 1692 1693 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1694 out_s8 = String8(s); 1695 free(s); 1696 return out_s8; 1697} 1698 1699// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1700void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1701 AudioSystem::OutputDescriptor desc; 1702 void *param2 = 0; 1703 1704 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1705 1706 switch (event) { 1707 case AudioSystem::OUTPUT_OPENED: 1708 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1709 desc.channels = mChannelMask; 1710 desc.samplingRate = mSampleRate; 1711 desc.format = mFormat; 1712 desc.frameCount = mFrameCount; 1713 desc.latency = latency(); 1714 param2 = &desc; 1715 break; 1716 1717 case AudioSystem::STREAM_CONFIG_CHANGED: 1718 param2 = ¶m; 1719 case AudioSystem::OUTPUT_CLOSED: 1720 default: 1721 break; 1722 } 1723 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1724} 1725 1726void AudioFlinger::PlaybackThread::readOutputParameters() 1727{ 1728 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1729 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1730 mChannelCount = (uint16_t)popcount(mChannelMask); 1731 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1732 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1733 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1734 1735 // FIXME - Current mixer implementation only supports stereo output: Always 1736 // Allocate a stereo buffer even if HW output is mono. 1737 if (mMixBuffer != NULL) delete[] mMixBuffer; 1738 mMixBuffer = new int16_t[mFrameCount * 2]; 1739 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1740 1741 // force reconfiguration of effect chains and engines to take new buffer size and audio 1742 // parameters into account 1743 // Note that mLock is not held when readOutputParameters() is called from the constructor 1744 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1745 // matter. 1746 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1747 Vector< sp<EffectChain> > effectChains = mEffectChains; 1748 for (size_t i = 0; i < effectChains.size(); i ++) { 1749 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1750 } 1751} 1752 1753status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1754{ 1755 if (halFrames == 0 || dspFrames == 0) { 1756 return BAD_VALUE; 1757 } 1758 Mutex::Autolock _l(mLock); 1759 if (initCheck() != NO_ERROR) { 1760 return INVALID_OPERATION; 1761 } 1762 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1763 1764 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1765} 1766 1767uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1768{ 1769 Mutex::Autolock _l(mLock); 1770 uint32_t result = 0; 1771 if (getEffectChain_l(sessionId) != 0) { 1772 result = EFFECT_SESSION; 1773 } 1774 1775 for (size_t i = 0; i < mTracks.size(); ++i) { 1776 sp<Track> track = mTracks[i]; 1777 if (sessionId == track->sessionId() && 1778 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1779 result |= TRACK_SESSION; 1780 break; 1781 } 1782 } 1783 1784 return result; 1785} 1786 1787uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1788{ 1789 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1790 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1791 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1792 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1793 } 1794 for (size_t i = 0; i < mTracks.size(); i++) { 1795 sp<Track> track = mTracks[i]; 1796 if (sessionId == track->sessionId() && 1797 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1798 return AudioSystem::getStrategyForStream((audio_stream_type_t) track->type()); 1799 } 1800 } 1801 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1802} 1803 1804 1805AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1806{ 1807 Mutex::Autolock _l(mLock); 1808 return mOutput; 1809} 1810 1811AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1812{ 1813 Mutex::Autolock _l(mLock); 1814 AudioStreamOut *output = mOutput; 1815 mOutput = NULL; 1816 return output; 1817} 1818 1819// this method must always be called either with ThreadBase mLock held or inside the thread loop 1820audio_stream_t* AudioFlinger::PlaybackThread::stream() 1821{ 1822 if (mOutput == NULL) { 1823 return NULL; 1824 } 1825 return &mOutput->stream->common; 1826} 1827 1828uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1829{ 1830 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1831 // decoding and transfer time. So sleeping for half of the latency would likely cause 1832 // underruns 1833 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1834 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1835 } else { 1836 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1837 } 1838} 1839 1840// ---------------------------------------------------------------------------- 1841 1842AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 1843 : PlaybackThread(audioFlinger, output, id, device), 1844 mAudioMixer(NULL), mPrevMixerStatus(MIXER_IDLE) 1845{ 1846 mType = ThreadBase::MIXER; 1847 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1848 1849 // FIXME - Current mixer implementation only supports stereo output 1850 if (mChannelCount == 1) { 1851 ALOGE("Invalid audio hardware channel count"); 1852 } 1853} 1854 1855AudioFlinger::MixerThread::~MixerThread() 1856{ 1857 delete mAudioMixer; 1858} 1859 1860bool AudioFlinger::MixerThread::threadLoop() 1861{ 1862 Vector< sp<Track> > tracksToRemove; 1863 mixer_state mixerStatus = MIXER_IDLE; 1864 nsecs_t standbyTime = systemTime(); 1865 size_t mixBufferSize = mFrameCount * mFrameSize; 1866 // FIXME: Relaxed timing because of a certain device that can't meet latency 1867 // Should be reduced to 2x after the vendor fixes the driver issue 1868 // increase threshold again due to low power audio mode. The way this warning threshold is 1869 // calculated and its usefulness should be reconsidered anyway. 1870 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1871 nsecs_t lastWarning = 0; 1872 bool longStandbyExit = false; 1873 uint32_t activeSleepTime = activeSleepTimeUs(); 1874 uint32_t idleSleepTime = idleSleepTimeUs(); 1875 uint32_t sleepTime = idleSleepTime; 1876 uint32_t sleepTimeShift = 0; 1877 Vector< sp<EffectChain> > effectChains; 1878#ifdef DEBUG_CPU_USAGE 1879 ThreadCpuUsage cpu; 1880 const CentralTendencyStatistics& stats = cpu.statistics(); 1881#endif 1882 1883 acquireWakeLock(); 1884 1885 while (!exitPending()) 1886 { 1887#ifdef DEBUG_CPU_USAGE 1888 cpu.sampleAndEnable(); 1889 unsigned n = stats.n(); 1890 // cpu.elapsed() is expensive, so don't call it every loop 1891 if ((n & 127) == 1) { 1892 long long elapsed = cpu.elapsed(); 1893 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1894 double perLoop = elapsed / (double) n; 1895 double perLoop100 = perLoop * 0.01; 1896 double mean = stats.mean(); 1897 double stddev = stats.stddev(); 1898 double minimum = stats.minimum(); 1899 double maximum = stats.maximum(); 1900 cpu.resetStatistics(); 1901 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1902 elapsed * .000000001, n, perLoop * .000001, 1903 mean * .001, 1904 stddev * .001, 1905 minimum * .001, 1906 maximum * .001, 1907 mean / perLoop100, 1908 stddev / perLoop100, 1909 minimum / perLoop100, 1910 maximum / perLoop100); 1911 } 1912 } 1913#endif 1914 processConfigEvents(); 1915 1916 mixerStatus = MIXER_IDLE; 1917 { // scope for mLock 1918 1919 Mutex::Autolock _l(mLock); 1920 1921 if (checkForNewParameters_l()) { 1922 mixBufferSize = mFrameCount * mFrameSize; 1923 // FIXME: Relaxed timing because of a certain device that can't meet latency 1924 // Should be reduced to 2x after the vendor fixes the driver issue 1925 // increase threshold again due to low power audio mode. The way this warning 1926 // threshold is calculated and its usefulness should be reconsidered anyway. 1927 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 1928 activeSleepTime = activeSleepTimeUs(); 1929 idleSleepTime = idleSleepTimeUs(); 1930 } 1931 1932 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 1933 1934 // put audio hardware into standby after short delay 1935 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 1936 mSuspended)) { 1937 if (!mStandby) { 1938 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d\n", this, mSuspended); 1939 mOutput->stream->common.standby(&mOutput->stream->common); 1940 mStandby = true; 1941 mBytesWritten = 0; 1942 } 1943 1944 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 1945 // we're about to wait, flush the binder command buffer 1946 IPCThreadState::self()->flushCommands(); 1947 1948 if (exitPending()) break; 1949 1950 releaseWakeLock_l(); 1951 // wait until we have something to do... 1952 ALOGV("MixerThread %p TID %d going to sleep\n", this, gettid()); 1953 mWaitWorkCV.wait(mLock); 1954 ALOGV("MixerThread %p TID %d waking up\n", this, gettid()); 1955 acquireWakeLock_l(); 1956 1957 mPrevMixerStatus = MIXER_IDLE; 1958 if (!mMasterMute) { 1959 char value[PROPERTY_VALUE_MAX]; 1960 property_get("ro.audio.silent", value, "0"); 1961 if (atoi(value)) { 1962 ALOGD("Silence is golden"); 1963 setMasterMute(true); 1964 } 1965 } 1966 1967 standbyTime = systemTime() + kStandbyTimeInNsecs; 1968 sleepTime = idleSleepTime; 1969 sleepTimeShift = 0; 1970 continue; 1971 } 1972 } 1973 1974 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 1975 1976 // prevent any changes in effect chain list and in each effect chain 1977 // during mixing and effect process as the audio buffers could be deleted 1978 // or modified if an effect is created or deleted 1979 lockEffectChains_l(effectChains); 1980 } 1981 1982 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 1983 // mix buffers... 1984 mAudioMixer->process(); 1985 // increase sleep time progressively when application underrun condition clears. 1986 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 1987 // that a steady state of alternating ready/not ready conditions keeps the sleep time 1988 // such that we would underrun the audio HAL. 1989 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 1990 sleepTimeShift--; 1991 } 1992 sleepTime = 0; 1993 standbyTime = systemTime() + kStandbyTimeInNsecs; 1994 //TODO: delay standby when effects have a tail 1995 } else { 1996 // If no tracks are ready, sleep once for the duration of an output 1997 // buffer size, then write 0s to the output 1998 if (sleepTime == 0) { 1999 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2000 sleepTime = activeSleepTime >> sleepTimeShift; 2001 if (sleepTime < kMinThreadSleepTimeUs) { 2002 sleepTime = kMinThreadSleepTimeUs; 2003 } 2004 // reduce sleep time in case of consecutive application underruns to avoid 2005 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2006 // duration we would end up writing less data than needed by the audio HAL if 2007 // the condition persists. 2008 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2009 sleepTimeShift++; 2010 } 2011 } else { 2012 sleepTime = idleSleepTime; 2013 } 2014 } else if (mBytesWritten != 0 || 2015 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2016 memset (mMixBuffer, 0, mixBufferSize); 2017 sleepTime = 0; 2018 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2019 } 2020 // TODO add standby time extension fct of effect tail 2021 } 2022 2023 if (mSuspended) { 2024 sleepTime = suspendSleepTimeUs(); 2025 } 2026 // sleepTime == 0 means we must write to audio hardware 2027 if (sleepTime == 0) { 2028 for (size_t i = 0; i < effectChains.size(); i ++) { 2029 effectChains[i]->process_l(); 2030 } 2031 // enable changes in effect chain 2032 unlockEffectChains(effectChains); 2033 mLastWriteTime = systemTime(); 2034 mInWrite = true; 2035 mBytesWritten += mixBufferSize; 2036 2037 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2038 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2039 mNumWrites++; 2040 mInWrite = false; 2041 nsecs_t now = systemTime(); 2042 nsecs_t delta = now - mLastWriteTime; 2043 if (!mStandby && delta > maxPeriod) { 2044 mNumDelayedWrites++; 2045 if ((now - lastWarning) > kWarningThrottleNs) { 2046 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2047 ns2ms(delta), mNumDelayedWrites, this); 2048 lastWarning = now; 2049 } 2050 if (mStandby) { 2051 longStandbyExit = true; 2052 } 2053 } 2054 mStandby = false; 2055 } else { 2056 // enable changes in effect chain 2057 unlockEffectChains(effectChains); 2058 usleep(sleepTime); 2059 } 2060 2061 // finally let go of all our tracks, without the lock held 2062 // since we can't guarantee the destructors won't acquire that 2063 // same lock. 2064 tracksToRemove.clear(); 2065 2066 // Effect chains will be actually deleted here if they were removed from 2067 // mEffectChains list during mixing or effects processing 2068 effectChains.clear(); 2069 } 2070 2071 if (!mStandby) { 2072 mOutput->stream->common.standby(&mOutput->stream->common); 2073 } 2074 2075 releaseWakeLock(); 2076 2077 ALOGV("MixerThread %p exiting", this); 2078 return false; 2079} 2080 2081// prepareTracks_l() must be called with ThreadBase::mLock held 2082AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2083 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2084{ 2085 2086 mixer_state mixerStatus = MIXER_IDLE; 2087 // find out which tracks need to be processed 2088 size_t count = activeTracks.size(); 2089 size_t mixedTracks = 0; 2090 size_t tracksWithEffect = 0; 2091 2092 float masterVolume = mMasterVolume; 2093 bool masterMute = mMasterMute; 2094 2095 if (masterMute) { 2096 masterVolume = 0; 2097 } 2098 // Delegate master volume control to effect in output mix effect chain if needed 2099 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2100 if (chain != 0) { 2101 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2102 chain->setVolume_l(&v, &v); 2103 masterVolume = (float)((v + (1 << 23)) >> 24); 2104 chain.clear(); 2105 } 2106 2107 for (size_t i=0 ; i<count ; i++) { 2108 sp<Track> t = activeTracks[i].promote(); 2109 if (t == 0) continue; 2110 2111 // this const just means the local variable doesn't change 2112 Track* const track = t.get(); 2113 audio_track_cblk_t* cblk = track->cblk(); 2114 2115 // The first time a track is added we wait 2116 // for all its buffers to be filled before processing it 2117 int name = track->name(); 2118 // make sure that we have enough frames to mix one full buffer. 2119 // enforce this condition only once to enable draining the buffer in case the client 2120 // app does not call stop() and relies on underrun to stop: 2121 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2122 // during last round 2123 uint32_t minFrames = 1; 2124 if (!track->isStopped() && !track->isPausing() && 2125 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2126 if (t->sampleRate() == (int)mSampleRate) { 2127 minFrames = mFrameCount; 2128 } else { 2129 // +1 for rounding and +1 for additional sample needed for interpolation 2130 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2131 // add frames already consumed but not yet released by the resampler 2132 // because cblk->framesReady() will include these frames 2133 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2134 // the minimum track buffer size is normally twice the number of frames necessary 2135 // to fill one buffer and the resampler should not leave more than one buffer worth 2136 // of unreleased frames after each pass, but just in case... 2137 ALOG_ASSERT(minFrames <= cblk->frameCount); 2138 } 2139 } 2140 if ((cblk->framesReady() >= minFrames) && track->isReady() && 2141 !track->isPaused() && !track->isTerminated()) 2142 { 2143 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2144 2145 mixedTracks++; 2146 2147 // track->mainBuffer() != mMixBuffer means there is an effect chain 2148 // connected to the track 2149 chain.clear(); 2150 if (track->mainBuffer() != mMixBuffer) { 2151 chain = getEffectChain_l(track->sessionId()); 2152 // Delegate volume control to effect in track effect chain if needed 2153 if (chain != 0) { 2154 tracksWithEffect++; 2155 } else { 2156 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2157 name, track->sessionId()); 2158 } 2159 } 2160 2161 2162 int param = AudioMixer::VOLUME; 2163 if (track->mFillingUpStatus == Track::FS_FILLED) { 2164 // no ramp for the first volume setting 2165 track->mFillingUpStatus = Track::FS_ACTIVE; 2166 if (track->mState == TrackBase::RESUMING) { 2167 track->mState = TrackBase::ACTIVE; 2168 param = AudioMixer::RAMP_VOLUME; 2169 } 2170 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2171 } else if (cblk->server != 0) { 2172 // If the track is stopped before the first frame was mixed, 2173 // do not apply ramp 2174 param = AudioMixer::RAMP_VOLUME; 2175 } 2176 2177 // compute volume for this track 2178 uint32_t vl, vr, va; 2179 if (track->isMuted() || track->isPausing() || 2180 mStreamTypes[track->type()].mute) { 2181 vl = vr = va = 0; 2182 if (track->isPausing()) { 2183 track->setPaused(); 2184 } 2185 } else { 2186 2187 // read original volumes with volume control 2188 float typeVolume = mStreamTypes[track->type()].volume; 2189 float v = masterVolume * typeVolume; 2190 uint32_t vlr = cblk->volumeLR; 2191 vl = vlr & 0xFFFF; 2192 vr = vlr >> 16; 2193 // track volumes come from shared memory, so can't be trusted and must be clamped 2194 if (vl > MAX_GAIN_INT) { 2195 ALOGV("Track left volume out of range: %04X", vl); 2196 vl = MAX_GAIN_INT; 2197 } 2198 if (vr > MAX_GAIN_INT) { 2199 ALOGV("Track right volume out of range: %04X", vr); 2200 vr = MAX_GAIN_INT; 2201 } 2202 // now apply the master volume and stream type volume 2203 vl = (uint32_t)(v * vl) << 12; 2204 vr = (uint32_t)(v * vr) << 12; 2205 // assuming master volume and stream type volume each go up to 1.0, 2206 // vl and vr are now in 8.24 format 2207 2208 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2209 // send level comes from shared memory and so may be corrupt 2210 if (sendLevel >= MAX_GAIN_INT) { 2211 ALOGV("Track send level out of range: %04X", sendLevel); 2212 sendLevel = MAX_GAIN_INT; 2213 } 2214 va = (uint32_t)(v * sendLevel); 2215 } 2216 // Delegate volume control to effect in track effect chain if needed 2217 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2218 // Do not ramp volume if volume is controlled by effect 2219 param = AudioMixer::VOLUME; 2220 track->mHasVolumeController = true; 2221 } else { 2222 // force no volume ramp when volume controller was just disabled or removed 2223 // from effect chain to avoid volume spike 2224 if (track->mHasVolumeController) { 2225 param = AudioMixer::VOLUME; 2226 } 2227 track->mHasVolumeController = false; 2228 } 2229 2230 // Convert volumes from 8.24 to 4.12 format 2231 int16_t left, right, aux; 2232 // This additional clamping is needed in case chain->setVolume_l() overshot 2233 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2234 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2235 left = int16_t(v_clamped); 2236 v_clamped = (vr + (1 << 11)) >> 12; 2237 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2238 right = int16_t(v_clamped); 2239 2240 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; 2241 aux = int16_t(va); 2242 2243 // XXX: these things DON'T need to be done each time 2244 mAudioMixer->setBufferProvider(name, track); 2245 mAudioMixer->enable(name); 2246 2247 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)left); 2248 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)right); 2249 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)aux); 2250 mAudioMixer->setParameter( 2251 name, 2252 AudioMixer::TRACK, 2253 AudioMixer::FORMAT, (void *)track->format()); 2254 mAudioMixer->setParameter( 2255 name, 2256 AudioMixer::TRACK, 2257 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2258 mAudioMixer->setParameter( 2259 name, 2260 AudioMixer::RESAMPLE, 2261 AudioMixer::SAMPLE_RATE, 2262 (void *)(cblk->sampleRate)); 2263 mAudioMixer->setParameter( 2264 name, 2265 AudioMixer::TRACK, 2266 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2267 mAudioMixer->setParameter( 2268 name, 2269 AudioMixer::TRACK, 2270 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2271 2272 // reset retry count 2273 track->mRetryCount = kMaxTrackRetries; 2274 // If one track is ready, set the mixer ready if: 2275 // - the mixer was not ready during previous round OR 2276 // - no other track is not ready 2277 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2278 mixerStatus != MIXER_TRACKS_ENABLED) { 2279 mixerStatus = MIXER_TRACKS_READY; 2280 } 2281 } else { 2282 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2283 if (track->isStopped()) { 2284 track->reset(); 2285 } 2286 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2287 // We have consumed all the buffers of this track. 2288 // Remove it from the list of active tracks. 2289 tracksToRemove->add(track); 2290 } else { 2291 // No buffers for this track. Give it a few chances to 2292 // fill a buffer, then remove it from active list. 2293 if (--(track->mRetryCount) <= 0) { 2294 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2295 tracksToRemove->add(track); 2296 // indicate to client process that the track was disabled because of underrun 2297 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2298 // If one track is not ready, mark the mixer also not ready if: 2299 // - the mixer was ready during previous round OR 2300 // - no other track is ready 2301 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2302 mixerStatus != MIXER_TRACKS_READY) { 2303 mixerStatus = MIXER_TRACKS_ENABLED; 2304 } 2305 } 2306 mAudioMixer->disable(name); 2307 } 2308 } 2309 2310 // remove all the tracks that need to be... 2311 count = tracksToRemove->size(); 2312 if (CC_UNLIKELY(count)) { 2313 for (size_t i=0 ; i<count ; i++) { 2314 const sp<Track>& track = tracksToRemove->itemAt(i); 2315 mActiveTracks.remove(track); 2316 if (track->mainBuffer() != mMixBuffer) { 2317 chain = getEffectChain_l(track->sessionId()); 2318 if (chain != 0) { 2319 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2320 chain->decActiveTrackCnt(); 2321 } 2322 } 2323 if (track->isTerminated()) { 2324 removeTrack_l(track); 2325 } 2326 } 2327 } 2328 2329 // mix buffer must be cleared if all tracks are connected to an 2330 // effect chain as in this case the mixer will not write to 2331 // mix buffer and track effects will accumulate into it 2332 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2333 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2334 } 2335 2336 mPrevMixerStatus = mixerStatus; 2337 return mixerStatus; 2338} 2339 2340void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2341{ 2342 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2343 this, streamType, mTracks.size()); 2344 Mutex::Autolock _l(mLock); 2345 2346 size_t size = mTracks.size(); 2347 for (size_t i = 0; i < size; i++) { 2348 sp<Track> t = mTracks[i]; 2349 if (t->type() == streamType) { 2350 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2351 t->mCblk->cv.signal(); 2352 } 2353 } 2354} 2355 2356void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2357{ 2358 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2359 this, streamType, valid); 2360 Mutex::Autolock _l(mLock); 2361 2362 mStreamTypes[streamType].valid = valid; 2363} 2364 2365// getTrackName_l() must be called with ThreadBase::mLock held 2366int AudioFlinger::MixerThread::getTrackName_l() 2367{ 2368 return mAudioMixer->getTrackName(); 2369} 2370 2371// deleteTrackName_l() must be called with ThreadBase::mLock held 2372void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2373{ 2374 ALOGV("remove track (%d) and delete from mixer", name); 2375 mAudioMixer->deleteTrackName(name); 2376} 2377 2378// checkForNewParameters_l() must be called with ThreadBase::mLock held 2379bool AudioFlinger::MixerThread::checkForNewParameters_l() 2380{ 2381 bool reconfig = false; 2382 2383 while (!mNewParameters.isEmpty()) { 2384 status_t status = NO_ERROR; 2385 String8 keyValuePair = mNewParameters[0]; 2386 AudioParameter param = AudioParameter(keyValuePair); 2387 int value; 2388 2389 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2390 reconfig = true; 2391 } 2392 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2393 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2394 status = BAD_VALUE; 2395 } else { 2396 reconfig = true; 2397 } 2398 } 2399 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2400 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2401 status = BAD_VALUE; 2402 } else { 2403 reconfig = true; 2404 } 2405 } 2406 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2407 // do not accept frame count changes if tracks are open as the track buffer 2408 // size depends on frame count and correct behavior would not be guaranteed 2409 // if frame count is changed after track creation 2410 if (!mTracks.isEmpty()) { 2411 status = INVALID_OPERATION; 2412 } else { 2413 reconfig = true; 2414 } 2415 } 2416 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2417 // when changing the audio output device, call addBatteryData to notify 2418 // the change 2419 if ((int)mDevice != value) { 2420 uint32_t params = 0; 2421 // check whether speaker is on 2422 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2423 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2424 } 2425 2426 int deviceWithoutSpeaker 2427 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2428 // check if any other device (except speaker) is on 2429 if (value & deviceWithoutSpeaker ) { 2430 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2431 } 2432 2433 if (params != 0) { 2434 addBatteryData(params); 2435 } 2436 } 2437 2438 // forward device change to effects that have requested to be 2439 // aware of attached audio device. 2440 mDevice = (uint32_t)value; 2441 for (size_t i = 0; i < mEffectChains.size(); i++) { 2442 mEffectChains[i]->setDevice_l(mDevice); 2443 } 2444 } 2445 2446 if (status == NO_ERROR) { 2447 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2448 keyValuePair.string()); 2449 if (!mStandby && status == INVALID_OPERATION) { 2450 mOutput->stream->common.standby(&mOutput->stream->common); 2451 mStandby = true; 2452 mBytesWritten = 0; 2453 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2454 keyValuePair.string()); 2455 } 2456 if (status == NO_ERROR && reconfig) { 2457 delete mAudioMixer; 2458 readOutputParameters(); 2459 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2460 for (size_t i = 0; i < mTracks.size() ; i++) { 2461 int name = getTrackName_l(); 2462 if (name < 0) break; 2463 mTracks[i]->mName = name; 2464 // limit track sample rate to 2 x new output sample rate 2465 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2466 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2467 } 2468 } 2469 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2470 } 2471 } 2472 2473 mNewParameters.removeAt(0); 2474 2475 mParamStatus = status; 2476 mParamCond.signal(); 2477 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2478 // already timed out waiting for the status and will never signal the condition. 2479 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2480 } 2481 return reconfig; 2482} 2483 2484status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2485{ 2486 const size_t SIZE = 256; 2487 char buffer[SIZE]; 2488 String8 result; 2489 2490 PlaybackThread::dumpInternals(fd, args); 2491 2492 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2493 result.append(buffer); 2494 write(fd, result.string(), result.size()); 2495 return NO_ERROR; 2496} 2497 2498uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2499{ 2500 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2501} 2502 2503uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2504{ 2505 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2506} 2507 2508// ---------------------------------------------------------------------------- 2509AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, int id, uint32_t device) 2510 : PlaybackThread(audioFlinger, output, id, device) 2511{ 2512 mType = ThreadBase::DIRECT; 2513} 2514 2515AudioFlinger::DirectOutputThread::~DirectOutputThread() 2516{ 2517} 2518 2519static inline 2520int32_t mul(int16_t in, int16_t v) 2521{ 2522#if defined(__arm__) && !defined(__thumb__) 2523 int32_t out; 2524 asm( "smulbb %[out], %[in], %[v] \n" 2525 : [out]"=r"(out) 2526 : [in]"%r"(in), [v]"r"(v) 2527 : ); 2528 return out; 2529#else 2530 return in * int32_t(v); 2531#endif 2532} 2533 2534void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2535{ 2536 // Do not apply volume on compressed audio 2537 if (!audio_is_linear_pcm(mFormat)) { 2538 return; 2539 } 2540 2541 // convert to signed 16 bit before volume calculation 2542 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2543 size_t count = mFrameCount * mChannelCount; 2544 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2545 int16_t *dst = mMixBuffer + count-1; 2546 while(count--) { 2547 *dst-- = (int16_t)(*src--^0x80) << 8; 2548 } 2549 } 2550 2551 size_t frameCount = mFrameCount; 2552 int16_t *out = mMixBuffer; 2553 if (ramp) { 2554 if (mChannelCount == 1) { 2555 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2556 int32_t vlInc = d / (int32_t)frameCount; 2557 int32_t vl = ((int32_t)mLeftVolShort << 16); 2558 do { 2559 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2560 out++; 2561 vl += vlInc; 2562 } while (--frameCount); 2563 2564 } else { 2565 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2566 int32_t vlInc = d / (int32_t)frameCount; 2567 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2568 int32_t vrInc = d / (int32_t)frameCount; 2569 int32_t vl = ((int32_t)mLeftVolShort << 16); 2570 int32_t vr = ((int32_t)mRightVolShort << 16); 2571 do { 2572 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2573 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2574 out += 2; 2575 vl += vlInc; 2576 vr += vrInc; 2577 } while (--frameCount); 2578 } 2579 } else { 2580 if (mChannelCount == 1) { 2581 do { 2582 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2583 out++; 2584 } while (--frameCount); 2585 } else { 2586 do { 2587 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2588 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2589 out += 2; 2590 } while (--frameCount); 2591 } 2592 } 2593 2594 // convert back to unsigned 8 bit after volume calculation 2595 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2596 size_t count = mFrameCount * mChannelCount; 2597 int16_t *src = mMixBuffer; 2598 uint8_t *dst = (uint8_t *)mMixBuffer; 2599 while(count--) { 2600 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2601 } 2602 } 2603 2604 mLeftVolShort = leftVol; 2605 mRightVolShort = rightVol; 2606} 2607 2608bool AudioFlinger::DirectOutputThread::threadLoop() 2609{ 2610 mixer_state mixerStatus = MIXER_IDLE; 2611 sp<Track> trackToRemove; 2612 sp<Track> activeTrack; 2613 nsecs_t standbyTime = systemTime(); 2614 int8_t *curBuf; 2615 size_t mixBufferSize = mFrameCount*mFrameSize; 2616 uint32_t activeSleepTime = activeSleepTimeUs(); 2617 uint32_t idleSleepTime = idleSleepTimeUs(); 2618 uint32_t sleepTime = idleSleepTime; 2619 // use shorter standby delay as on normal output to release 2620 // hardware resources as soon as possible 2621 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2622 2623 acquireWakeLock(); 2624 2625 while (!exitPending()) 2626 { 2627 bool rampVolume; 2628 uint16_t leftVol; 2629 uint16_t rightVol; 2630 Vector< sp<EffectChain> > effectChains; 2631 2632 processConfigEvents(); 2633 2634 mixerStatus = MIXER_IDLE; 2635 2636 { // scope for the mLock 2637 2638 Mutex::Autolock _l(mLock); 2639 2640 if (checkForNewParameters_l()) { 2641 mixBufferSize = mFrameCount*mFrameSize; 2642 activeSleepTime = activeSleepTimeUs(); 2643 idleSleepTime = idleSleepTimeUs(); 2644 standbyDelay = microseconds(activeSleepTime*2); 2645 } 2646 2647 // put audio hardware into standby after short delay 2648 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2649 mSuspended)) { 2650 // wait until we have something to do... 2651 if (!mStandby) { 2652 ALOGV("Audio hardware entering standby, mixer %p\n", this); 2653 mOutput->stream->common.standby(&mOutput->stream->common); 2654 mStandby = true; 2655 mBytesWritten = 0; 2656 } 2657 2658 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2659 // we're about to wait, flush the binder command buffer 2660 IPCThreadState::self()->flushCommands(); 2661 2662 if (exitPending()) break; 2663 2664 releaseWakeLock_l(); 2665 ALOGV("DirectOutputThread %p TID %d going to sleep\n", this, gettid()); 2666 mWaitWorkCV.wait(mLock); 2667 ALOGV("DirectOutputThread %p TID %d waking up in active mode\n", this, gettid()); 2668 acquireWakeLock_l(); 2669 2670 if (!mMasterMute) { 2671 char value[PROPERTY_VALUE_MAX]; 2672 property_get("ro.audio.silent", value, "0"); 2673 if (atoi(value)) { 2674 ALOGD("Silence is golden"); 2675 setMasterMute(true); 2676 } 2677 } 2678 2679 standbyTime = systemTime() + standbyDelay; 2680 sleepTime = idleSleepTime; 2681 continue; 2682 } 2683 } 2684 2685 effectChains = mEffectChains; 2686 2687 // find out which tracks need to be processed 2688 if (mActiveTracks.size() != 0) { 2689 sp<Track> t = mActiveTracks[0].promote(); 2690 if (t == 0) continue; 2691 2692 Track* const track = t.get(); 2693 audio_track_cblk_t* cblk = track->cblk(); 2694 2695 // The first time a track is added we wait 2696 // for all its buffers to be filled before processing it 2697 if (cblk->framesReady() && track->isReady() && 2698 !track->isPaused() && !track->isTerminated()) 2699 { 2700 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2701 2702 if (track->mFillingUpStatus == Track::FS_FILLED) { 2703 track->mFillingUpStatus = Track::FS_ACTIVE; 2704 mLeftVolFloat = mRightVolFloat = 0; 2705 mLeftVolShort = mRightVolShort = 0; 2706 if (track->mState == TrackBase::RESUMING) { 2707 track->mState = TrackBase::ACTIVE; 2708 rampVolume = true; 2709 } 2710 } else if (cblk->server != 0) { 2711 // If the track is stopped before the first frame was mixed, 2712 // do not apply ramp 2713 rampVolume = true; 2714 } 2715 // compute volume for this track 2716 float left, right; 2717 if (track->isMuted() || mMasterMute || track->isPausing() || 2718 mStreamTypes[track->type()].mute) { 2719 left = right = 0; 2720 if (track->isPausing()) { 2721 track->setPaused(); 2722 } 2723 } else { 2724 float typeVolume = mStreamTypes[track->type()].volume; 2725 float v = mMasterVolume * typeVolume; 2726 uint32_t vlr = cblk->volumeLR; 2727 float v_clamped = v * (vlr & 0xFFFF); 2728 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2729 left = v_clamped/MAX_GAIN; 2730 v_clamped = v * (vlr >> 16); 2731 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2732 right = v_clamped/MAX_GAIN; 2733 } 2734 2735 if (left != mLeftVolFloat || right != mRightVolFloat) { 2736 mLeftVolFloat = left; 2737 mRightVolFloat = right; 2738 2739 // If audio HAL implements volume control, 2740 // force software volume to nominal value 2741 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2742 left = 1.0f; 2743 right = 1.0f; 2744 } 2745 2746 // Convert volumes from float to 8.24 2747 uint32_t vl = (uint32_t)(left * (1 << 24)); 2748 uint32_t vr = (uint32_t)(right * (1 << 24)); 2749 2750 // Delegate volume control to effect in track effect chain if needed 2751 // only one effect chain can be present on DirectOutputThread, so if 2752 // there is one, the track is connected to it 2753 if (!effectChains.isEmpty()) { 2754 // Do not ramp volume if volume is controlled by effect 2755 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2756 rampVolume = false; 2757 } 2758 } 2759 2760 // Convert volumes from 8.24 to 4.12 format 2761 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2762 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2763 leftVol = (uint16_t)v_clamped; 2764 v_clamped = (vr + (1 << 11)) >> 12; 2765 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2766 rightVol = (uint16_t)v_clamped; 2767 } else { 2768 leftVol = mLeftVolShort; 2769 rightVol = mRightVolShort; 2770 rampVolume = false; 2771 } 2772 2773 // reset retry count 2774 track->mRetryCount = kMaxTrackRetriesDirect; 2775 activeTrack = t; 2776 mixerStatus = MIXER_TRACKS_READY; 2777 } else { 2778 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2779 if (track->isStopped()) { 2780 track->reset(); 2781 } 2782 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2783 // We have consumed all the buffers of this track. 2784 // Remove it from the list of active tracks. 2785 trackToRemove = track; 2786 } else { 2787 // No buffers for this track. Give it a few chances to 2788 // fill a buffer, then remove it from active list. 2789 if (--(track->mRetryCount) <= 0) { 2790 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2791 trackToRemove = track; 2792 } else { 2793 mixerStatus = MIXER_TRACKS_ENABLED; 2794 } 2795 } 2796 } 2797 } 2798 2799 // remove all the tracks that need to be... 2800 if (CC_UNLIKELY(trackToRemove != 0)) { 2801 mActiveTracks.remove(trackToRemove); 2802 if (!effectChains.isEmpty()) { 2803 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2804 trackToRemove->sessionId()); 2805 effectChains[0]->decActiveTrackCnt(); 2806 } 2807 if (trackToRemove->isTerminated()) { 2808 removeTrack_l(trackToRemove); 2809 } 2810 } 2811 2812 lockEffectChains_l(effectChains); 2813 } 2814 2815 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2816 AudioBufferProvider::Buffer buffer; 2817 size_t frameCount = mFrameCount; 2818 curBuf = (int8_t *)mMixBuffer; 2819 // output audio to hardware 2820 while (frameCount) { 2821 buffer.frameCount = frameCount; 2822 activeTrack->getNextBuffer(&buffer); 2823 if (CC_UNLIKELY(buffer.raw == NULL)) { 2824 memset(curBuf, 0, frameCount * mFrameSize); 2825 break; 2826 } 2827 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2828 frameCount -= buffer.frameCount; 2829 curBuf += buffer.frameCount * mFrameSize; 2830 activeTrack->releaseBuffer(&buffer); 2831 } 2832 sleepTime = 0; 2833 standbyTime = systemTime() + standbyDelay; 2834 } else { 2835 if (sleepTime == 0) { 2836 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2837 sleepTime = activeSleepTime; 2838 } else { 2839 sleepTime = idleSleepTime; 2840 } 2841 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2842 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2843 sleepTime = 0; 2844 } 2845 } 2846 2847 if (mSuspended) { 2848 sleepTime = suspendSleepTimeUs(); 2849 } 2850 // sleepTime == 0 means we must write to audio hardware 2851 if (sleepTime == 0) { 2852 if (mixerStatus == MIXER_TRACKS_READY) { 2853 applyVolume(leftVol, rightVol, rampVolume); 2854 } 2855 for (size_t i = 0; i < effectChains.size(); i ++) { 2856 effectChains[i]->process_l(); 2857 } 2858 unlockEffectChains(effectChains); 2859 2860 mLastWriteTime = systemTime(); 2861 mInWrite = true; 2862 mBytesWritten += mixBufferSize; 2863 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2864 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2865 mNumWrites++; 2866 mInWrite = false; 2867 mStandby = false; 2868 } else { 2869 unlockEffectChains(effectChains); 2870 usleep(sleepTime); 2871 } 2872 2873 // finally let go of removed track, without the lock held 2874 // since we can't guarantee the destructors won't acquire that 2875 // same lock. 2876 trackToRemove.clear(); 2877 activeTrack.clear(); 2878 2879 // Effect chains will be actually deleted here if they were removed from 2880 // mEffectChains list during mixing or effects processing 2881 effectChains.clear(); 2882 } 2883 2884 if (!mStandby) { 2885 mOutput->stream->common.standby(&mOutput->stream->common); 2886 } 2887 2888 releaseWakeLock(); 2889 2890 ALOGV("DirectOutputThread %p exiting", this); 2891 return false; 2892} 2893 2894// getTrackName_l() must be called with ThreadBase::mLock held 2895int AudioFlinger::DirectOutputThread::getTrackName_l() 2896{ 2897 return 0; 2898} 2899 2900// deleteTrackName_l() must be called with ThreadBase::mLock held 2901void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2902{ 2903} 2904 2905// checkForNewParameters_l() must be called with ThreadBase::mLock held 2906bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2907{ 2908 bool reconfig = false; 2909 2910 while (!mNewParameters.isEmpty()) { 2911 status_t status = NO_ERROR; 2912 String8 keyValuePair = mNewParameters[0]; 2913 AudioParameter param = AudioParameter(keyValuePair); 2914 int value; 2915 2916 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2917 // do not accept frame count changes if tracks are open as the track buffer 2918 // size depends on frame count and correct behavior would not be garantied 2919 // if frame count is changed after track creation 2920 if (!mTracks.isEmpty()) { 2921 status = INVALID_OPERATION; 2922 } else { 2923 reconfig = true; 2924 } 2925 } 2926 if (status == NO_ERROR) { 2927 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2928 keyValuePair.string()); 2929 if (!mStandby && status == INVALID_OPERATION) { 2930 mOutput->stream->common.standby(&mOutput->stream->common); 2931 mStandby = true; 2932 mBytesWritten = 0; 2933 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2934 keyValuePair.string()); 2935 } 2936 if (status == NO_ERROR && reconfig) { 2937 readOutputParameters(); 2938 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2939 } 2940 } 2941 2942 mNewParameters.removeAt(0); 2943 2944 mParamStatus = status; 2945 mParamCond.signal(); 2946 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2947 // already timed out waiting for the status and will never signal the condition. 2948 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2949 } 2950 return reconfig; 2951} 2952 2953uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 2954{ 2955 uint32_t time; 2956 if (audio_is_linear_pcm(mFormat)) { 2957 time = PlaybackThread::activeSleepTimeUs(); 2958 } else { 2959 time = 10000; 2960 } 2961 return time; 2962} 2963 2964uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 2965{ 2966 uint32_t time; 2967 if (audio_is_linear_pcm(mFormat)) { 2968 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2969 } else { 2970 time = 10000; 2971 } 2972 return time; 2973} 2974 2975uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 2976{ 2977 uint32_t time; 2978 if (audio_is_linear_pcm(mFormat)) { 2979 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2980 } else { 2981 time = 10000; 2982 } 2983 return time; 2984} 2985 2986 2987// ---------------------------------------------------------------------------- 2988 2989AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, AudioFlinger::MixerThread* mainThread, int id) 2990 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device()), mWaitTimeMs(UINT_MAX) 2991{ 2992 mType = ThreadBase::DUPLICATING; 2993 addOutputTrack(mainThread); 2994} 2995 2996AudioFlinger::DuplicatingThread::~DuplicatingThread() 2997{ 2998 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2999 mOutputTracks[i]->destroy(); 3000 } 3001 mOutputTracks.clear(); 3002} 3003 3004bool AudioFlinger::DuplicatingThread::threadLoop() 3005{ 3006 Vector< sp<Track> > tracksToRemove; 3007 mixer_state mixerStatus = MIXER_IDLE; 3008 nsecs_t standbyTime = systemTime(); 3009 size_t mixBufferSize = mFrameCount*mFrameSize; 3010 SortedVector< sp<OutputTrack> > outputTracks; 3011 uint32_t writeFrames = 0; 3012 uint32_t activeSleepTime = activeSleepTimeUs(); 3013 uint32_t idleSleepTime = idleSleepTimeUs(); 3014 uint32_t sleepTime = idleSleepTime; 3015 Vector< sp<EffectChain> > effectChains; 3016 3017 acquireWakeLock(); 3018 3019 while (!exitPending()) 3020 { 3021 processConfigEvents(); 3022 3023 mixerStatus = MIXER_IDLE; 3024 { // scope for the mLock 3025 3026 Mutex::Autolock _l(mLock); 3027 3028 if (checkForNewParameters_l()) { 3029 mixBufferSize = mFrameCount*mFrameSize; 3030 updateWaitTime(); 3031 activeSleepTime = activeSleepTimeUs(); 3032 idleSleepTime = idleSleepTimeUs(); 3033 } 3034 3035 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3036 3037 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3038 outputTracks.add(mOutputTracks[i]); 3039 } 3040 3041 // put audio hardware into standby after short delay 3042 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3043 mSuspended)) { 3044 if (!mStandby) { 3045 for (size_t i = 0; i < outputTracks.size(); i++) { 3046 outputTracks[i]->stop(); 3047 } 3048 mStandby = true; 3049 mBytesWritten = 0; 3050 } 3051 3052 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3053 // we're about to wait, flush the binder command buffer 3054 IPCThreadState::self()->flushCommands(); 3055 outputTracks.clear(); 3056 3057 if (exitPending()) break; 3058 3059 releaseWakeLock_l(); 3060 ALOGV("DuplicatingThread %p TID %d going to sleep\n", this, gettid()); 3061 mWaitWorkCV.wait(mLock); 3062 ALOGV("DuplicatingThread %p TID %d waking up\n", this, gettid()); 3063 acquireWakeLock_l(); 3064 3065 mPrevMixerStatus = MIXER_IDLE; 3066 if (!mMasterMute) { 3067 char value[PROPERTY_VALUE_MAX]; 3068 property_get("ro.audio.silent", value, "0"); 3069 if (atoi(value)) { 3070 ALOGD("Silence is golden"); 3071 setMasterMute(true); 3072 } 3073 } 3074 3075 standbyTime = systemTime() + kStandbyTimeInNsecs; 3076 sleepTime = idleSleepTime; 3077 continue; 3078 } 3079 } 3080 3081 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3082 3083 // prevent any changes in effect chain list and in each effect chain 3084 // during mixing and effect process as the audio buffers could be deleted 3085 // or modified if an effect is created or deleted 3086 lockEffectChains_l(effectChains); 3087 } 3088 3089 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3090 // mix buffers... 3091 if (outputsReady(outputTracks)) { 3092 mAudioMixer->process(); 3093 } else { 3094 memset(mMixBuffer, 0, mixBufferSize); 3095 } 3096 sleepTime = 0; 3097 writeFrames = mFrameCount; 3098 } else { 3099 if (sleepTime == 0) { 3100 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3101 sleepTime = activeSleepTime; 3102 } else { 3103 sleepTime = idleSleepTime; 3104 } 3105 } else if (mBytesWritten != 0) { 3106 // flush remaining overflow buffers in output tracks 3107 for (size_t i = 0; i < outputTracks.size(); i++) { 3108 if (outputTracks[i]->isActive()) { 3109 sleepTime = 0; 3110 writeFrames = 0; 3111 memset(mMixBuffer, 0, mixBufferSize); 3112 break; 3113 } 3114 } 3115 } 3116 } 3117 3118 if (mSuspended) { 3119 sleepTime = suspendSleepTimeUs(); 3120 } 3121 // sleepTime == 0 means we must write to audio hardware 3122 if (sleepTime == 0) { 3123 for (size_t i = 0; i < effectChains.size(); i ++) { 3124 effectChains[i]->process_l(); 3125 } 3126 // enable changes in effect chain 3127 unlockEffectChains(effectChains); 3128 3129 standbyTime = systemTime() + kStandbyTimeInNsecs; 3130 for (size_t i = 0; i < outputTracks.size(); i++) { 3131 outputTracks[i]->write(mMixBuffer, writeFrames); 3132 } 3133 mStandby = false; 3134 mBytesWritten += mixBufferSize; 3135 } else { 3136 // enable changes in effect chain 3137 unlockEffectChains(effectChains); 3138 usleep(sleepTime); 3139 } 3140 3141 // finally let go of all our tracks, without the lock held 3142 // since we can't guarantee the destructors won't acquire that 3143 // same lock. 3144 tracksToRemove.clear(); 3145 outputTracks.clear(); 3146 3147 // Effect chains will be actually deleted here if they were removed from 3148 // mEffectChains list during mixing or effects processing 3149 effectChains.clear(); 3150 } 3151 3152 releaseWakeLock(); 3153 3154 return false; 3155} 3156 3157void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3158{ 3159 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3160 OutputTrack *outputTrack = new OutputTrack((ThreadBase *)thread, 3161 this, 3162 mSampleRate, 3163 mFormat, 3164 mChannelMask, 3165 frameCount); 3166 if (outputTrack->cblk() != NULL) { 3167 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3168 mOutputTracks.add(outputTrack); 3169 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3170 updateWaitTime(); 3171 } 3172} 3173 3174void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3175{ 3176 Mutex::Autolock _l(mLock); 3177 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3178 if (mOutputTracks[i]->thread() == (ThreadBase *)thread) { 3179 mOutputTracks[i]->destroy(); 3180 mOutputTracks.removeAt(i); 3181 updateWaitTime(); 3182 return; 3183 } 3184 } 3185 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3186} 3187 3188void AudioFlinger::DuplicatingThread::updateWaitTime() 3189{ 3190 mWaitTimeMs = UINT_MAX; 3191 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3192 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3193 if (strong != NULL) { 3194 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3195 if (waitTimeMs < mWaitTimeMs) { 3196 mWaitTimeMs = waitTimeMs; 3197 } 3198 } 3199 } 3200} 3201 3202 3203bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3204{ 3205 for (size_t i = 0; i < outputTracks.size(); i++) { 3206 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3207 if (thread == 0) { 3208 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3209 return false; 3210 } 3211 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3212 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3213 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3214 return false; 3215 } 3216 } 3217 return true; 3218} 3219 3220uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3221{ 3222 return (mWaitTimeMs * 1000) / 2; 3223} 3224 3225// ---------------------------------------------------------------------------- 3226 3227// TrackBase constructor must be called with AudioFlinger::mLock held 3228AudioFlinger::ThreadBase::TrackBase::TrackBase( 3229 const wp<ThreadBase>& thread, 3230 const sp<Client>& client, 3231 uint32_t sampleRate, 3232 audio_format_t format, 3233 uint32_t channelMask, 3234 int frameCount, 3235 uint32_t flags, 3236 const sp<IMemory>& sharedBuffer, 3237 int sessionId) 3238 : RefBase(), 3239 mThread(thread), 3240 mClient(client), 3241 mCblk(0), 3242 mFrameCount(0), 3243 mState(IDLE), 3244 mClientTid(-1), 3245 mFormat(format), 3246 mFlags(flags & ~SYSTEM_FLAGS_MASK), 3247 mSessionId(sessionId) 3248{ 3249 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3250 3251 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3252 size_t size = sizeof(audio_track_cblk_t); 3253 uint8_t channelCount = popcount(channelMask); 3254 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3255 if (sharedBuffer == 0) { 3256 size += bufferSize; 3257 } 3258 3259 if (client != NULL) { 3260 mCblkMemory = client->heap()->allocate(size); 3261 if (mCblkMemory != 0) { 3262 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3263 if (mCblk) { // construct the shared structure in-place. 3264 new(mCblk) audio_track_cblk_t(); 3265 // clear all buffers 3266 mCblk->frameCount = frameCount; 3267 mCblk->sampleRate = sampleRate; 3268 mChannelCount = channelCount; 3269 mChannelMask = channelMask; 3270 if (sharedBuffer == 0) { 3271 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3272 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3273 // Force underrun condition to avoid false underrun callback until first data is 3274 // written to buffer (other flags are cleared) 3275 mCblk->flags = CBLK_UNDERRUN_ON; 3276 } else { 3277 mBuffer = sharedBuffer->pointer(); 3278 } 3279 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3280 } 3281 } else { 3282 ALOGE("not enough memory for AudioTrack size=%u", size); 3283 client->heap()->dump("AudioTrack"); 3284 return; 3285 } 3286 } else { 3287 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3288 // construct the shared structure in-place. 3289 new(mCblk) audio_track_cblk_t(); 3290 // clear all buffers 3291 mCblk->frameCount = frameCount; 3292 mCblk->sampleRate = sampleRate; 3293 mChannelCount = channelCount; 3294 mChannelMask = channelMask; 3295 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3296 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3297 // Force underrun condition to avoid false underrun callback until first data is 3298 // written to buffer (other flags are cleared) 3299 mCblk->flags = CBLK_UNDERRUN_ON; 3300 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3301 } 3302} 3303 3304AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3305{ 3306 if (mCblk) { 3307 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3308 if (mClient == NULL) { 3309 delete mCblk; 3310 } 3311 } 3312 mCblkMemory.clear(); // and free the shared memory 3313 if (mClient != NULL) { 3314 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3315 mClient.clear(); 3316 } 3317} 3318 3319void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3320{ 3321 buffer->raw = NULL; 3322 mFrameCount = buffer->frameCount; 3323 step(); 3324 buffer->frameCount = 0; 3325} 3326 3327bool AudioFlinger::ThreadBase::TrackBase::step() { 3328 bool result; 3329 audio_track_cblk_t* cblk = this->cblk(); 3330 3331 result = cblk->stepServer(mFrameCount); 3332 if (!result) { 3333 ALOGV("stepServer failed acquiring cblk mutex"); 3334 mFlags |= STEPSERVER_FAILED; 3335 } 3336 return result; 3337} 3338 3339void AudioFlinger::ThreadBase::TrackBase::reset() { 3340 audio_track_cblk_t* cblk = this->cblk(); 3341 3342 cblk->user = 0; 3343 cblk->server = 0; 3344 cblk->userBase = 0; 3345 cblk->serverBase = 0; 3346 mFlags &= (uint32_t)(~SYSTEM_FLAGS_MASK); 3347 ALOGV("TrackBase::reset"); 3348} 3349 3350sp<IMemory> AudioFlinger::ThreadBase::TrackBase::getCblk() const 3351{ 3352 return mCblkMemory; 3353} 3354 3355int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3356 return (int)mCblk->sampleRate; 3357} 3358 3359int AudioFlinger::ThreadBase::TrackBase::channelCount() const { 3360 return (const int)mChannelCount; 3361} 3362 3363uint32_t AudioFlinger::ThreadBase::TrackBase::channelMask() const { 3364 return mChannelMask; 3365} 3366 3367void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3368 audio_track_cblk_t* cblk = this->cblk(); 3369 size_t frameSize = cblk->frameSize; 3370 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3371 int8_t *bufferEnd = bufferStart + frames * frameSize; 3372 3373 // Check validity of returned pointer in case the track control block would have been corrupted. 3374 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3375 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3376 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3377 server %d, serverBase %d, user %d, userBase %d", 3378 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3379 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3380 return 0; 3381 } 3382 3383 return bufferStart; 3384} 3385 3386// ---------------------------------------------------------------------------- 3387 3388// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3389AudioFlinger::PlaybackThread::Track::Track( 3390 const wp<ThreadBase>& thread, 3391 const sp<Client>& client, 3392 audio_stream_type_t streamType, 3393 uint32_t sampleRate, 3394 audio_format_t format, 3395 uint32_t channelMask, 3396 int frameCount, 3397 const sp<IMemory>& sharedBuffer, 3398 int sessionId) 3399 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, 0, sharedBuffer, sessionId), 3400 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3401 mAuxEffectId(0), mHasVolumeController(false) 3402{ 3403 if (mCblk != NULL) { 3404 sp<ThreadBase> baseThread = thread.promote(); 3405 if (baseThread != 0) { 3406 PlaybackThread *playbackThread = (PlaybackThread *)baseThread.get(); 3407 mName = playbackThread->getTrackName_l(); 3408 mMainBuffer = playbackThread->mixBuffer(); 3409 } 3410 ALOGV("Track constructor name %d, calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3411 if (mName < 0) { 3412 ALOGE("no more track names available"); 3413 } 3414 mStreamType = streamType; 3415 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3416 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3417 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3418 } 3419} 3420 3421AudioFlinger::PlaybackThread::Track::~Track() 3422{ 3423 ALOGV("PlaybackThread::Track destructor"); 3424 sp<ThreadBase> thread = mThread.promote(); 3425 if (thread != 0) { 3426 Mutex::Autolock _l(thread->mLock); 3427 mState = TERMINATED; 3428 } 3429} 3430 3431void AudioFlinger::PlaybackThread::Track::destroy() 3432{ 3433 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3434 // by removing it from mTracks vector, so there is a risk that this Tracks's 3435 // desctructor is called. As the destructor needs to lock mLock, 3436 // we must acquire a strong reference on this Track before locking mLock 3437 // here so that the destructor is called only when exiting this function. 3438 // On the other hand, as long as Track::destroy() is only called by 3439 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3440 // this Track with its member mTrack. 3441 sp<Track> keep(this); 3442 { // scope for mLock 3443 sp<ThreadBase> thread = mThread.promote(); 3444 if (thread != 0) { 3445 if (!isOutputTrack()) { 3446 if (mState == ACTIVE || mState == RESUMING) { 3447 AudioSystem::stopOutput(thread->id(), 3448 (audio_stream_type_t)mStreamType, 3449 mSessionId); 3450 3451 // to track the speaker usage 3452 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3453 } 3454 AudioSystem::releaseOutput(thread->id()); 3455 } 3456 Mutex::Autolock _l(thread->mLock); 3457 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3458 playbackThread->destroyTrack_l(this); 3459 } 3460 } 3461} 3462 3463void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3464{ 3465 uint32_t vlr = mCblk->volumeLR; 3466 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", 3467 mName - AudioMixer::TRACK0, 3468 (mClient == NULL) ? getpid() : mClient->pid(), 3469 mStreamType, 3470 mFormat, 3471 mChannelMask, 3472 mSessionId, 3473 mFrameCount, 3474 mState, 3475 mMute, 3476 mFillingUpStatus, 3477 mCblk->sampleRate, 3478 vlr & 0xFFFF, 3479 vlr >> 16, 3480 mCblk->server, 3481 mCblk->user, 3482 (int)mMainBuffer, 3483 (int)mAuxBuffer); 3484} 3485 3486status_t AudioFlinger::PlaybackThread::Track::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3487{ 3488 audio_track_cblk_t* cblk = this->cblk(); 3489 uint32_t framesReady; 3490 uint32_t framesReq = buffer->frameCount; 3491 3492 // Check if last stepServer failed, try to step now 3493 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3494 if (!step()) goto getNextBuffer_exit; 3495 ALOGV("stepServer recovered"); 3496 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3497 } 3498 3499 framesReady = cblk->framesReady(); 3500 3501 if (CC_LIKELY(framesReady)) { 3502 uint32_t s = cblk->server; 3503 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3504 3505 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3506 if (framesReq > framesReady) { 3507 framesReq = framesReady; 3508 } 3509 if (s + framesReq > bufferEnd) { 3510 framesReq = bufferEnd - s; 3511 } 3512 3513 buffer->raw = getBuffer(s, framesReq); 3514 if (buffer->raw == NULL) goto getNextBuffer_exit; 3515 3516 buffer->frameCount = framesReq; 3517 return NO_ERROR; 3518 } 3519 3520getNextBuffer_exit: 3521 buffer->raw = NULL; 3522 buffer->frameCount = 0; 3523 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3524 return NOT_ENOUGH_DATA; 3525} 3526 3527bool AudioFlinger::PlaybackThread::Track::isReady() const { 3528 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3529 3530 if (mCblk->framesReady() >= mCblk->frameCount || 3531 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3532 mFillingUpStatus = FS_FILLED; 3533 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3534 return true; 3535 } 3536 return false; 3537} 3538 3539status_t AudioFlinger::PlaybackThread::Track::start() 3540{ 3541 status_t status = NO_ERROR; 3542 ALOGV("start(%d), calling thread %d session %d", 3543 mName, IPCThreadState::self()->getCallingPid(), mSessionId); 3544 sp<ThreadBase> thread = mThread.promote(); 3545 if (thread != 0) { 3546 Mutex::Autolock _l(thread->mLock); 3547 track_state state = mState; 3548 // here the track could be either new, or restarted 3549 // in both cases "unstop" the track 3550 if (mState == PAUSED) { 3551 mState = TrackBase::RESUMING; 3552 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3553 } else { 3554 mState = TrackBase::ACTIVE; 3555 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3556 } 3557 3558 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3559 thread->mLock.unlock(); 3560 status = AudioSystem::startOutput(thread->id(), 3561 (audio_stream_type_t)mStreamType, 3562 mSessionId); 3563 thread->mLock.lock(); 3564 3565 // to track the speaker usage 3566 if (status == NO_ERROR) { 3567 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3568 } 3569 } 3570 if (status == NO_ERROR) { 3571 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3572 playbackThread->addTrack_l(this); 3573 } else { 3574 mState = state; 3575 } 3576 } else { 3577 status = BAD_VALUE; 3578 } 3579 return status; 3580} 3581 3582void AudioFlinger::PlaybackThread::Track::stop() 3583{ 3584 ALOGV("stop(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3585 sp<ThreadBase> thread = mThread.promote(); 3586 if (thread != 0) { 3587 Mutex::Autolock _l(thread->mLock); 3588 track_state state = mState; 3589 if (mState > STOPPED) { 3590 mState = STOPPED; 3591 // If the track is not active (PAUSED and buffers full), flush buffers 3592 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3593 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3594 reset(); 3595 } 3596 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3597 } 3598 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3599 thread->mLock.unlock(); 3600 AudioSystem::stopOutput(thread->id(), 3601 (audio_stream_type_t)mStreamType, 3602 mSessionId); 3603 thread->mLock.lock(); 3604 3605 // to track the speaker usage 3606 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3607 } 3608 } 3609} 3610 3611void AudioFlinger::PlaybackThread::Track::pause() 3612{ 3613 ALOGV("pause(%d), calling thread %d", mName, IPCThreadState::self()->getCallingPid()); 3614 sp<ThreadBase> thread = mThread.promote(); 3615 if (thread != 0) { 3616 Mutex::Autolock _l(thread->mLock); 3617 if (mState == ACTIVE || mState == RESUMING) { 3618 mState = PAUSING; 3619 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3620 if (!isOutputTrack()) { 3621 thread->mLock.unlock(); 3622 AudioSystem::stopOutput(thread->id(), 3623 (audio_stream_type_t)mStreamType, 3624 mSessionId); 3625 thread->mLock.lock(); 3626 3627 // to track the speaker usage 3628 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3629 } 3630 } 3631 } 3632} 3633 3634void AudioFlinger::PlaybackThread::Track::flush() 3635{ 3636 ALOGV("flush(%d)", mName); 3637 sp<ThreadBase> thread = mThread.promote(); 3638 if (thread != 0) { 3639 Mutex::Autolock _l(thread->mLock); 3640 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3641 return; 3642 } 3643 // No point remaining in PAUSED state after a flush => go to 3644 // STOPPED state 3645 mState = STOPPED; 3646 3647 // do not reset the track if it is still in the process of being stopped or paused. 3648 // this will be done by prepareTracks_l() when the track is stopped. 3649 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3650 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3651 reset(); 3652 } 3653 } 3654} 3655 3656void AudioFlinger::PlaybackThread::Track::reset() 3657{ 3658 // Do not reset twice to avoid discarding data written just after a flush and before 3659 // the audioflinger thread detects the track is stopped. 3660 if (!mResetDone) { 3661 TrackBase::reset(); 3662 // Force underrun condition to avoid false underrun callback until first data is 3663 // written to buffer 3664 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3665 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3666 mFillingUpStatus = FS_FILLING; 3667 mResetDone = true; 3668 } 3669} 3670 3671void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3672{ 3673 mMute = muted; 3674} 3675 3676status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3677{ 3678 status_t status = DEAD_OBJECT; 3679 sp<ThreadBase> thread = mThread.promote(); 3680 if (thread != 0) { 3681 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3682 status = playbackThread->attachAuxEffect(this, EffectId); 3683 } 3684 return status; 3685} 3686 3687void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3688{ 3689 mAuxEffectId = EffectId; 3690 mAuxBuffer = buffer; 3691} 3692 3693// ---------------------------------------------------------------------------- 3694 3695// RecordTrack constructor must be called with AudioFlinger::mLock held 3696AudioFlinger::RecordThread::RecordTrack::RecordTrack( 3697 const wp<ThreadBase>& thread, 3698 const sp<Client>& client, 3699 uint32_t sampleRate, 3700 audio_format_t format, 3701 uint32_t channelMask, 3702 int frameCount, 3703 uint32_t flags, 3704 int sessionId) 3705 : TrackBase(thread, client, sampleRate, format, 3706 channelMask, frameCount, flags, 0, sessionId), 3707 mOverflow(false) 3708{ 3709 if (mCblk != NULL) { 3710 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 3711 if (format == AUDIO_FORMAT_PCM_16_BIT) { 3712 mCblk->frameSize = mChannelCount * sizeof(int16_t); 3713 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 3714 mCblk->frameSize = mChannelCount * sizeof(int8_t); 3715 } else { 3716 mCblk->frameSize = sizeof(int8_t); 3717 } 3718 } 3719} 3720 3721AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 3722{ 3723 sp<ThreadBase> thread = mThread.promote(); 3724 if (thread != 0) { 3725 AudioSystem::releaseInput(thread->id()); 3726 } 3727} 3728 3729status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer) 3730{ 3731 audio_track_cblk_t* cblk = this->cblk(); 3732 uint32_t framesAvail; 3733 uint32_t framesReq = buffer->frameCount; 3734 3735 // Check if last stepServer failed, try to step now 3736 if (mFlags & TrackBase::STEPSERVER_FAILED) { 3737 if (!step()) goto getNextBuffer_exit; 3738 ALOGV("stepServer recovered"); 3739 mFlags &= ~TrackBase::STEPSERVER_FAILED; 3740 } 3741 3742 framesAvail = cblk->framesAvailable_l(); 3743 3744 if (CC_LIKELY(framesAvail)) { 3745 uint32_t s = cblk->server; 3746 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3747 3748 if (framesReq > framesAvail) { 3749 framesReq = framesAvail; 3750 } 3751 if (s + framesReq > bufferEnd) { 3752 framesReq = bufferEnd - s; 3753 } 3754 3755 buffer->raw = getBuffer(s, framesReq); 3756 if (buffer->raw == NULL) goto getNextBuffer_exit; 3757 3758 buffer->frameCount = framesReq; 3759 return NO_ERROR; 3760 } 3761 3762getNextBuffer_exit: 3763 buffer->raw = NULL; 3764 buffer->frameCount = 0; 3765 return NOT_ENOUGH_DATA; 3766} 3767 3768status_t AudioFlinger::RecordThread::RecordTrack::start() 3769{ 3770 sp<ThreadBase> thread = mThread.promote(); 3771 if (thread != 0) { 3772 RecordThread *recordThread = (RecordThread *)thread.get(); 3773 return recordThread->start(this); 3774 } else { 3775 return BAD_VALUE; 3776 } 3777} 3778 3779void AudioFlinger::RecordThread::RecordTrack::stop() 3780{ 3781 sp<ThreadBase> thread = mThread.promote(); 3782 if (thread != 0) { 3783 RecordThread *recordThread = (RecordThread *)thread.get(); 3784 recordThread->stop(this); 3785 TrackBase::reset(); 3786 // Force overerrun condition to avoid false overrun callback until first data is 3787 // read from buffer 3788 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3789 } 3790} 3791 3792void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 3793{ 3794 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 3795 (mClient == NULL) ? getpid() : mClient->pid(), 3796 mFormat, 3797 mChannelMask, 3798 mSessionId, 3799 mFrameCount, 3800 mState, 3801 mCblk->sampleRate, 3802 mCblk->server, 3803 mCblk->user); 3804} 3805 3806 3807// ---------------------------------------------------------------------------- 3808 3809AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 3810 const wp<ThreadBase>& thread, 3811 DuplicatingThread *sourceThread, 3812 uint32_t sampleRate, 3813 audio_format_t format, 3814 uint32_t channelMask, 3815 int frameCount) 3816 : Track(thread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 3817 mActive(false), mSourceThread(sourceThread) 3818{ 3819 3820 PlaybackThread *playbackThread = (PlaybackThread *)thread.unsafe_get(); 3821 if (mCblk != NULL) { 3822 mCblk->flags |= CBLK_DIRECTION_OUT; 3823 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 3824 mCblk->volumeLR = (MAX_GAIN_INT << 16) | MAX_GAIN_INT; 3825 mOutBuffer.frameCount = 0; 3826 playbackThread->mTracks.add(this); 3827 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 3828 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 3829 mCblk, mBuffer, mCblk->buffers, 3830 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 3831 } else { 3832 ALOGW("Error creating output track on thread %p", playbackThread); 3833 } 3834} 3835 3836AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 3837{ 3838 clearBufferQueue(); 3839} 3840 3841status_t AudioFlinger::PlaybackThread::OutputTrack::start() 3842{ 3843 status_t status = Track::start(); 3844 if (status != NO_ERROR) { 3845 return status; 3846 } 3847 3848 mActive = true; 3849 mRetryCount = 127; 3850 return status; 3851} 3852 3853void AudioFlinger::PlaybackThread::OutputTrack::stop() 3854{ 3855 Track::stop(); 3856 clearBufferQueue(); 3857 mOutBuffer.frameCount = 0; 3858 mActive = false; 3859} 3860 3861bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 3862{ 3863 Buffer *pInBuffer; 3864 Buffer inBuffer; 3865 uint32_t channelCount = mChannelCount; 3866 bool outputBufferFull = false; 3867 inBuffer.frameCount = frames; 3868 inBuffer.i16 = data; 3869 3870 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 3871 3872 if (!mActive && frames != 0) { 3873 start(); 3874 sp<ThreadBase> thread = mThread.promote(); 3875 if (thread != 0) { 3876 MixerThread *mixerThread = (MixerThread *)thread.get(); 3877 if (mCblk->frameCount > frames){ 3878 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3879 uint32_t startFrames = (mCblk->frameCount - frames); 3880 pInBuffer = new Buffer; 3881 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 3882 pInBuffer->frameCount = startFrames; 3883 pInBuffer->i16 = pInBuffer->mBuffer; 3884 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 3885 mBufferQueue.add(pInBuffer); 3886 } else { 3887 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 3888 } 3889 } 3890 } 3891 } 3892 3893 while (waitTimeLeftMs) { 3894 // First write pending buffers, then new data 3895 if (mBufferQueue.size()) { 3896 pInBuffer = mBufferQueue.itemAt(0); 3897 } else { 3898 pInBuffer = &inBuffer; 3899 } 3900 3901 if (pInBuffer->frameCount == 0) { 3902 break; 3903 } 3904 3905 if (mOutBuffer.frameCount == 0) { 3906 mOutBuffer.frameCount = pInBuffer->frameCount; 3907 nsecs_t startTime = systemTime(); 3908 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 3909 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 3910 outputBufferFull = true; 3911 break; 3912 } 3913 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 3914 if (waitTimeLeftMs >= waitTimeMs) { 3915 waitTimeLeftMs -= waitTimeMs; 3916 } else { 3917 waitTimeLeftMs = 0; 3918 } 3919 } 3920 3921 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 3922 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 3923 mCblk->stepUser(outFrames); 3924 pInBuffer->frameCount -= outFrames; 3925 pInBuffer->i16 += outFrames * channelCount; 3926 mOutBuffer.frameCount -= outFrames; 3927 mOutBuffer.i16 += outFrames * channelCount; 3928 3929 if (pInBuffer->frameCount == 0) { 3930 if (mBufferQueue.size()) { 3931 mBufferQueue.removeAt(0); 3932 delete [] pInBuffer->mBuffer; 3933 delete pInBuffer; 3934 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3935 } else { 3936 break; 3937 } 3938 } 3939 } 3940 3941 // If we could not write all frames, allocate a buffer and queue it for next time. 3942 if (inBuffer.frameCount) { 3943 sp<ThreadBase> thread = mThread.promote(); 3944 if (thread != 0 && !thread->standby()) { 3945 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 3946 pInBuffer = new Buffer; 3947 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 3948 pInBuffer->frameCount = inBuffer.frameCount; 3949 pInBuffer->i16 = pInBuffer->mBuffer; 3950 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 3951 mBufferQueue.add(pInBuffer); 3952 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 3953 } else { 3954 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 3955 } 3956 } 3957 } 3958 3959 // Calling write() with a 0 length buffer, means that no more data will be written: 3960 // If no more buffers are pending, fill output track buffer to make sure it is started 3961 // by output mixer. 3962 if (frames == 0 && mBufferQueue.size() == 0) { 3963 if (mCblk->user < mCblk->frameCount) { 3964 frames = mCblk->frameCount - mCblk->user; 3965 pInBuffer = new Buffer; 3966 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 3967 pInBuffer->frameCount = frames; 3968 pInBuffer->i16 = pInBuffer->mBuffer; 3969 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 3970 mBufferQueue.add(pInBuffer); 3971 } else if (mActive) { 3972 stop(); 3973 } 3974 } 3975 3976 return outputBufferFull; 3977} 3978 3979status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 3980{ 3981 int active; 3982 status_t result; 3983 audio_track_cblk_t* cblk = mCblk; 3984 uint32_t framesReq = buffer->frameCount; 3985 3986// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 3987 buffer->frameCount = 0; 3988 3989 uint32_t framesAvail = cblk->framesAvailable(); 3990 3991 3992 if (framesAvail == 0) { 3993 Mutex::Autolock _l(cblk->lock); 3994 goto start_loop_here; 3995 while (framesAvail == 0) { 3996 active = mActive; 3997 if (CC_UNLIKELY(!active)) { 3998 ALOGV("Not active and NO_MORE_BUFFERS"); 3999 return NO_MORE_BUFFERS; 4000 } 4001 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4002 if (result != NO_ERROR) { 4003 return NO_MORE_BUFFERS; 4004 } 4005 // read the server count again 4006 start_loop_here: 4007 framesAvail = cblk->framesAvailable_l(); 4008 } 4009 } 4010 4011// if (framesAvail < framesReq) { 4012// return NO_MORE_BUFFERS; 4013// } 4014 4015 if (framesReq > framesAvail) { 4016 framesReq = framesAvail; 4017 } 4018 4019 uint32_t u = cblk->user; 4020 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4021 4022 if (u + framesReq > bufferEnd) { 4023 framesReq = bufferEnd - u; 4024 } 4025 4026 buffer->frameCount = framesReq; 4027 buffer->raw = (void *)cblk->buffer(u); 4028 return NO_ERROR; 4029} 4030 4031 4032void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4033{ 4034 size_t size = mBufferQueue.size(); 4035 Buffer *pBuffer; 4036 4037 for (size_t i = 0; i < size; i++) { 4038 pBuffer = mBufferQueue.itemAt(i); 4039 delete [] pBuffer->mBuffer; 4040 delete pBuffer; 4041 } 4042 mBufferQueue.clear(); 4043} 4044 4045// ---------------------------------------------------------------------------- 4046 4047AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4048 : RefBase(), 4049 mAudioFlinger(audioFlinger), 4050 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4051 mPid(pid) 4052{ 4053 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4054} 4055 4056// Client destructor must be called with AudioFlinger::mLock held 4057AudioFlinger::Client::~Client() 4058{ 4059 mAudioFlinger->removeClient_l(mPid); 4060} 4061 4062const sp<MemoryDealer>& AudioFlinger::Client::heap() const 4063{ 4064 return mMemoryDealer; 4065} 4066 4067// ---------------------------------------------------------------------------- 4068 4069AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4070 const sp<IAudioFlingerClient>& client, 4071 pid_t pid) 4072 : mAudioFlinger(audioFlinger), mPid(pid), mClient(client) 4073{ 4074} 4075 4076AudioFlinger::NotificationClient::~NotificationClient() 4077{ 4078 mClient.clear(); 4079} 4080 4081void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4082{ 4083 sp<NotificationClient> keep(this); 4084 { 4085 mAudioFlinger->removeNotificationClient(mPid); 4086 } 4087} 4088 4089// ---------------------------------------------------------------------------- 4090 4091AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4092 : BnAudioTrack(), 4093 mTrack(track) 4094{ 4095} 4096 4097AudioFlinger::TrackHandle::~TrackHandle() { 4098 // just stop the track on deletion, associated resources 4099 // will be freed from the main thread once all pending buffers have 4100 // been played. Unless it's not in the active track list, in which 4101 // case we free everything now... 4102 mTrack->destroy(); 4103} 4104 4105sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4106 return mTrack->getCblk(); 4107} 4108 4109status_t AudioFlinger::TrackHandle::start() { 4110 return mTrack->start(); 4111} 4112 4113void AudioFlinger::TrackHandle::stop() { 4114 mTrack->stop(); 4115} 4116 4117void AudioFlinger::TrackHandle::flush() { 4118 mTrack->flush(); 4119} 4120 4121void AudioFlinger::TrackHandle::mute(bool e) { 4122 mTrack->mute(e); 4123} 4124 4125void AudioFlinger::TrackHandle::pause() { 4126 mTrack->pause(); 4127} 4128 4129status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4130{ 4131 return mTrack->attachAuxEffect(EffectId); 4132} 4133 4134status_t AudioFlinger::TrackHandle::onTransact( 4135 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4136{ 4137 return BnAudioTrack::onTransact(code, data, reply, flags); 4138} 4139 4140// ---------------------------------------------------------------------------- 4141 4142sp<IAudioRecord> AudioFlinger::openRecord( 4143 pid_t pid, 4144 int input, 4145 uint32_t sampleRate, 4146 audio_format_t format, 4147 uint32_t channelMask, 4148 int frameCount, 4149 uint32_t flags, 4150 int *sessionId, 4151 status_t *status) 4152{ 4153 sp<RecordThread::RecordTrack> recordTrack; 4154 sp<RecordHandle> recordHandle; 4155 sp<Client> client; 4156 wp<Client> wclient; 4157 status_t lStatus; 4158 RecordThread *thread; 4159 size_t inFrameCount; 4160 int lSessionId; 4161 4162 // check calling permissions 4163 if (!recordingAllowed()) { 4164 lStatus = PERMISSION_DENIED; 4165 goto Exit; 4166 } 4167 4168 // add client to list 4169 { // scope for mLock 4170 Mutex::Autolock _l(mLock); 4171 thread = checkRecordThread_l(input); 4172 if (thread == NULL) { 4173 lStatus = BAD_VALUE; 4174 goto Exit; 4175 } 4176 4177 wclient = mClients.valueFor(pid); 4178 if (wclient != NULL) { 4179 client = wclient.promote(); 4180 } else { 4181 client = new Client(this, pid); 4182 mClients.add(pid, client); 4183 } 4184 4185 // If no audio session id is provided, create one here 4186 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4187 lSessionId = *sessionId; 4188 } else { 4189 lSessionId = nextUniqueId(); 4190 if (sessionId != NULL) { 4191 *sessionId = lSessionId; 4192 } 4193 } 4194 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4195 recordTrack = thread->createRecordTrack_l(client, 4196 sampleRate, 4197 format, 4198 channelMask, 4199 frameCount, 4200 flags, 4201 lSessionId, 4202 &lStatus); 4203 } 4204 if (lStatus != NO_ERROR) { 4205 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4206 // destructor is called by the TrackBase destructor with mLock held 4207 client.clear(); 4208 recordTrack.clear(); 4209 goto Exit; 4210 } 4211 4212 // return to handle to client 4213 recordHandle = new RecordHandle(recordTrack); 4214 lStatus = NO_ERROR; 4215 4216Exit: 4217 if (status) { 4218 *status = lStatus; 4219 } 4220 return recordHandle; 4221} 4222 4223// ---------------------------------------------------------------------------- 4224 4225AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4226 : BnAudioRecord(), 4227 mRecordTrack(recordTrack) 4228{ 4229} 4230 4231AudioFlinger::RecordHandle::~RecordHandle() { 4232 stop(); 4233} 4234 4235sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4236 return mRecordTrack->getCblk(); 4237} 4238 4239status_t AudioFlinger::RecordHandle::start() { 4240 ALOGV("RecordHandle::start()"); 4241 return mRecordTrack->start(); 4242} 4243 4244void AudioFlinger::RecordHandle::stop() { 4245 ALOGV("RecordHandle::stop()"); 4246 mRecordTrack->stop(); 4247} 4248 4249status_t AudioFlinger::RecordHandle::onTransact( 4250 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4251{ 4252 return BnAudioRecord::onTransact(code, data, reply, flags); 4253} 4254 4255// ---------------------------------------------------------------------------- 4256 4257AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4258 AudioStreamIn *input, 4259 uint32_t sampleRate, 4260 uint32_t channels, 4261 int id, 4262 uint32_t device) : 4263 ThreadBase(audioFlinger, id, device), 4264 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL) 4265{ 4266 mType = ThreadBase::RECORD; 4267 4268 snprintf(mName, kNameLength, "AudioIn_%d", id); 4269 4270 mReqChannelCount = popcount(channels); 4271 mReqSampleRate = sampleRate; 4272 readInputParameters(); 4273} 4274 4275 4276AudioFlinger::RecordThread::~RecordThread() 4277{ 4278 delete[] mRsmpInBuffer; 4279 if (mResampler != NULL) { 4280 delete mResampler; 4281 delete[] mRsmpOutBuffer; 4282 } 4283} 4284 4285void AudioFlinger::RecordThread::onFirstRef() 4286{ 4287 run(mName, PRIORITY_URGENT_AUDIO); 4288} 4289 4290status_t AudioFlinger::RecordThread::readyToRun() 4291{ 4292 status_t status = initCheck(); 4293 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4294 return status; 4295} 4296 4297bool AudioFlinger::RecordThread::threadLoop() 4298{ 4299 AudioBufferProvider::Buffer buffer; 4300 sp<RecordTrack> activeTrack; 4301 Vector< sp<EffectChain> > effectChains; 4302 4303 nsecs_t lastWarning = 0; 4304 4305 acquireWakeLock(); 4306 4307 // start recording 4308 while (!exitPending()) { 4309 4310 processConfigEvents(); 4311 4312 { // scope for mLock 4313 Mutex::Autolock _l(mLock); 4314 checkForNewParameters_l(); 4315 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4316 if (!mStandby) { 4317 mInput->stream->common.standby(&mInput->stream->common); 4318 mStandby = true; 4319 } 4320 4321 if (exitPending()) break; 4322 4323 releaseWakeLock_l(); 4324 ALOGV("RecordThread: loop stopping"); 4325 // go to sleep 4326 mWaitWorkCV.wait(mLock); 4327 ALOGV("RecordThread: loop starting"); 4328 acquireWakeLock_l(); 4329 continue; 4330 } 4331 if (mActiveTrack != 0) { 4332 if (mActiveTrack->mState == TrackBase::PAUSING) { 4333 if (!mStandby) { 4334 mInput->stream->common.standby(&mInput->stream->common); 4335 mStandby = true; 4336 } 4337 mActiveTrack.clear(); 4338 mStartStopCond.broadcast(); 4339 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4340 if (mReqChannelCount != mActiveTrack->channelCount()) { 4341 mActiveTrack.clear(); 4342 mStartStopCond.broadcast(); 4343 } else if (mBytesRead != 0) { 4344 // record start succeeds only if first read from audio input 4345 // succeeds 4346 if (mBytesRead > 0) { 4347 mActiveTrack->mState = TrackBase::ACTIVE; 4348 } else { 4349 mActiveTrack.clear(); 4350 } 4351 mStartStopCond.broadcast(); 4352 } 4353 mStandby = false; 4354 } 4355 } 4356 lockEffectChains_l(effectChains); 4357 } 4358 4359 if (mActiveTrack != 0) { 4360 if (mActiveTrack->mState != TrackBase::ACTIVE && 4361 mActiveTrack->mState != TrackBase::RESUMING) { 4362 unlockEffectChains(effectChains); 4363 usleep(kRecordThreadSleepUs); 4364 continue; 4365 } 4366 for (size_t i = 0; i < effectChains.size(); i ++) { 4367 effectChains[i]->process_l(); 4368 } 4369 4370 buffer.frameCount = mFrameCount; 4371 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4372 size_t framesOut = buffer.frameCount; 4373 if (mResampler == NULL) { 4374 // no resampling 4375 while (framesOut) { 4376 size_t framesIn = mFrameCount - mRsmpInIndex; 4377 if (framesIn) { 4378 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4379 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4380 if (framesIn > framesOut) 4381 framesIn = framesOut; 4382 mRsmpInIndex += framesIn; 4383 framesOut -= framesIn; 4384 if ((int)mChannelCount == mReqChannelCount || 4385 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4386 memcpy(dst, src, framesIn * mFrameSize); 4387 } else { 4388 int16_t *src16 = (int16_t *)src; 4389 int16_t *dst16 = (int16_t *)dst; 4390 if (mChannelCount == 1) { 4391 while (framesIn--) { 4392 *dst16++ = *src16; 4393 *dst16++ = *src16++; 4394 } 4395 } else { 4396 while (framesIn--) { 4397 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4398 src16 += 2; 4399 } 4400 } 4401 } 4402 } 4403 if (framesOut && mFrameCount == mRsmpInIndex) { 4404 if (framesOut == mFrameCount && 4405 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4406 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4407 framesOut = 0; 4408 } else { 4409 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4410 mRsmpInIndex = 0; 4411 } 4412 if (mBytesRead < 0) { 4413 ALOGE("Error reading audio input"); 4414 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4415 // Force input into standby so that it tries to 4416 // recover at next read attempt 4417 mInput->stream->common.standby(&mInput->stream->common); 4418 usleep(kRecordThreadSleepUs); 4419 } 4420 mRsmpInIndex = mFrameCount; 4421 framesOut = 0; 4422 buffer.frameCount = 0; 4423 } 4424 } 4425 } 4426 } else { 4427 // resampling 4428 4429 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4430 // alter output frame count as if we were expecting stereo samples 4431 if (mChannelCount == 1 && mReqChannelCount == 1) { 4432 framesOut >>= 1; 4433 } 4434 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4435 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4436 // are 32 bit aligned which should be always true. 4437 if (mChannelCount == 2 && mReqChannelCount == 1) { 4438 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4439 // the resampler always outputs stereo samples: do post stereo to mono conversion 4440 int16_t *src = (int16_t *)mRsmpOutBuffer; 4441 int16_t *dst = buffer.i16; 4442 while (framesOut--) { 4443 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4444 src += 2; 4445 } 4446 } else { 4447 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4448 } 4449 4450 } 4451 mActiveTrack->releaseBuffer(&buffer); 4452 mActiveTrack->overflow(); 4453 } 4454 // client isn't retrieving buffers fast enough 4455 else { 4456 if (!mActiveTrack->setOverflow()) { 4457 nsecs_t now = systemTime(); 4458 if ((now - lastWarning) > kWarningThrottleNs) { 4459 ALOGW("RecordThread: buffer overflow"); 4460 lastWarning = now; 4461 } 4462 } 4463 // Release the processor for a while before asking for a new buffer. 4464 // This will give the application more chance to read from the buffer and 4465 // clear the overflow. 4466 usleep(kRecordThreadSleepUs); 4467 } 4468 } 4469 // enable changes in effect chain 4470 unlockEffectChains(effectChains); 4471 effectChains.clear(); 4472 } 4473 4474 if (!mStandby) { 4475 mInput->stream->common.standby(&mInput->stream->common); 4476 } 4477 mActiveTrack.clear(); 4478 4479 mStartStopCond.broadcast(); 4480 4481 releaseWakeLock(); 4482 4483 ALOGV("RecordThread %p exiting", this); 4484 return false; 4485} 4486 4487 4488sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4489 const sp<AudioFlinger::Client>& client, 4490 uint32_t sampleRate, 4491 audio_format_t format, 4492 int channelMask, 4493 int frameCount, 4494 uint32_t flags, 4495 int sessionId, 4496 status_t *status) 4497{ 4498 sp<RecordTrack> track; 4499 status_t lStatus; 4500 4501 lStatus = initCheck(); 4502 if (lStatus != NO_ERROR) { 4503 ALOGE("Audio driver not initialized."); 4504 goto Exit; 4505 } 4506 4507 { // scope for mLock 4508 Mutex::Autolock _l(mLock); 4509 4510 track = new RecordTrack(this, client, sampleRate, 4511 format, channelMask, frameCount, flags, sessionId); 4512 4513 if (track->getCblk() == NULL) { 4514 lStatus = NO_MEMORY; 4515 goto Exit; 4516 } 4517 4518 mTrack = track.get(); 4519 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4520 bool suspend = audio_is_bluetooth_sco_device( 4521 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4522 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4523 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4524 } 4525 lStatus = NO_ERROR; 4526 4527Exit: 4528 if (status) { 4529 *status = lStatus; 4530 } 4531 return track; 4532} 4533 4534status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack) 4535{ 4536 ALOGV("RecordThread::start"); 4537 sp <ThreadBase> strongMe = this; 4538 status_t status = NO_ERROR; 4539 { 4540 AutoMutex lock(mLock); 4541 if (mActiveTrack != 0) { 4542 if (recordTrack != mActiveTrack.get()) { 4543 status = -EBUSY; 4544 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4545 mActiveTrack->mState = TrackBase::ACTIVE; 4546 } 4547 return status; 4548 } 4549 4550 recordTrack->mState = TrackBase::IDLE; 4551 mActiveTrack = recordTrack; 4552 mLock.unlock(); 4553 status_t status = AudioSystem::startInput(mId); 4554 mLock.lock(); 4555 if (status != NO_ERROR) { 4556 mActiveTrack.clear(); 4557 return status; 4558 } 4559 mRsmpInIndex = mFrameCount; 4560 mBytesRead = 0; 4561 if (mResampler != NULL) { 4562 mResampler->reset(); 4563 } 4564 mActiveTrack->mState = TrackBase::RESUMING; 4565 // signal thread to start 4566 ALOGV("Signal record thread"); 4567 mWaitWorkCV.signal(); 4568 // do not wait for mStartStopCond if exiting 4569 if (mExiting) { 4570 mActiveTrack.clear(); 4571 status = INVALID_OPERATION; 4572 goto startError; 4573 } 4574 mStartStopCond.wait(mLock); 4575 if (mActiveTrack == 0) { 4576 ALOGV("Record failed to start"); 4577 status = BAD_VALUE; 4578 goto startError; 4579 } 4580 ALOGV("Record started OK"); 4581 return status; 4582 } 4583startError: 4584 AudioSystem::stopInput(mId); 4585 return status; 4586} 4587 4588void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 4589 ALOGV("RecordThread::stop"); 4590 sp <ThreadBase> strongMe = this; 4591 { 4592 AutoMutex lock(mLock); 4593 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 4594 mActiveTrack->mState = TrackBase::PAUSING; 4595 // do not wait for mStartStopCond if exiting 4596 if (mExiting) { 4597 return; 4598 } 4599 mStartStopCond.wait(mLock); 4600 // if we have been restarted, recordTrack == mActiveTrack.get() here 4601 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 4602 mLock.unlock(); 4603 AudioSystem::stopInput(mId); 4604 mLock.lock(); 4605 ALOGV("Record stopped OK"); 4606 } 4607 } 4608 } 4609} 4610 4611status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 4612{ 4613 const size_t SIZE = 256; 4614 char buffer[SIZE]; 4615 String8 result; 4616 pid_t pid = 0; 4617 4618 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 4619 result.append(buffer); 4620 4621 if (mActiveTrack != 0) { 4622 result.append("Active Track:\n"); 4623 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 4624 mActiveTrack->dump(buffer, SIZE); 4625 result.append(buffer); 4626 4627 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 4628 result.append(buffer); 4629 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 4630 result.append(buffer); 4631 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 4632 result.append(buffer); 4633 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 4634 result.append(buffer); 4635 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 4636 result.append(buffer); 4637 4638 4639 } else { 4640 result.append("No record client\n"); 4641 } 4642 write(fd, result.string(), result.size()); 4643 4644 dumpBase(fd, args); 4645 dumpEffectChains(fd, args); 4646 4647 return NO_ERROR; 4648} 4649 4650status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer) 4651{ 4652 size_t framesReq = buffer->frameCount; 4653 size_t framesReady = mFrameCount - mRsmpInIndex; 4654 int channelCount; 4655 4656 if (framesReady == 0) { 4657 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4658 if (mBytesRead < 0) { 4659 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 4660 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4661 // Force input into standby so that it tries to 4662 // recover at next read attempt 4663 mInput->stream->common.standby(&mInput->stream->common); 4664 usleep(kRecordThreadSleepUs); 4665 } 4666 buffer->raw = NULL; 4667 buffer->frameCount = 0; 4668 return NOT_ENOUGH_DATA; 4669 } 4670 mRsmpInIndex = 0; 4671 framesReady = mFrameCount; 4672 } 4673 4674 if (framesReq > framesReady) { 4675 framesReq = framesReady; 4676 } 4677 4678 if (mChannelCount == 1 && mReqChannelCount == 2) { 4679 channelCount = 1; 4680 } else { 4681 channelCount = 2; 4682 } 4683 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 4684 buffer->frameCount = framesReq; 4685 return NO_ERROR; 4686} 4687 4688void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 4689{ 4690 mRsmpInIndex += buffer->frameCount; 4691 buffer->frameCount = 0; 4692} 4693 4694bool AudioFlinger::RecordThread::checkForNewParameters_l() 4695{ 4696 bool reconfig = false; 4697 4698 while (!mNewParameters.isEmpty()) { 4699 status_t status = NO_ERROR; 4700 String8 keyValuePair = mNewParameters[0]; 4701 AudioParameter param = AudioParameter(keyValuePair); 4702 int value; 4703 audio_format_t reqFormat = mFormat; 4704 int reqSamplingRate = mReqSampleRate; 4705 int reqChannelCount = mReqChannelCount; 4706 4707 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 4708 reqSamplingRate = value; 4709 reconfig = true; 4710 } 4711 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 4712 reqFormat = (audio_format_t) value; 4713 reconfig = true; 4714 } 4715 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 4716 reqChannelCount = popcount(value); 4717 reconfig = true; 4718 } 4719 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 4720 // do not accept frame count changes if tracks are open as the track buffer 4721 // size depends on frame count and correct behavior would not be garantied 4722 // if frame count is changed after track creation 4723 if (mActiveTrack != 0) { 4724 status = INVALID_OPERATION; 4725 } else { 4726 reconfig = true; 4727 } 4728 } 4729 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 4730 // forward device change to effects that have requested to be 4731 // aware of attached audio device. 4732 for (size_t i = 0; i < mEffectChains.size(); i++) { 4733 mEffectChains[i]->setDevice_l(value); 4734 } 4735 // store input device and output device but do not forward output device to audio HAL. 4736 // Note that status is ignored by the caller for output device 4737 // (see AudioFlinger::setParameters() 4738 if (value & AUDIO_DEVICE_OUT_ALL) { 4739 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 4740 status = BAD_VALUE; 4741 } else { 4742 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 4743 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4744 if (mTrack != NULL) { 4745 bool suspend = audio_is_bluetooth_sco_device( 4746 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 4747 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 4748 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 4749 } 4750 } 4751 mDevice |= (uint32_t)value; 4752 } 4753 if (status == NO_ERROR) { 4754 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4755 if (status == INVALID_OPERATION) { 4756 mInput->stream->common.standby(&mInput->stream->common); 4757 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 4758 } 4759 if (reconfig) { 4760 if (status == BAD_VALUE && 4761 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 4762 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 4763 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 4764 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 4765 (reqChannelCount < 3)) { 4766 status = NO_ERROR; 4767 } 4768 if (status == NO_ERROR) { 4769 readInputParameters(); 4770 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 4771 } 4772 } 4773 } 4774 4775 mNewParameters.removeAt(0); 4776 4777 mParamStatus = status; 4778 mParamCond.signal(); 4779 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 4780 // already timed out waiting for the status and will never signal the condition. 4781 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 4782 } 4783 return reconfig; 4784} 4785 4786String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 4787{ 4788 char *s; 4789 String8 out_s8 = String8(); 4790 4791 Mutex::Autolock _l(mLock); 4792 if (initCheck() != NO_ERROR) { 4793 return out_s8; 4794 } 4795 4796 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 4797 out_s8 = String8(s); 4798 free(s); 4799 return out_s8; 4800} 4801 4802void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 4803 AudioSystem::OutputDescriptor desc; 4804 void *param2 = 0; 4805 4806 switch (event) { 4807 case AudioSystem::INPUT_OPENED: 4808 case AudioSystem::INPUT_CONFIG_CHANGED: 4809 desc.channels = mChannelMask; 4810 desc.samplingRate = mSampleRate; 4811 desc.format = mFormat; 4812 desc.frameCount = mFrameCount; 4813 desc.latency = 0; 4814 param2 = &desc; 4815 break; 4816 4817 case AudioSystem::INPUT_CLOSED: 4818 default: 4819 break; 4820 } 4821 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 4822} 4823 4824void AudioFlinger::RecordThread::readInputParameters() 4825{ 4826 if (mRsmpInBuffer) delete mRsmpInBuffer; 4827 if (mRsmpOutBuffer) delete mRsmpOutBuffer; 4828 if (mResampler) delete mResampler; 4829 mResampler = NULL; 4830 4831 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 4832 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 4833 mChannelCount = (uint16_t)popcount(mChannelMask); 4834 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 4835 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 4836 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 4837 mFrameCount = mInputBytes / mFrameSize; 4838 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 4839 4840 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 4841 { 4842 int channelCount; 4843 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 4844 // stereo to mono post process as the resampler always outputs stereo. 4845 if (mChannelCount == 1 && mReqChannelCount == 2) { 4846 channelCount = 1; 4847 } else { 4848 channelCount = 2; 4849 } 4850 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 4851 mResampler->setSampleRate(mSampleRate); 4852 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 4853 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 4854 4855 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 4856 if (mChannelCount == 1 && mReqChannelCount == 1) { 4857 mFrameCount >>= 1; 4858 } 4859 4860 } 4861 mRsmpInIndex = mFrameCount; 4862} 4863 4864unsigned int AudioFlinger::RecordThread::getInputFramesLost() 4865{ 4866 Mutex::Autolock _l(mLock); 4867 if (initCheck() != NO_ERROR) { 4868 return 0; 4869 } 4870 4871 return mInput->stream->get_input_frames_lost(mInput->stream); 4872} 4873 4874uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 4875{ 4876 Mutex::Autolock _l(mLock); 4877 uint32_t result = 0; 4878 if (getEffectChain_l(sessionId) != 0) { 4879 result = EFFECT_SESSION; 4880 } 4881 4882 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 4883 result |= TRACK_SESSION; 4884 } 4885 4886 return result; 4887} 4888 4889AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 4890{ 4891 Mutex::Autolock _l(mLock); 4892 return mTrack; 4893} 4894 4895AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 4896{ 4897 Mutex::Autolock _l(mLock); 4898 return mInput; 4899} 4900 4901AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 4902{ 4903 Mutex::Autolock _l(mLock); 4904 AudioStreamIn *input = mInput; 4905 mInput = NULL; 4906 return input; 4907} 4908 4909// this method must always be called either with ThreadBase mLock held or inside the thread loop 4910audio_stream_t* AudioFlinger::RecordThread::stream() 4911{ 4912 if (mInput == NULL) { 4913 return NULL; 4914 } 4915 return &mInput->stream->common; 4916} 4917 4918 4919// ---------------------------------------------------------------------------- 4920 4921int AudioFlinger::openOutput(uint32_t *pDevices, 4922 uint32_t *pSamplingRate, 4923 audio_format_t *pFormat, 4924 uint32_t *pChannels, 4925 uint32_t *pLatencyMs, 4926 uint32_t flags) 4927{ 4928 status_t status; 4929 PlaybackThread *thread = NULL; 4930 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 4931 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 4932 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 4933 uint32_t channels = pChannels ? *pChannels : 0; 4934 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 4935 audio_stream_out_t *outStream; 4936 audio_hw_device_t *outHwDev; 4937 4938 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 4939 pDevices ? *pDevices : 0, 4940 samplingRate, 4941 format, 4942 channels, 4943 flags); 4944 4945 if (pDevices == NULL || *pDevices == 0) { 4946 return 0; 4947 } 4948 4949 Mutex::Autolock _l(mLock); 4950 4951 outHwDev = findSuitableHwDev_l(*pDevices); 4952 if (outHwDev == NULL) 4953 return 0; 4954 4955 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 4956 &channels, &samplingRate, &outStream); 4957 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 4958 outStream, 4959 samplingRate, 4960 format, 4961 channels, 4962 status); 4963 4964 mHardwareStatus = AUDIO_HW_IDLE; 4965 if (outStream != NULL) { 4966 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 4967 int id = nextUniqueId(); 4968 4969 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 4970 (format != AUDIO_FORMAT_PCM_16_BIT) || 4971 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 4972 thread = new DirectOutputThread(this, output, id, *pDevices); 4973 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 4974 } else { 4975 thread = new MixerThread(this, output, id, *pDevices); 4976 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 4977 } 4978 mPlaybackThreads.add(id, thread); 4979 4980 if (pSamplingRate) *pSamplingRate = samplingRate; 4981 if (pFormat) *pFormat = format; 4982 if (pChannels) *pChannels = channels; 4983 if (pLatencyMs) *pLatencyMs = thread->latency(); 4984 4985 // notify client processes of the new output creation 4986 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 4987 return id; 4988 } 4989 4990 return 0; 4991} 4992 4993int AudioFlinger::openDuplicateOutput(int output1, int output2) 4994{ 4995 Mutex::Autolock _l(mLock); 4996 MixerThread *thread1 = checkMixerThread_l(output1); 4997 MixerThread *thread2 = checkMixerThread_l(output2); 4998 4999 if (thread1 == NULL || thread2 == NULL) { 5000 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5001 return 0; 5002 } 5003 5004 int id = nextUniqueId(); 5005 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5006 thread->addOutputTrack(thread2); 5007 mPlaybackThreads.add(id, thread); 5008 // notify client processes of the new output creation 5009 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5010 return id; 5011} 5012 5013status_t AudioFlinger::closeOutput(int output) 5014{ 5015 // keep strong reference on the playback thread so that 5016 // it is not destroyed while exit() is executed 5017 sp <PlaybackThread> thread; 5018 { 5019 Mutex::Autolock _l(mLock); 5020 thread = checkPlaybackThread_l(output); 5021 if (thread == NULL) { 5022 return BAD_VALUE; 5023 } 5024 5025 ALOGV("closeOutput() %d", output); 5026 5027 if (thread->type() == ThreadBase::MIXER) { 5028 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5029 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5030 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5031 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5032 } 5033 } 5034 } 5035 void *param2 = 0; 5036 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, param2); 5037 mPlaybackThreads.removeItem(output); 5038 } 5039 thread->exit(); 5040 5041 if (thread->type() != ThreadBase::DUPLICATING) { 5042 AudioStreamOut *out = thread->clearOutput(); 5043 assert(out != NULL); 5044 // from now on thread->mOutput is NULL 5045 out->hwDev->close_output_stream(out->hwDev, out->stream); 5046 delete out; 5047 } 5048 return NO_ERROR; 5049} 5050 5051status_t AudioFlinger::suspendOutput(int output) 5052{ 5053 Mutex::Autolock _l(mLock); 5054 PlaybackThread *thread = checkPlaybackThread_l(output); 5055 5056 if (thread == NULL) { 5057 return BAD_VALUE; 5058 } 5059 5060 ALOGV("suspendOutput() %d", output); 5061 thread->suspend(); 5062 5063 return NO_ERROR; 5064} 5065 5066status_t AudioFlinger::restoreOutput(int output) 5067{ 5068 Mutex::Autolock _l(mLock); 5069 PlaybackThread *thread = checkPlaybackThread_l(output); 5070 5071 if (thread == NULL) { 5072 return BAD_VALUE; 5073 } 5074 5075 ALOGV("restoreOutput() %d", output); 5076 5077 thread->restore(); 5078 5079 return NO_ERROR; 5080} 5081 5082int AudioFlinger::openInput(uint32_t *pDevices, 5083 uint32_t *pSamplingRate, 5084 audio_format_t *pFormat, 5085 uint32_t *pChannels, 5086 uint32_t acoustics) 5087{ 5088 status_t status; 5089 RecordThread *thread = NULL; 5090 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5091 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5092 uint32_t channels = pChannels ? *pChannels : 0; 5093 uint32_t reqSamplingRate = samplingRate; 5094 audio_format_t reqFormat = format; 5095 uint32_t reqChannels = channels; 5096 audio_stream_in_t *inStream; 5097 audio_hw_device_t *inHwDev; 5098 5099 if (pDevices == NULL || *pDevices == 0) { 5100 return 0; 5101 } 5102 5103 Mutex::Autolock _l(mLock); 5104 5105 inHwDev = findSuitableHwDev_l(*pDevices); 5106 if (inHwDev == NULL) 5107 return 0; 5108 5109 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5110 &channels, &samplingRate, 5111 (audio_in_acoustics_t)acoustics, 5112 &inStream); 5113 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5114 inStream, 5115 samplingRate, 5116 format, 5117 channels, 5118 acoustics, 5119 status); 5120 5121 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5122 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5123 // or stereo to mono conversions on 16 bit PCM inputs. 5124 if (inStream == NULL && status == BAD_VALUE && 5125 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5126 (samplingRate <= 2 * reqSamplingRate) && 5127 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5128 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5129 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5130 &channels, &samplingRate, 5131 (audio_in_acoustics_t)acoustics, 5132 &inStream); 5133 } 5134 5135 if (inStream != NULL) { 5136 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5137 5138 int id = nextUniqueId(); 5139 // Start record thread 5140 // RecorThread require both input and output device indication to forward to audio 5141 // pre processing modules 5142 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5143 thread = new RecordThread(this, 5144 input, 5145 reqSamplingRate, 5146 reqChannels, 5147 id, 5148 device); 5149 mRecordThreads.add(id, thread); 5150 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5151 if (pSamplingRate) *pSamplingRate = reqSamplingRate; 5152 if (pFormat) *pFormat = format; 5153 if (pChannels) *pChannels = reqChannels; 5154 5155 input->stream->common.standby(&input->stream->common); 5156 5157 // notify client processes of the new input creation 5158 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5159 return id; 5160 } 5161 5162 return 0; 5163} 5164 5165status_t AudioFlinger::closeInput(int input) 5166{ 5167 // keep strong reference on the record thread so that 5168 // it is not destroyed while exit() is executed 5169 sp <RecordThread> thread; 5170 { 5171 Mutex::Autolock _l(mLock); 5172 thread = checkRecordThread_l(input); 5173 if (thread == NULL) { 5174 return BAD_VALUE; 5175 } 5176 5177 ALOGV("closeInput() %d", input); 5178 void *param2 = 0; 5179 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, param2); 5180 mRecordThreads.removeItem(input); 5181 } 5182 thread->exit(); 5183 5184 AudioStreamIn *in = thread->clearInput(); 5185 assert(in != NULL); 5186 // from now on thread->mInput is NULL 5187 in->hwDev->close_input_stream(in->hwDev, in->stream); 5188 delete in; 5189 5190 return NO_ERROR; 5191} 5192 5193status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, int output) 5194{ 5195 Mutex::Autolock _l(mLock); 5196 MixerThread *dstThread = checkMixerThread_l(output); 5197 if (dstThread == NULL) { 5198 ALOGW("setStreamOutput() bad output id %d", output); 5199 return BAD_VALUE; 5200 } 5201 5202 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5203 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5204 5205 dstThread->setStreamValid(stream, true); 5206 5207 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5208 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5209 if (thread != dstThread && 5210 thread->type() != ThreadBase::DIRECT) { 5211 MixerThread *srcThread = (MixerThread *)thread; 5212 srcThread->setStreamValid(stream, false); 5213 srcThread->invalidateTracks(stream); 5214 } 5215 } 5216 5217 return NO_ERROR; 5218} 5219 5220 5221int AudioFlinger::newAudioSessionId() 5222{ 5223 return nextUniqueId(); 5224} 5225 5226void AudioFlinger::acquireAudioSessionId(int audioSession) 5227{ 5228 Mutex::Autolock _l(mLock); 5229 int caller = IPCThreadState::self()->getCallingPid(); 5230 ALOGV("acquiring %d from %d", audioSession, caller); 5231 int num = mAudioSessionRefs.size(); 5232 for (int i = 0; i< num; i++) { 5233 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5234 if (ref->sessionid == audioSession && ref->pid == caller) { 5235 ref->cnt++; 5236 ALOGV(" incremented refcount to %d", ref->cnt); 5237 return; 5238 } 5239 } 5240 AudioSessionRef *ref = new AudioSessionRef(); 5241 ref->sessionid = audioSession; 5242 ref->pid = caller; 5243 ref->cnt = 1; 5244 mAudioSessionRefs.push(ref); 5245 ALOGV(" added new entry for %d", ref->sessionid); 5246} 5247 5248void AudioFlinger::releaseAudioSessionId(int audioSession) 5249{ 5250 Mutex::Autolock _l(mLock); 5251 int caller = IPCThreadState::self()->getCallingPid(); 5252 ALOGV("releasing %d from %d", audioSession, caller); 5253 int num = mAudioSessionRefs.size(); 5254 for (int i = 0; i< num; i++) { 5255 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5256 if (ref->sessionid == audioSession && ref->pid == caller) { 5257 ref->cnt--; 5258 ALOGV(" decremented refcount to %d", ref->cnt); 5259 if (ref->cnt == 0) { 5260 mAudioSessionRefs.removeAt(i); 5261 delete ref; 5262 purgeStaleEffects_l(); 5263 } 5264 return; 5265 } 5266 } 5267 ALOGW("session id %d not found for pid %d", audioSession, caller); 5268} 5269 5270void AudioFlinger::purgeStaleEffects_l() { 5271 5272 ALOGV("purging stale effects"); 5273 5274 Vector< sp<EffectChain> > chains; 5275 5276 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5277 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5278 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5279 sp<EffectChain> ec = t->mEffectChains[j]; 5280 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5281 chains.push(ec); 5282 } 5283 } 5284 } 5285 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5286 sp<RecordThread> t = mRecordThreads.valueAt(i); 5287 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5288 sp<EffectChain> ec = t->mEffectChains[j]; 5289 chains.push(ec); 5290 } 5291 } 5292 5293 for (size_t i = 0; i < chains.size(); i++) { 5294 sp<EffectChain> ec = chains[i]; 5295 int sessionid = ec->sessionId(); 5296 sp<ThreadBase> t = ec->mThread.promote(); 5297 if (t == 0) { 5298 continue; 5299 } 5300 size_t numsessionrefs = mAudioSessionRefs.size(); 5301 bool found = false; 5302 for (size_t k = 0; k < numsessionrefs; k++) { 5303 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5304 if (ref->sessionid == sessionid) { 5305 ALOGV(" session %d still exists for %d with %d refs", 5306 sessionid, ref->pid, ref->cnt); 5307 found = true; 5308 break; 5309 } 5310 } 5311 if (!found) { 5312 // remove all effects from the chain 5313 while (ec->mEffects.size()) { 5314 sp<EffectModule> effect = ec->mEffects[0]; 5315 effect->unPin(); 5316 Mutex::Autolock _l (t->mLock); 5317 t->removeEffect_l(effect); 5318 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5319 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5320 if (handle != 0) { 5321 handle->mEffect.clear(); 5322 if (handle->mHasControl && handle->mEnabled) { 5323 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5324 } 5325 } 5326 } 5327 AudioSystem::unregisterEffect(effect->id()); 5328 } 5329 } 5330 } 5331 return; 5332} 5333 5334// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5335AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(int output) const 5336{ 5337 PlaybackThread *thread = NULL; 5338 if (mPlaybackThreads.indexOfKey(output) >= 0) { 5339 thread = (PlaybackThread *)mPlaybackThreads.valueFor(output).get(); 5340 } 5341 return thread; 5342} 5343 5344// checkMixerThread_l() must be called with AudioFlinger::mLock held 5345AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(int output) const 5346{ 5347 PlaybackThread *thread = checkPlaybackThread_l(output); 5348 if (thread != NULL) { 5349 if (thread->type() == ThreadBase::DIRECT) { 5350 thread = NULL; 5351 } 5352 } 5353 return (MixerThread *)thread; 5354} 5355 5356// checkRecordThread_l() must be called with AudioFlinger::mLock held 5357AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(int input) const 5358{ 5359 RecordThread *thread = NULL; 5360 if (mRecordThreads.indexOfKey(input) >= 0) { 5361 thread = (RecordThread *)mRecordThreads.valueFor(input).get(); 5362 } 5363 return thread; 5364} 5365 5366uint32_t AudioFlinger::nextUniqueId() 5367{ 5368 return android_atomic_inc(&mNextUniqueId); 5369} 5370 5371AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5372{ 5373 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5374 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5375 AudioStreamOut *output = thread->getOutput(); 5376 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5377 return thread; 5378 } 5379 } 5380 return NULL; 5381} 5382 5383uint32_t AudioFlinger::primaryOutputDevice_l() 5384{ 5385 PlaybackThread *thread = primaryPlaybackThread_l(); 5386 5387 if (thread == NULL) { 5388 return 0; 5389 } 5390 5391 return thread->device(); 5392} 5393 5394 5395// ---------------------------------------------------------------------------- 5396// Effect management 5397// ---------------------------------------------------------------------------- 5398 5399 5400status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) 5401{ 5402 Mutex::Autolock _l(mLock); 5403 return EffectQueryNumberEffects(numEffects); 5404} 5405 5406status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) 5407{ 5408 Mutex::Autolock _l(mLock); 5409 return EffectQueryEffect(index, descriptor); 5410} 5411 5412status_t AudioFlinger::getEffectDescriptor(effect_uuid_t *pUuid, effect_descriptor_t *descriptor) 5413{ 5414 Mutex::Autolock _l(mLock); 5415 return EffectGetDescriptor(pUuid, descriptor); 5416} 5417 5418 5419sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5420 effect_descriptor_t *pDesc, 5421 const sp<IEffectClient>& effectClient, 5422 int32_t priority, 5423 int io, 5424 int sessionId, 5425 status_t *status, 5426 int *id, 5427 int *enabled) 5428{ 5429 status_t lStatus = NO_ERROR; 5430 sp<EffectHandle> handle; 5431 effect_descriptor_t desc; 5432 sp<Client> client; 5433 wp<Client> wclient; 5434 5435 ALOGV("createEffect pid %d, client %p, priority %d, sessionId %d, io %d", 5436 pid, effectClient.get(), priority, sessionId, io); 5437 5438 if (pDesc == NULL) { 5439 lStatus = BAD_VALUE; 5440 goto Exit; 5441 } 5442 5443 // check audio settings permission for global effects 5444 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5445 lStatus = PERMISSION_DENIED; 5446 goto Exit; 5447 } 5448 5449 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5450 // that can only be created by audio policy manager (running in same process) 5451 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid() != pid) { 5452 lStatus = PERMISSION_DENIED; 5453 goto Exit; 5454 } 5455 5456 if (io == 0) { 5457 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5458 // output must be specified by AudioPolicyManager when using session 5459 // AUDIO_SESSION_OUTPUT_STAGE 5460 lStatus = BAD_VALUE; 5461 goto Exit; 5462 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5463 // if the output returned by getOutputForEffect() is removed before we lock the 5464 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5465 // and we will exit safely 5466 io = AudioSystem::getOutputForEffect(&desc); 5467 } 5468 } 5469 5470 { 5471 Mutex::Autolock _l(mLock); 5472 5473 5474 if (!EffectIsNullUuid(&pDesc->uuid)) { 5475 // if uuid is specified, request effect descriptor 5476 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5477 if (lStatus < 0) { 5478 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5479 goto Exit; 5480 } 5481 } else { 5482 // if uuid is not specified, look for an available implementation 5483 // of the required type in effect factory 5484 if (EffectIsNullUuid(&pDesc->type)) { 5485 ALOGW("createEffect() no effect type"); 5486 lStatus = BAD_VALUE; 5487 goto Exit; 5488 } 5489 uint32_t numEffects = 0; 5490 effect_descriptor_t d; 5491 d.flags = 0; // prevent compiler warning 5492 bool found = false; 5493 5494 lStatus = EffectQueryNumberEffects(&numEffects); 5495 if (lStatus < 0) { 5496 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5497 goto Exit; 5498 } 5499 for (uint32_t i = 0; i < numEffects; i++) { 5500 lStatus = EffectQueryEffect(i, &desc); 5501 if (lStatus < 0) { 5502 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5503 continue; 5504 } 5505 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5506 // If matching type found save effect descriptor. If the session is 5507 // 0 and the effect is not auxiliary, continue enumeration in case 5508 // an auxiliary version of this effect type is available 5509 found = true; 5510 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5511 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5512 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5513 break; 5514 } 5515 } 5516 } 5517 if (!found) { 5518 lStatus = BAD_VALUE; 5519 ALOGW("createEffect() effect not found"); 5520 goto Exit; 5521 } 5522 // For same effect type, chose auxiliary version over insert version if 5523 // connect to output mix (Compliance to OpenSL ES) 5524 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5525 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5526 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5527 } 5528 } 5529 5530 // Do not allow auxiliary effects on a session different from 0 (output mix) 5531 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5532 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5533 lStatus = INVALID_OPERATION; 5534 goto Exit; 5535 } 5536 5537 // check recording permission for visualizer 5538 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5539 !recordingAllowed()) { 5540 lStatus = PERMISSION_DENIED; 5541 goto Exit; 5542 } 5543 5544 // return effect descriptor 5545 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5546 5547 // If output is not specified try to find a matching audio session ID in one of the 5548 // output threads. 5549 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5550 // because of code checking output when entering the function. 5551 // Note: io is never 0 when creating an effect on an input 5552 if (io == 0) { 5553 // look for the thread where the specified audio session is present 5554 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5555 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5556 io = mPlaybackThreads.keyAt(i); 5557 break; 5558 } 5559 } 5560 if (io == 0) { 5561 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5562 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5563 io = mRecordThreads.keyAt(i); 5564 break; 5565 } 5566 } 5567 } 5568 // If no output thread contains the requested session ID, default to 5569 // first output. The effect chain will be moved to the correct output 5570 // thread when a track with the same session ID is created 5571 if (io == 0 && mPlaybackThreads.size()) { 5572 io = mPlaybackThreads.keyAt(0); 5573 } 5574 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5575 } 5576 ThreadBase *thread = checkRecordThread_l(io); 5577 if (thread == NULL) { 5578 thread = checkPlaybackThread_l(io); 5579 if (thread == NULL) { 5580 ALOGE("createEffect() unknown output thread"); 5581 lStatus = BAD_VALUE; 5582 goto Exit; 5583 } 5584 } 5585 5586 wclient = mClients.valueFor(pid); 5587 5588 if (wclient != NULL) { 5589 client = wclient.promote(); 5590 } else { 5591 client = new Client(this, pid); 5592 mClients.add(pid, client); 5593 } 5594 5595 // create effect on selected output thread 5596 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 5597 &desc, enabled, &lStatus); 5598 if (handle != 0 && id != NULL) { 5599 *id = handle->id(); 5600 } 5601 } 5602 5603Exit: 5604 if(status) { 5605 *status = lStatus; 5606 } 5607 return handle; 5608} 5609 5610status_t AudioFlinger::moveEffects(int sessionId, int srcOutput, int dstOutput) 5611{ 5612 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 5613 sessionId, srcOutput, dstOutput); 5614 Mutex::Autolock _l(mLock); 5615 if (srcOutput == dstOutput) { 5616 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 5617 return NO_ERROR; 5618 } 5619 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 5620 if (srcThread == NULL) { 5621 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 5622 return BAD_VALUE; 5623 } 5624 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 5625 if (dstThread == NULL) { 5626 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 5627 return BAD_VALUE; 5628 } 5629 5630 Mutex::Autolock _dl(dstThread->mLock); 5631 Mutex::Autolock _sl(srcThread->mLock); 5632 moveEffectChain_l(sessionId, srcThread, dstThread, false); 5633 5634 return NO_ERROR; 5635} 5636 5637// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 5638status_t AudioFlinger::moveEffectChain_l(int sessionId, 5639 AudioFlinger::PlaybackThread *srcThread, 5640 AudioFlinger::PlaybackThread *dstThread, 5641 bool reRegister) 5642{ 5643 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 5644 sessionId, srcThread, dstThread); 5645 5646 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 5647 if (chain == 0) { 5648 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 5649 sessionId, srcThread); 5650 return INVALID_OPERATION; 5651 } 5652 5653 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 5654 // so that a new chain is created with correct parameters when first effect is added. This is 5655 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 5656 // removed. 5657 srcThread->removeEffectChain_l(chain); 5658 5659 // transfer all effects one by one so that new effect chain is created on new thread with 5660 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 5661 int dstOutput = dstThread->id(); 5662 sp<EffectChain> dstChain; 5663 uint32_t strategy = 0; // prevent compiler warning 5664 sp<EffectModule> effect = chain->getEffectFromId_l(0); 5665 while (effect != 0) { 5666 srcThread->removeEffect_l(effect); 5667 dstThread->addEffect_l(effect); 5668 // removeEffect_l() has stopped the effect if it was active so it must be restarted 5669 if (effect->state() == EffectModule::ACTIVE || 5670 effect->state() == EffectModule::STOPPING) { 5671 effect->start(); 5672 } 5673 // if the move request is not received from audio policy manager, the effect must be 5674 // re-registered with the new strategy and output 5675 if (dstChain == 0) { 5676 dstChain = effect->chain().promote(); 5677 if (dstChain == 0) { 5678 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 5679 srcThread->addEffect_l(effect); 5680 return NO_INIT; 5681 } 5682 strategy = dstChain->strategy(); 5683 } 5684 if (reRegister) { 5685 AudioSystem::unregisterEffect(effect->id()); 5686 AudioSystem::registerEffect(&effect->desc(), 5687 dstOutput, 5688 strategy, 5689 sessionId, 5690 effect->id()); 5691 } 5692 effect = chain->getEffectFromId_l(0); 5693 } 5694 5695 return NO_ERROR; 5696} 5697 5698 5699// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 5700sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 5701 const sp<AudioFlinger::Client>& client, 5702 const sp<IEffectClient>& effectClient, 5703 int32_t priority, 5704 int sessionId, 5705 effect_descriptor_t *desc, 5706 int *enabled, 5707 status_t *status 5708 ) 5709{ 5710 sp<EffectModule> effect; 5711 sp<EffectHandle> handle; 5712 status_t lStatus; 5713 sp<EffectChain> chain; 5714 bool chainCreated = false; 5715 bool effectCreated = false; 5716 bool effectRegistered = false; 5717 5718 lStatus = initCheck(); 5719 if (lStatus != NO_ERROR) { 5720 ALOGW("createEffect_l() Audio driver not initialized."); 5721 goto Exit; 5722 } 5723 5724 // Do not allow effects with session ID 0 on direct output or duplicating threads 5725 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 5726 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 5727 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 5728 desc->name, sessionId); 5729 lStatus = BAD_VALUE; 5730 goto Exit; 5731 } 5732 // Only Pre processor effects are allowed on input threads and only on input threads 5733 if ((mType == RECORD && 5734 (desc->flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_PRE_PROC) || 5735 (mType != RECORD && 5736 (desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 5737 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 5738 desc->name, desc->flags, mType); 5739 lStatus = BAD_VALUE; 5740 goto Exit; 5741 } 5742 5743 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 5744 5745 { // scope for mLock 5746 Mutex::Autolock _l(mLock); 5747 5748 // check for existing effect chain with the requested audio session 5749 chain = getEffectChain_l(sessionId); 5750 if (chain == 0) { 5751 // create a new chain for this session 5752 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 5753 chain = new EffectChain(this, sessionId); 5754 addEffectChain_l(chain); 5755 chain->setStrategy(getStrategyForSession_l(sessionId)); 5756 chainCreated = true; 5757 } else { 5758 effect = chain->getEffectFromDesc_l(desc); 5759 } 5760 5761 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 5762 5763 if (effect == 0) { 5764 int id = mAudioFlinger->nextUniqueId(); 5765 // Check CPU and memory usage 5766 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 5767 if (lStatus != NO_ERROR) { 5768 goto Exit; 5769 } 5770 effectRegistered = true; 5771 // create a new effect module if none present in the chain 5772 effect = new EffectModule(this, chain, desc, id, sessionId); 5773 lStatus = effect->status(); 5774 if (lStatus != NO_ERROR) { 5775 goto Exit; 5776 } 5777 lStatus = chain->addEffect_l(effect); 5778 if (lStatus != NO_ERROR) { 5779 goto Exit; 5780 } 5781 effectCreated = true; 5782 5783 effect->setDevice(mDevice); 5784 effect->setMode(mAudioFlinger->getMode()); 5785 } 5786 // create effect handle and connect it to effect module 5787 handle = new EffectHandle(effect, client, effectClient, priority); 5788 lStatus = effect->addHandle(handle); 5789 if (enabled) { 5790 *enabled = (int)effect->isEnabled(); 5791 } 5792 } 5793 5794Exit: 5795 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 5796 Mutex::Autolock _l(mLock); 5797 if (effectCreated) { 5798 chain->removeEffect_l(effect); 5799 } 5800 if (effectRegistered) { 5801 AudioSystem::unregisterEffect(effect->id()); 5802 } 5803 if (chainCreated) { 5804 removeEffectChain_l(chain); 5805 } 5806 handle.clear(); 5807 } 5808 5809 if(status) { 5810 *status = lStatus; 5811 } 5812 return handle; 5813} 5814 5815sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 5816{ 5817 sp<EffectModule> effect; 5818 5819 sp<EffectChain> chain = getEffectChain_l(sessionId); 5820 if (chain != 0) { 5821 effect = chain->getEffectFromId_l(effectId); 5822 } 5823 return effect; 5824} 5825 5826// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 5827// PlaybackThread::mLock held 5828status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 5829{ 5830 // check for existing effect chain with the requested audio session 5831 int sessionId = effect->sessionId(); 5832 sp<EffectChain> chain = getEffectChain_l(sessionId); 5833 bool chainCreated = false; 5834 5835 if (chain == 0) { 5836 // create a new chain for this session 5837 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 5838 chain = new EffectChain(this, sessionId); 5839 addEffectChain_l(chain); 5840 chain->setStrategy(getStrategyForSession_l(sessionId)); 5841 chainCreated = true; 5842 } 5843 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 5844 5845 if (chain->getEffectFromId_l(effect->id()) != 0) { 5846 ALOGW("addEffect_l() %p effect %s already present in chain %p", 5847 this, effect->desc().name, chain.get()); 5848 return BAD_VALUE; 5849 } 5850 5851 status_t status = chain->addEffect_l(effect); 5852 if (status != NO_ERROR) { 5853 if (chainCreated) { 5854 removeEffectChain_l(chain); 5855 } 5856 return status; 5857 } 5858 5859 effect->setDevice(mDevice); 5860 effect->setMode(mAudioFlinger->getMode()); 5861 return NO_ERROR; 5862} 5863 5864void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 5865 5866 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 5867 effect_descriptor_t desc = effect->desc(); 5868 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5869 detachAuxEffect_l(effect->id()); 5870 } 5871 5872 sp<EffectChain> chain = effect->chain().promote(); 5873 if (chain != 0) { 5874 // remove effect chain if removing last effect 5875 if (chain->removeEffect_l(effect) == 0) { 5876 removeEffectChain_l(chain); 5877 } 5878 } else { 5879 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 5880 } 5881} 5882 5883void AudioFlinger::ThreadBase::lockEffectChains_l( 5884 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5885{ 5886 effectChains = mEffectChains; 5887 for (size_t i = 0; i < mEffectChains.size(); i++) { 5888 mEffectChains[i]->lock(); 5889 } 5890} 5891 5892void AudioFlinger::ThreadBase::unlockEffectChains( 5893 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 5894{ 5895 for (size_t i = 0; i < effectChains.size(); i++) { 5896 effectChains[i]->unlock(); 5897 } 5898} 5899 5900sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 5901{ 5902 Mutex::Autolock _l(mLock); 5903 return getEffectChain_l(sessionId); 5904} 5905 5906sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 5907{ 5908 sp<EffectChain> chain; 5909 5910 size_t size = mEffectChains.size(); 5911 for (size_t i = 0; i < size; i++) { 5912 if (mEffectChains[i]->sessionId() == sessionId) { 5913 chain = mEffectChains[i]; 5914 break; 5915 } 5916 } 5917 return chain; 5918} 5919 5920void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 5921{ 5922 Mutex::Autolock _l(mLock); 5923 size_t size = mEffectChains.size(); 5924 for (size_t i = 0; i < size; i++) { 5925 mEffectChains[i]->setMode_l(mode); 5926 } 5927} 5928 5929void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 5930 const wp<EffectHandle>& handle, 5931 bool unpiniflast) { 5932 5933 Mutex::Autolock _l(mLock); 5934 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 5935 // delete the effect module if removing last handle on it 5936 if (effect->removeHandle(handle) == 0) { 5937 if (!effect->isPinned() || unpiniflast) { 5938 removeEffect_l(effect); 5939 AudioSystem::unregisterEffect(effect->id()); 5940 } 5941 } 5942} 5943 5944status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 5945{ 5946 int session = chain->sessionId(); 5947 int16_t *buffer = mMixBuffer; 5948 bool ownsBuffer = false; 5949 5950 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 5951 if (session > 0) { 5952 // Only one effect chain can be present in direct output thread and it uses 5953 // the mix buffer as input 5954 if (mType != DIRECT) { 5955 size_t numSamples = mFrameCount * mChannelCount; 5956 buffer = new int16_t[numSamples]; 5957 memset(buffer, 0, numSamples * sizeof(int16_t)); 5958 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 5959 ownsBuffer = true; 5960 } 5961 5962 // Attach all tracks with same session ID to this chain. 5963 for (size_t i = 0; i < mTracks.size(); ++i) { 5964 sp<Track> track = mTracks[i]; 5965 if (session == track->sessionId()) { 5966 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 5967 track->setMainBuffer(buffer); 5968 chain->incTrackCnt(); 5969 } 5970 } 5971 5972 // indicate all active tracks in the chain 5973 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 5974 sp<Track> track = mActiveTracks[i].promote(); 5975 if (track == 0) continue; 5976 if (session == track->sessionId()) { 5977 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 5978 chain->incActiveTrackCnt(); 5979 } 5980 } 5981 } 5982 5983 chain->setInBuffer(buffer, ownsBuffer); 5984 chain->setOutBuffer(mMixBuffer); 5985 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 5986 // chains list in order to be processed last as it contains output stage effects 5987 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 5988 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 5989 // after track specific effects and before output stage 5990 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 5991 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 5992 // Effect chain for other sessions are inserted at beginning of effect 5993 // chains list to be processed before output mix effects. Relative order between other 5994 // sessions is not important 5995 size_t size = mEffectChains.size(); 5996 size_t i = 0; 5997 for (i = 0; i < size; i++) { 5998 if (mEffectChains[i]->sessionId() < session) break; 5999 } 6000 mEffectChains.insertAt(chain, i); 6001 checkSuspendOnAddEffectChain_l(chain); 6002 6003 return NO_ERROR; 6004} 6005 6006size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6007{ 6008 int session = chain->sessionId(); 6009 6010 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6011 6012 for (size_t i = 0; i < mEffectChains.size(); i++) { 6013 if (chain == mEffectChains[i]) { 6014 mEffectChains.removeAt(i); 6015 // detach all active tracks from the chain 6016 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6017 sp<Track> track = mActiveTracks[i].promote(); 6018 if (track == 0) continue; 6019 if (session == track->sessionId()) { 6020 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6021 chain.get(), session); 6022 chain->decActiveTrackCnt(); 6023 } 6024 } 6025 6026 // detach all tracks with same session ID from this chain 6027 for (size_t i = 0; i < mTracks.size(); ++i) { 6028 sp<Track> track = mTracks[i]; 6029 if (session == track->sessionId()) { 6030 track->setMainBuffer(mMixBuffer); 6031 chain->decTrackCnt(); 6032 } 6033 } 6034 break; 6035 } 6036 } 6037 return mEffectChains.size(); 6038} 6039 6040status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6041 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6042{ 6043 Mutex::Autolock _l(mLock); 6044 return attachAuxEffect_l(track, EffectId); 6045} 6046 6047status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6048 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6049{ 6050 status_t status = NO_ERROR; 6051 6052 if (EffectId == 0) { 6053 track->setAuxBuffer(0, NULL); 6054 } else { 6055 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6056 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6057 if (effect != 0) { 6058 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6059 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6060 } else { 6061 status = INVALID_OPERATION; 6062 } 6063 } else { 6064 status = BAD_VALUE; 6065 } 6066 } 6067 return status; 6068} 6069 6070void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6071{ 6072 for (size_t i = 0; i < mTracks.size(); ++i) { 6073 sp<Track> track = mTracks[i]; 6074 if (track->auxEffectId() == effectId) { 6075 attachAuxEffect_l(track, 0); 6076 } 6077 } 6078} 6079 6080status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6081{ 6082 // only one chain per input thread 6083 if (mEffectChains.size() != 0) { 6084 return INVALID_OPERATION; 6085 } 6086 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6087 6088 chain->setInBuffer(NULL); 6089 chain->setOutBuffer(NULL); 6090 6091 checkSuspendOnAddEffectChain_l(chain); 6092 6093 mEffectChains.add(chain); 6094 6095 return NO_ERROR; 6096} 6097 6098size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6099{ 6100 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6101 ALOGW_IF(mEffectChains.size() != 1, 6102 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6103 chain.get(), mEffectChains.size(), this); 6104 if (mEffectChains.size() == 1) { 6105 mEffectChains.removeAt(0); 6106 } 6107 return 0; 6108} 6109 6110// ---------------------------------------------------------------------------- 6111// EffectModule implementation 6112// ---------------------------------------------------------------------------- 6113 6114#undef LOG_TAG 6115#define LOG_TAG "AudioFlinger::EffectModule" 6116 6117AudioFlinger::EffectModule::EffectModule(const wp<ThreadBase>& wThread, 6118 const wp<AudioFlinger::EffectChain>& chain, 6119 effect_descriptor_t *desc, 6120 int id, 6121 int sessionId) 6122 : mThread(wThread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6123 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6124{ 6125 ALOGV("Constructor %p", this); 6126 int lStatus; 6127 sp<ThreadBase> thread = mThread.promote(); 6128 if (thread == 0) { 6129 return; 6130 } 6131 6132 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6133 6134 // create effect engine from effect factory 6135 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6136 6137 if (mStatus != NO_ERROR) { 6138 return; 6139 } 6140 lStatus = init(); 6141 if (lStatus < 0) { 6142 mStatus = lStatus; 6143 goto Error; 6144 } 6145 6146 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6147 mPinned = true; 6148 } 6149 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6150 return; 6151Error: 6152 EffectRelease(mEffectInterface); 6153 mEffectInterface = NULL; 6154 ALOGV("Constructor Error %d", mStatus); 6155} 6156 6157AudioFlinger::EffectModule::~EffectModule() 6158{ 6159 ALOGV("Destructor %p", this); 6160 if (mEffectInterface != NULL) { 6161 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6162 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6163 sp<ThreadBase> thread = mThread.promote(); 6164 if (thread != 0) { 6165 audio_stream_t *stream = thread->stream(); 6166 if (stream != NULL) { 6167 stream->remove_audio_effect(stream, mEffectInterface); 6168 } 6169 } 6170 } 6171 // release effect engine 6172 EffectRelease(mEffectInterface); 6173 } 6174} 6175 6176status_t AudioFlinger::EffectModule::addHandle(sp<EffectHandle>& handle) 6177{ 6178 status_t status; 6179 6180 Mutex::Autolock _l(mLock); 6181 // First handle in mHandles has highest priority and controls the effect module 6182 int priority = handle->priority(); 6183 size_t size = mHandles.size(); 6184 sp<EffectHandle> h; 6185 size_t i; 6186 for (i = 0; i < size; i++) { 6187 h = mHandles[i].promote(); 6188 if (h == 0) continue; 6189 if (h->priority() <= priority) break; 6190 } 6191 // if inserted in first place, move effect control from previous owner to this handle 6192 if (i == 0) { 6193 bool enabled = false; 6194 if (h != 0) { 6195 enabled = h->enabled(); 6196 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6197 } 6198 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6199 status = NO_ERROR; 6200 } else { 6201 status = ALREADY_EXISTS; 6202 } 6203 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6204 mHandles.insertAt(handle, i); 6205 return status; 6206} 6207 6208size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6209{ 6210 Mutex::Autolock _l(mLock); 6211 size_t size = mHandles.size(); 6212 size_t i; 6213 for (i = 0; i < size; i++) { 6214 if (mHandles[i] == handle) break; 6215 } 6216 if (i == size) { 6217 return size; 6218 } 6219 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6220 6221 bool enabled = false; 6222 EffectHandle *hdl = handle.unsafe_get(); 6223 if (hdl) { 6224 ALOGV("removeHandle() unsafe_get OK"); 6225 enabled = hdl->enabled(); 6226 } 6227 mHandles.removeAt(i); 6228 size = mHandles.size(); 6229 // if removed from first place, move effect control from this handle to next in line 6230 if (i == 0 && size != 0) { 6231 sp<EffectHandle> h = mHandles[0].promote(); 6232 if (h != 0) { 6233 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6234 } 6235 } 6236 6237 // Prevent calls to process() and other functions on effect interface from now on. 6238 // The effect engine will be released by the destructor when the last strong reference on 6239 // this object is released which can happen after next process is called. 6240 if (size == 0 && !mPinned) { 6241 mState = DESTROYED; 6242 } 6243 6244 return size; 6245} 6246 6247sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6248{ 6249 Mutex::Autolock _l(mLock); 6250 sp<EffectHandle> handle; 6251 if (mHandles.size() != 0) { 6252 handle = mHandles[0].promote(); 6253 } 6254 return handle; 6255} 6256 6257void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpiniflast) 6258{ 6259 ALOGV("disconnect() %p handle %p ", this, handle.unsafe_get()); 6260 // keep a strong reference on this EffectModule to avoid calling the 6261 // destructor before we exit 6262 sp<EffectModule> keep(this); 6263 { 6264 sp<ThreadBase> thread = mThread.promote(); 6265 if (thread != 0) { 6266 thread->disconnectEffect(keep, handle, unpiniflast); 6267 } 6268 } 6269} 6270 6271void AudioFlinger::EffectModule::updateState() { 6272 Mutex::Autolock _l(mLock); 6273 6274 switch (mState) { 6275 case RESTART: 6276 reset_l(); 6277 // FALL THROUGH 6278 6279 case STARTING: 6280 // clear auxiliary effect input buffer for next accumulation 6281 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6282 memset(mConfig.inputCfg.buffer.raw, 6283 0, 6284 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6285 } 6286 start_l(); 6287 mState = ACTIVE; 6288 break; 6289 case STOPPING: 6290 stop_l(); 6291 mDisableWaitCnt = mMaxDisableWaitCnt; 6292 mState = STOPPED; 6293 break; 6294 case STOPPED: 6295 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6296 // turn off sequence. 6297 if (--mDisableWaitCnt == 0) { 6298 reset_l(); 6299 mState = IDLE; 6300 } 6301 break; 6302 default: //IDLE , ACTIVE, DESTROYED 6303 break; 6304 } 6305} 6306 6307void AudioFlinger::EffectModule::process() 6308{ 6309 Mutex::Autolock _l(mLock); 6310 6311 if (mState == DESTROYED || mEffectInterface == NULL || 6312 mConfig.inputCfg.buffer.raw == NULL || 6313 mConfig.outputCfg.buffer.raw == NULL) { 6314 return; 6315 } 6316 6317 if (isProcessEnabled()) { 6318 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6319 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6320 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6321 mConfig.inputCfg.buffer.s32, 6322 mConfig.inputCfg.buffer.frameCount/2); 6323 } 6324 6325 // do the actual processing in the effect engine 6326 int ret = (*mEffectInterface)->process(mEffectInterface, 6327 &mConfig.inputCfg.buffer, 6328 &mConfig.outputCfg.buffer); 6329 6330 // force transition to IDLE state when engine is ready 6331 if (mState == STOPPED && ret == -ENODATA) { 6332 mDisableWaitCnt = 1; 6333 } 6334 6335 // clear auxiliary effect input buffer for next accumulation 6336 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6337 memset(mConfig.inputCfg.buffer.raw, 0, 6338 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6339 } 6340 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6341 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6342 // If an insert effect is idle and input buffer is different from output buffer, 6343 // accumulate input onto output 6344 sp<EffectChain> chain = mChain.promote(); 6345 if (chain != 0 && chain->activeTrackCnt() != 0) { 6346 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6347 int16_t *in = mConfig.inputCfg.buffer.s16; 6348 int16_t *out = mConfig.outputCfg.buffer.s16; 6349 for (size_t i = 0; i < frameCnt; i++) { 6350 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6351 } 6352 } 6353 } 6354} 6355 6356void AudioFlinger::EffectModule::reset_l() 6357{ 6358 if (mEffectInterface == NULL) { 6359 return; 6360 } 6361 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6362} 6363 6364status_t AudioFlinger::EffectModule::configure() 6365{ 6366 uint32_t channels; 6367 if (mEffectInterface == NULL) { 6368 return NO_INIT; 6369 } 6370 6371 sp<ThreadBase> thread = mThread.promote(); 6372 if (thread == 0) { 6373 return DEAD_OBJECT; 6374 } 6375 6376 // TODO: handle configuration of effects replacing track process 6377 if (thread->channelCount() == 1) { 6378 channels = AUDIO_CHANNEL_OUT_MONO; 6379 } else { 6380 channels = AUDIO_CHANNEL_OUT_STEREO; 6381 } 6382 6383 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6384 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6385 } else { 6386 mConfig.inputCfg.channels = channels; 6387 } 6388 mConfig.outputCfg.channels = channels; 6389 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6390 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6391 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6392 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6393 mConfig.inputCfg.bufferProvider.cookie = NULL; 6394 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6395 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6396 mConfig.outputCfg.bufferProvider.cookie = NULL; 6397 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6398 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6399 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6400 // Insert effect: 6401 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6402 // always overwrites output buffer: input buffer == output buffer 6403 // - in other sessions: 6404 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6405 // other effect: overwrites output buffer: input buffer == output buffer 6406 // Auxiliary effect: 6407 // accumulates in output buffer: input buffer != output buffer 6408 // Therefore: accumulate <=> input buffer != output buffer 6409 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6410 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6411 } else { 6412 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6413 } 6414 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6415 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6416 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6417 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6418 6419 ALOGV("configure() %p thread %p buffer %p framecount %d", 6420 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6421 6422 status_t cmdStatus; 6423 uint32_t size = sizeof(int); 6424 status_t status = (*mEffectInterface)->command(mEffectInterface, 6425 EFFECT_CMD_SET_CONFIG, 6426 sizeof(effect_config_t), 6427 &mConfig, 6428 &size, 6429 &cmdStatus); 6430 if (status == 0) { 6431 status = cmdStatus; 6432 } 6433 6434 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6435 (1000 * mConfig.outputCfg.buffer.frameCount); 6436 6437 return status; 6438} 6439 6440status_t AudioFlinger::EffectModule::init() 6441{ 6442 Mutex::Autolock _l(mLock); 6443 if (mEffectInterface == NULL) { 6444 return NO_INIT; 6445 } 6446 status_t cmdStatus; 6447 uint32_t size = sizeof(status_t); 6448 status_t status = (*mEffectInterface)->command(mEffectInterface, 6449 EFFECT_CMD_INIT, 6450 0, 6451 NULL, 6452 &size, 6453 &cmdStatus); 6454 if (status == 0) { 6455 status = cmdStatus; 6456 } 6457 return status; 6458} 6459 6460status_t AudioFlinger::EffectModule::start() 6461{ 6462 Mutex::Autolock _l(mLock); 6463 return start_l(); 6464} 6465 6466status_t AudioFlinger::EffectModule::start_l() 6467{ 6468 if (mEffectInterface == NULL) { 6469 return NO_INIT; 6470 } 6471 status_t cmdStatus; 6472 uint32_t size = sizeof(status_t); 6473 status_t status = (*mEffectInterface)->command(mEffectInterface, 6474 EFFECT_CMD_ENABLE, 6475 0, 6476 NULL, 6477 &size, 6478 &cmdStatus); 6479 if (status == 0) { 6480 status = cmdStatus; 6481 } 6482 if (status == 0 && 6483 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6484 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6485 sp<ThreadBase> thread = mThread.promote(); 6486 if (thread != 0) { 6487 audio_stream_t *stream = thread->stream(); 6488 if (stream != NULL) { 6489 stream->add_audio_effect(stream, mEffectInterface); 6490 } 6491 } 6492 } 6493 return status; 6494} 6495 6496status_t AudioFlinger::EffectModule::stop() 6497{ 6498 Mutex::Autolock _l(mLock); 6499 return stop_l(); 6500} 6501 6502status_t AudioFlinger::EffectModule::stop_l() 6503{ 6504 if (mEffectInterface == NULL) { 6505 return NO_INIT; 6506 } 6507 status_t cmdStatus; 6508 uint32_t size = sizeof(status_t); 6509 status_t status = (*mEffectInterface)->command(mEffectInterface, 6510 EFFECT_CMD_DISABLE, 6511 0, 6512 NULL, 6513 &size, 6514 &cmdStatus); 6515 if (status == 0) { 6516 status = cmdStatus; 6517 } 6518 if (status == 0 && 6519 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6520 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6521 sp<ThreadBase> thread = mThread.promote(); 6522 if (thread != 0) { 6523 audio_stream_t *stream = thread->stream(); 6524 if (stream != NULL) { 6525 stream->remove_audio_effect(stream, mEffectInterface); 6526 } 6527 } 6528 } 6529 return status; 6530} 6531 6532status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6533 uint32_t cmdSize, 6534 void *pCmdData, 6535 uint32_t *replySize, 6536 void *pReplyData) 6537{ 6538 Mutex::Autolock _l(mLock); 6539// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6540 6541 if (mState == DESTROYED || mEffectInterface == NULL) { 6542 return NO_INIT; 6543 } 6544 status_t status = (*mEffectInterface)->command(mEffectInterface, 6545 cmdCode, 6546 cmdSize, 6547 pCmdData, 6548 replySize, 6549 pReplyData); 6550 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6551 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6552 for (size_t i = 1; i < mHandles.size(); i++) { 6553 sp<EffectHandle> h = mHandles[i].promote(); 6554 if (h != 0) { 6555 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6556 } 6557 } 6558 } 6559 return status; 6560} 6561 6562status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6563{ 6564 6565 Mutex::Autolock _l(mLock); 6566 ALOGV("setEnabled %p enabled %d", this, enabled); 6567 6568 if (enabled != isEnabled()) { 6569 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6570 if (enabled && status != NO_ERROR) { 6571 return status; 6572 } 6573 6574 switch (mState) { 6575 // going from disabled to enabled 6576 case IDLE: 6577 mState = STARTING; 6578 break; 6579 case STOPPED: 6580 mState = RESTART; 6581 break; 6582 case STOPPING: 6583 mState = ACTIVE; 6584 break; 6585 6586 // going from enabled to disabled 6587 case RESTART: 6588 mState = STOPPED; 6589 break; 6590 case STARTING: 6591 mState = IDLE; 6592 break; 6593 case ACTIVE: 6594 mState = STOPPING; 6595 break; 6596 case DESTROYED: 6597 return NO_ERROR; // simply ignore as we are being destroyed 6598 } 6599 for (size_t i = 1; i < mHandles.size(); i++) { 6600 sp<EffectHandle> h = mHandles[i].promote(); 6601 if (h != 0) { 6602 h->setEnabled(enabled); 6603 } 6604 } 6605 } 6606 return NO_ERROR; 6607} 6608 6609bool AudioFlinger::EffectModule::isEnabled() 6610{ 6611 switch (mState) { 6612 case RESTART: 6613 case STARTING: 6614 case ACTIVE: 6615 return true; 6616 case IDLE: 6617 case STOPPING: 6618 case STOPPED: 6619 case DESTROYED: 6620 default: 6621 return false; 6622 } 6623} 6624 6625bool AudioFlinger::EffectModule::isProcessEnabled() 6626{ 6627 switch (mState) { 6628 case RESTART: 6629 case ACTIVE: 6630 case STOPPING: 6631 case STOPPED: 6632 return true; 6633 case IDLE: 6634 case STARTING: 6635 case DESTROYED: 6636 default: 6637 return false; 6638 } 6639} 6640 6641status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 6642{ 6643 Mutex::Autolock _l(mLock); 6644 status_t status = NO_ERROR; 6645 6646 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 6647 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 6648 if (isProcessEnabled() && 6649 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 6650 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 6651 status_t cmdStatus; 6652 uint32_t volume[2]; 6653 uint32_t *pVolume = NULL; 6654 uint32_t size = sizeof(volume); 6655 volume[0] = *left; 6656 volume[1] = *right; 6657 if (controller) { 6658 pVolume = volume; 6659 } 6660 status = (*mEffectInterface)->command(mEffectInterface, 6661 EFFECT_CMD_SET_VOLUME, 6662 size, 6663 volume, 6664 &size, 6665 pVolume); 6666 if (controller && status == NO_ERROR && size == sizeof(volume)) { 6667 *left = volume[0]; 6668 *right = volume[1]; 6669 } 6670 } 6671 return status; 6672} 6673 6674status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 6675{ 6676 Mutex::Autolock _l(mLock); 6677 status_t status = NO_ERROR; 6678 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 6679 // audio pre processing modules on RecordThread can receive both output and 6680 // input device indication in the same call 6681 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 6682 if (dev) { 6683 status_t cmdStatus; 6684 uint32_t size = sizeof(status_t); 6685 6686 status = (*mEffectInterface)->command(mEffectInterface, 6687 EFFECT_CMD_SET_DEVICE, 6688 sizeof(uint32_t), 6689 &dev, 6690 &size, 6691 &cmdStatus); 6692 if (status == NO_ERROR) { 6693 status = cmdStatus; 6694 } 6695 } 6696 dev = device & AUDIO_DEVICE_IN_ALL; 6697 if (dev) { 6698 status_t cmdStatus; 6699 uint32_t size = sizeof(status_t); 6700 6701 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 6702 EFFECT_CMD_SET_INPUT_DEVICE, 6703 sizeof(uint32_t), 6704 &dev, 6705 &size, 6706 &cmdStatus); 6707 if (status2 == NO_ERROR) { 6708 status2 = cmdStatus; 6709 } 6710 if (status == NO_ERROR) { 6711 status = status2; 6712 } 6713 } 6714 } 6715 return status; 6716} 6717 6718status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 6719{ 6720 Mutex::Autolock _l(mLock); 6721 status_t status = NO_ERROR; 6722 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 6723 status_t cmdStatus; 6724 uint32_t size = sizeof(status_t); 6725 status = (*mEffectInterface)->command(mEffectInterface, 6726 EFFECT_CMD_SET_AUDIO_MODE, 6727 sizeof(audio_mode_t), 6728 &mode, 6729 &size, 6730 &cmdStatus); 6731 if (status == NO_ERROR) { 6732 status = cmdStatus; 6733 } 6734 } 6735 return status; 6736} 6737 6738void AudioFlinger::EffectModule::setSuspended(bool suspended) 6739{ 6740 Mutex::Autolock _l(mLock); 6741 mSuspended = suspended; 6742} 6743 6744bool AudioFlinger::EffectModule::suspended() const 6745{ 6746 Mutex::Autolock _l(mLock); 6747 return mSuspended; 6748} 6749 6750status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 6751{ 6752 const size_t SIZE = 256; 6753 char buffer[SIZE]; 6754 String8 result; 6755 6756 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 6757 result.append(buffer); 6758 6759 bool locked = tryLock(mLock); 6760 // failed to lock - AudioFlinger is probably deadlocked 6761 if (!locked) { 6762 result.append("\t\tCould not lock Fx mutex:\n"); 6763 } 6764 6765 result.append("\t\tSession Status State Engine:\n"); 6766 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 6767 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 6768 result.append(buffer); 6769 6770 result.append("\t\tDescriptor:\n"); 6771 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6772 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 6773 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 6774 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 6775 result.append(buffer); 6776 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 6777 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 6778 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 6779 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 6780 result.append(buffer); 6781 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 6782 mDescriptor.apiVersion, 6783 mDescriptor.flags); 6784 result.append(buffer); 6785 snprintf(buffer, SIZE, "\t\t- name: %s\n", 6786 mDescriptor.name); 6787 result.append(buffer); 6788 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 6789 mDescriptor.implementor); 6790 result.append(buffer); 6791 6792 result.append("\t\t- Input configuration:\n"); 6793 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6794 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6795 (uint32_t)mConfig.inputCfg.buffer.raw, 6796 mConfig.inputCfg.buffer.frameCount, 6797 mConfig.inputCfg.samplingRate, 6798 mConfig.inputCfg.channels, 6799 mConfig.inputCfg.format); 6800 result.append(buffer); 6801 6802 result.append("\t\t- Output configuration:\n"); 6803 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 6804 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 6805 (uint32_t)mConfig.outputCfg.buffer.raw, 6806 mConfig.outputCfg.buffer.frameCount, 6807 mConfig.outputCfg.samplingRate, 6808 mConfig.outputCfg.channels, 6809 mConfig.outputCfg.format); 6810 result.append(buffer); 6811 6812 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 6813 result.append(buffer); 6814 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 6815 for (size_t i = 0; i < mHandles.size(); ++i) { 6816 sp<EffectHandle> handle = mHandles[i].promote(); 6817 if (handle != 0) { 6818 handle->dump(buffer, SIZE); 6819 result.append(buffer); 6820 } 6821 } 6822 6823 result.append("\n"); 6824 6825 write(fd, result.string(), result.length()); 6826 6827 if (locked) { 6828 mLock.unlock(); 6829 } 6830 6831 return NO_ERROR; 6832} 6833 6834// ---------------------------------------------------------------------------- 6835// EffectHandle implementation 6836// ---------------------------------------------------------------------------- 6837 6838#undef LOG_TAG 6839#define LOG_TAG "AudioFlinger::EffectHandle" 6840 6841AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 6842 const sp<AudioFlinger::Client>& client, 6843 const sp<IEffectClient>& effectClient, 6844 int32_t priority) 6845 : BnEffect(), 6846 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 6847 mPriority(priority), mHasControl(false), mEnabled(false) 6848{ 6849 ALOGV("constructor %p", this); 6850 6851 if (client == 0) { 6852 return; 6853 } 6854 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 6855 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 6856 if (mCblkMemory != 0) { 6857 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 6858 6859 if (mCblk) { 6860 new(mCblk) effect_param_cblk_t(); 6861 mBuffer = (uint8_t *)mCblk + bufOffset; 6862 } 6863 } else { 6864 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 6865 return; 6866 } 6867} 6868 6869AudioFlinger::EffectHandle::~EffectHandle() 6870{ 6871 ALOGV("Destructor %p", this); 6872 disconnect(false); 6873 ALOGV("Destructor DONE %p", this); 6874} 6875 6876status_t AudioFlinger::EffectHandle::enable() 6877{ 6878 ALOGV("enable %p", this); 6879 if (!mHasControl) return INVALID_OPERATION; 6880 if (mEffect == 0) return DEAD_OBJECT; 6881 6882 if (mEnabled) { 6883 return NO_ERROR; 6884 } 6885 6886 mEnabled = true; 6887 6888 sp<ThreadBase> thread = mEffect->thread().promote(); 6889 if (thread != 0) { 6890 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 6891 } 6892 6893 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 6894 if (mEffect->suspended()) { 6895 return NO_ERROR; 6896 } 6897 6898 status_t status = mEffect->setEnabled(true); 6899 if (status != NO_ERROR) { 6900 if (thread != 0) { 6901 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6902 } 6903 mEnabled = false; 6904 } 6905 return status; 6906} 6907 6908status_t AudioFlinger::EffectHandle::disable() 6909{ 6910 ALOGV("disable %p", this); 6911 if (!mHasControl) return INVALID_OPERATION; 6912 if (mEffect == 0) return DEAD_OBJECT; 6913 6914 if (!mEnabled) { 6915 return NO_ERROR; 6916 } 6917 mEnabled = false; 6918 6919 if (mEffect->suspended()) { 6920 return NO_ERROR; 6921 } 6922 6923 status_t status = mEffect->setEnabled(false); 6924 6925 sp<ThreadBase> thread = mEffect->thread().promote(); 6926 if (thread != 0) { 6927 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6928 } 6929 6930 return status; 6931} 6932 6933void AudioFlinger::EffectHandle::disconnect() 6934{ 6935 disconnect(true); 6936} 6937 6938void AudioFlinger::EffectHandle::disconnect(bool unpiniflast) 6939{ 6940 ALOGV("disconnect(%s)", unpiniflast ? "true" : "false"); 6941 if (mEffect == 0) { 6942 return; 6943 } 6944 mEffect->disconnect(this, unpiniflast); 6945 6946 if (mHasControl && mEnabled) { 6947 sp<ThreadBase> thread = mEffect->thread().promote(); 6948 if (thread != 0) { 6949 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 6950 } 6951 } 6952 6953 // release sp on module => module destructor can be called now 6954 mEffect.clear(); 6955 if (mClient != 0) { 6956 if (mCblk) { 6957 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 6958 } 6959 mCblkMemory.clear(); // and free the shared memory 6960 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 6961 mClient.clear(); 6962 } 6963} 6964 6965status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 6966 uint32_t cmdSize, 6967 void *pCmdData, 6968 uint32_t *replySize, 6969 void *pReplyData) 6970{ 6971// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 6972// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 6973 6974 // only get parameter command is permitted for applications not controlling the effect 6975 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 6976 return INVALID_OPERATION; 6977 } 6978 if (mEffect == 0) return DEAD_OBJECT; 6979 if (mClient == 0) return INVALID_OPERATION; 6980 6981 // handle commands that are not forwarded transparently to effect engine 6982 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 6983 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 6984 // no risk to block the whole media server process or mixer threads is we are stuck here 6985 Mutex::Autolock _l(mCblk->lock); 6986 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 6987 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 6988 mCblk->serverIndex = 0; 6989 mCblk->clientIndex = 0; 6990 return BAD_VALUE; 6991 } 6992 status_t status = NO_ERROR; 6993 while (mCblk->serverIndex < mCblk->clientIndex) { 6994 int reply; 6995 uint32_t rsize = sizeof(int); 6996 int *p = (int *)(mBuffer + mCblk->serverIndex); 6997 int size = *p++; 6998 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 6999 ALOGW("command(): invalid parameter block size"); 7000 break; 7001 } 7002 effect_param_t *param = (effect_param_t *)p; 7003 if (param->psize == 0 || param->vsize == 0) { 7004 ALOGW("command(): null parameter or value size"); 7005 mCblk->serverIndex += size; 7006 continue; 7007 } 7008 uint32_t psize = sizeof(effect_param_t) + 7009 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7010 param->vsize; 7011 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7012 psize, 7013 p, 7014 &rsize, 7015 &reply); 7016 // stop at first error encountered 7017 if (ret != NO_ERROR) { 7018 status = ret; 7019 *(int *)pReplyData = reply; 7020 break; 7021 } else if (reply != NO_ERROR) { 7022 *(int *)pReplyData = reply; 7023 break; 7024 } 7025 mCblk->serverIndex += size; 7026 } 7027 mCblk->serverIndex = 0; 7028 mCblk->clientIndex = 0; 7029 return status; 7030 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7031 *(int *)pReplyData = NO_ERROR; 7032 return enable(); 7033 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7034 *(int *)pReplyData = NO_ERROR; 7035 return disable(); 7036 } 7037 7038 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7039} 7040 7041sp<IMemory> AudioFlinger::EffectHandle::getCblk() const { 7042 return mCblkMemory; 7043} 7044 7045void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7046{ 7047 ALOGV("setControl %p control %d", this, hasControl); 7048 7049 mHasControl = hasControl; 7050 mEnabled = enabled; 7051 7052 if (signal && mEffectClient != 0) { 7053 mEffectClient->controlStatusChanged(hasControl); 7054 } 7055} 7056 7057void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7058 uint32_t cmdSize, 7059 void *pCmdData, 7060 uint32_t replySize, 7061 void *pReplyData) 7062{ 7063 if (mEffectClient != 0) { 7064 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7065 } 7066} 7067 7068 7069 7070void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7071{ 7072 if (mEffectClient != 0) { 7073 mEffectClient->enableStatusChanged(enabled); 7074 } 7075} 7076 7077status_t AudioFlinger::EffectHandle::onTransact( 7078 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7079{ 7080 return BnEffect::onTransact(code, data, reply, flags); 7081} 7082 7083 7084void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7085{ 7086 bool locked = mCblk ? tryLock(mCblk->lock) : false; 7087 7088 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7089 (mClient == NULL) ? getpid() : mClient->pid(), 7090 mPriority, 7091 mHasControl, 7092 !locked, 7093 mCblk ? mCblk->clientIndex : 0, 7094 mCblk ? mCblk->serverIndex : 0 7095 ); 7096 7097 if (locked) { 7098 mCblk->lock.unlock(); 7099 } 7100} 7101 7102#undef LOG_TAG 7103#define LOG_TAG "AudioFlinger::EffectChain" 7104 7105AudioFlinger::EffectChain::EffectChain(const wp<ThreadBase>& wThread, 7106 int sessionId) 7107 : mThread(wThread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7108 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7109 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7110{ 7111 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7112 sp<ThreadBase> thread = mThread.promote(); 7113 if (thread == 0) { 7114 return; 7115 } 7116 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7117 thread->frameCount(); 7118} 7119 7120AudioFlinger::EffectChain::~EffectChain() 7121{ 7122 if (mOwnInBuffer) { 7123 delete mInBuffer; 7124 } 7125 7126} 7127 7128// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7129sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7130{ 7131 sp<EffectModule> effect; 7132 size_t size = mEffects.size(); 7133 7134 for (size_t i = 0; i < size; i++) { 7135 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7136 effect = mEffects[i]; 7137 break; 7138 } 7139 } 7140 return effect; 7141} 7142 7143// getEffectFromId_l() must be called with ThreadBase::mLock held 7144sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7145{ 7146 sp<EffectModule> effect; 7147 size_t size = mEffects.size(); 7148 7149 for (size_t i = 0; i < size; i++) { 7150 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7151 if (id == 0 || mEffects[i]->id() == id) { 7152 effect = mEffects[i]; 7153 break; 7154 } 7155 } 7156 return effect; 7157} 7158 7159// getEffectFromType_l() must be called with ThreadBase::mLock held 7160sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7161 const effect_uuid_t *type) 7162{ 7163 sp<EffectModule> effect; 7164 size_t size = mEffects.size(); 7165 7166 for (size_t i = 0; i < size; i++) { 7167 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7168 effect = mEffects[i]; 7169 break; 7170 } 7171 } 7172 return effect; 7173} 7174 7175// Must be called with EffectChain::mLock locked 7176void AudioFlinger::EffectChain::process_l() 7177{ 7178 sp<ThreadBase> thread = mThread.promote(); 7179 if (thread == 0) { 7180 ALOGW("process_l(): cannot promote mixer thread"); 7181 return; 7182 } 7183 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7184 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7185 // always process effects unless no more tracks are on the session and the effect tail 7186 // has been rendered 7187 bool doProcess = true; 7188 if (!isGlobalSession) { 7189 bool tracksOnSession = (trackCnt() != 0); 7190 7191 if (!tracksOnSession && mTailBufferCount == 0) { 7192 doProcess = false; 7193 } 7194 7195 if (activeTrackCnt() == 0) { 7196 // if no track is active and the effect tail has not been rendered, 7197 // the input buffer must be cleared here as the mixer process will not do it 7198 if (tracksOnSession || mTailBufferCount > 0) { 7199 size_t numSamples = thread->frameCount() * thread->channelCount(); 7200 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7201 if (mTailBufferCount > 0) { 7202 mTailBufferCount--; 7203 } 7204 } 7205 } 7206 } 7207 7208 size_t size = mEffects.size(); 7209 if (doProcess) { 7210 for (size_t i = 0; i < size; i++) { 7211 mEffects[i]->process(); 7212 } 7213 } 7214 for (size_t i = 0; i < size; i++) { 7215 mEffects[i]->updateState(); 7216 } 7217} 7218 7219// addEffect_l() must be called with PlaybackThread::mLock held 7220status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7221{ 7222 effect_descriptor_t desc = effect->desc(); 7223 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7224 7225 Mutex::Autolock _l(mLock); 7226 effect->setChain(this); 7227 sp<ThreadBase> thread = mThread.promote(); 7228 if (thread == 0) { 7229 return NO_INIT; 7230 } 7231 effect->setThread(thread); 7232 7233 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7234 // Auxiliary effects are inserted at the beginning of mEffects vector as 7235 // they are processed first and accumulated in chain input buffer 7236 mEffects.insertAt(effect, 0); 7237 7238 // the input buffer for auxiliary effect contains mono samples in 7239 // 32 bit format. This is to avoid saturation in AudoMixer 7240 // accumulation stage. Saturation is done in EffectModule::process() before 7241 // calling the process in effect engine 7242 size_t numSamples = thread->frameCount(); 7243 int32_t *buffer = new int32_t[numSamples]; 7244 memset(buffer, 0, numSamples * sizeof(int32_t)); 7245 effect->setInBuffer((int16_t *)buffer); 7246 // auxiliary effects output samples to chain input buffer for further processing 7247 // by insert effects 7248 effect->setOutBuffer(mInBuffer); 7249 } else { 7250 // Insert effects are inserted at the end of mEffects vector as they are processed 7251 // after track and auxiliary effects. 7252 // Insert effect order as a function of indicated preference: 7253 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7254 // another effect is present 7255 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7256 // last effect claiming first position 7257 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7258 // first effect claiming last position 7259 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7260 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7261 // already present 7262 7263 int size = (int)mEffects.size(); 7264 int idx_insert = size; 7265 int idx_insert_first = -1; 7266 int idx_insert_last = -1; 7267 7268 for (int i = 0; i < size; i++) { 7269 effect_descriptor_t d = mEffects[i]->desc(); 7270 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7271 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7272 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7273 // check invalid effect chaining combinations 7274 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7275 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7276 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7277 return INVALID_OPERATION; 7278 } 7279 // remember position of first insert effect and by default 7280 // select this as insert position for new effect 7281 if (idx_insert == size) { 7282 idx_insert = i; 7283 } 7284 // remember position of last insert effect claiming 7285 // first position 7286 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7287 idx_insert_first = i; 7288 } 7289 // remember position of first insert effect claiming 7290 // last position 7291 if (iPref == EFFECT_FLAG_INSERT_LAST && 7292 idx_insert_last == -1) { 7293 idx_insert_last = i; 7294 } 7295 } 7296 } 7297 7298 // modify idx_insert from first position if needed 7299 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7300 if (idx_insert_last != -1) { 7301 idx_insert = idx_insert_last; 7302 } else { 7303 idx_insert = size; 7304 } 7305 } else { 7306 if (idx_insert_first != -1) { 7307 idx_insert = idx_insert_first + 1; 7308 } 7309 } 7310 7311 // always read samples from chain input buffer 7312 effect->setInBuffer(mInBuffer); 7313 7314 // if last effect in the chain, output samples to chain 7315 // output buffer, otherwise to chain input buffer 7316 if (idx_insert == size) { 7317 if (idx_insert != 0) { 7318 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7319 mEffects[idx_insert-1]->configure(); 7320 } 7321 effect->setOutBuffer(mOutBuffer); 7322 } else { 7323 effect->setOutBuffer(mInBuffer); 7324 } 7325 mEffects.insertAt(effect, idx_insert); 7326 7327 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7328 } 7329 effect->configure(); 7330 return NO_ERROR; 7331} 7332 7333// removeEffect_l() must be called with PlaybackThread::mLock held 7334size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7335{ 7336 Mutex::Autolock _l(mLock); 7337 int size = (int)mEffects.size(); 7338 int i; 7339 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7340 7341 for (i = 0; i < size; i++) { 7342 if (effect == mEffects[i]) { 7343 // calling stop here will remove pre-processing effect from the audio HAL. 7344 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7345 // the middle of a read from audio HAL 7346 if (mEffects[i]->state() == EffectModule::ACTIVE || 7347 mEffects[i]->state() == EffectModule::STOPPING) { 7348 mEffects[i]->stop(); 7349 } 7350 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7351 delete[] effect->inBuffer(); 7352 } else { 7353 if (i == size - 1 && i != 0) { 7354 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7355 mEffects[i - 1]->configure(); 7356 } 7357 } 7358 mEffects.removeAt(i); 7359 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7360 break; 7361 } 7362 } 7363 7364 return mEffects.size(); 7365} 7366 7367// setDevice_l() must be called with PlaybackThread::mLock held 7368void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7369{ 7370 size_t size = mEffects.size(); 7371 for (size_t i = 0; i < size; i++) { 7372 mEffects[i]->setDevice(device); 7373 } 7374} 7375 7376// setMode_l() must be called with PlaybackThread::mLock held 7377void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7378{ 7379 size_t size = mEffects.size(); 7380 for (size_t i = 0; i < size; i++) { 7381 mEffects[i]->setMode(mode); 7382 } 7383} 7384 7385// setVolume_l() must be called with PlaybackThread::mLock held 7386bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7387{ 7388 uint32_t newLeft = *left; 7389 uint32_t newRight = *right; 7390 bool hasControl = false; 7391 int ctrlIdx = -1; 7392 size_t size = mEffects.size(); 7393 7394 // first update volume controller 7395 for (size_t i = size; i > 0; i--) { 7396 if (mEffects[i - 1]->isProcessEnabled() && 7397 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7398 ctrlIdx = i - 1; 7399 hasControl = true; 7400 break; 7401 } 7402 } 7403 7404 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7405 if (hasControl) { 7406 *left = mNewLeftVolume; 7407 *right = mNewRightVolume; 7408 } 7409 return hasControl; 7410 } 7411 7412 mVolumeCtrlIdx = ctrlIdx; 7413 mLeftVolume = newLeft; 7414 mRightVolume = newRight; 7415 7416 // second get volume update from volume controller 7417 if (ctrlIdx >= 0) { 7418 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7419 mNewLeftVolume = newLeft; 7420 mNewRightVolume = newRight; 7421 } 7422 // then indicate volume to all other effects in chain. 7423 // Pass altered volume to effects before volume controller 7424 // and requested volume to effects after controller 7425 uint32_t lVol = newLeft; 7426 uint32_t rVol = newRight; 7427 7428 for (size_t i = 0; i < size; i++) { 7429 if ((int)i == ctrlIdx) continue; 7430 // this also works for ctrlIdx == -1 when there is no volume controller 7431 if ((int)i > ctrlIdx) { 7432 lVol = *left; 7433 rVol = *right; 7434 } 7435 mEffects[i]->setVolume(&lVol, &rVol, false); 7436 } 7437 *left = newLeft; 7438 *right = newRight; 7439 7440 return hasControl; 7441} 7442 7443status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7444{ 7445 const size_t SIZE = 256; 7446 char buffer[SIZE]; 7447 String8 result; 7448 7449 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7450 result.append(buffer); 7451 7452 bool locked = tryLock(mLock); 7453 // failed to lock - AudioFlinger is probably deadlocked 7454 if (!locked) { 7455 result.append("\tCould not lock mutex:\n"); 7456 } 7457 7458 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7459 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7460 mEffects.size(), 7461 (uint32_t)mInBuffer, 7462 (uint32_t)mOutBuffer, 7463 mActiveTrackCnt); 7464 result.append(buffer); 7465 write(fd, result.string(), result.size()); 7466 7467 for (size_t i = 0; i < mEffects.size(); ++i) { 7468 sp<EffectModule> effect = mEffects[i]; 7469 if (effect != 0) { 7470 effect->dump(fd, args); 7471 } 7472 } 7473 7474 if (locked) { 7475 mLock.unlock(); 7476 } 7477 7478 return NO_ERROR; 7479} 7480 7481// must be called with ThreadBase::mLock held 7482void AudioFlinger::EffectChain::setEffectSuspended_l( 7483 const effect_uuid_t *type, bool suspend) 7484{ 7485 sp<SuspendedEffectDesc> desc; 7486 // use effect type UUID timelow as key as there is no real risk of identical 7487 // timeLow fields among effect type UUIDs. 7488 int index = mSuspendedEffects.indexOfKey(type->timeLow); 7489 if (suspend) { 7490 if (index >= 0) { 7491 desc = mSuspendedEffects.valueAt(index); 7492 } else { 7493 desc = new SuspendedEffectDesc(); 7494 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7495 mSuspendedEffects.add(type->timeLow, desc); 7496 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7497 } 7498 if (desc->mRefCount++ == 0) { 7499 sp<EffectModule> effect = getEffectIfEnabled(type); 7500 if (effect != 0) { 7501 desc->mEffect = effect; 7502 effect->setSuspended(true); 7503 effect->setEnabled(false); 7504 } 7505 } 7506 } else { 7507 if (index < 0) { 7508 return; 7509 } 7510 desc = mSuspendedEffects.valueAt(index); 7511 if (desc->mRefCount <= 0) { 7512 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7513 desc->mRefCount = 1; 7514 } 7515 if (--desc->mRefCount == 0) { 7516 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7517 if (desc->mEffect != 0) { 7518 sp<EffectModule> effect = desc->mEffect.promote(); 7519 if (effect != 0) { 7520 effect->setSuspended(false); 7521 sp<EffectHandle> handle = effect->controlHandle(); 7522 if (handle != 0) { 7523 effect->setEnabled(handle->enabled()); 7524 } 7525 } 7526 desc->mEffect.clear(); 7527 } 7528 mSuspendedEffects.removeItemsAt(index); 7529 } 7530 } 7531} 7532 7533// must be called with ThreadBase::mLock held 7534void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7535{ 7536 sp<SuspendedEffectDesc> desc; 7537 7538 int index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7539 if (suspend) { 7540 if (index >= 0) { 7541 desc = mSuspendedEffects.valueAt(index); 7542 } else { 7543 desc = new SuspendedEffectDesc(); 7544 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7545 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7546 } 7547 if (desc->mRefCount++ == 0) { 7548 Vector< sp<EffectModule> > effects = getSuspendEligibleEffects(); 7549 for (size_t i = 0; i < effects.size(); i++) { 7550 setEffectSuspended_l(&effects[i]->desc().type, true); 7551 } 7552 } 7553 } else { 7554 if (index < 0) { 7555 return; 7556 } 7557 desc = mSuspendedEffects.valueAt(index); 7558 if (desc->mRefCount <= 0) { 7559 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7560 desc->mRefCount = 1; 7561 } 7562 if (--desc->mRefCount == 0) { 7563 Vector<const effect_uuid_t *> types; 7564 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7565 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7566 continue; 7567 } 7568 types.add(&mSuspendedEffects.valueAt(i)->mType); 7569 } 7570 for (size_t i = 0; i < types.size(); i++) { 7571 setEffectSuspended_l(types[i], false); 7572 } 7573 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7574 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7575 } 7576 } 7577} 7578 7579 7580// The volume effect is used for automated tests only 7581#ifndef OPENSL_ES_H_ 7582static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7583 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7584const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7585#endif //OPENSL_ES_H_ 7586 7587bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7588{ 7589 // auxiliary effects and visualizer are never suspended on output mix 7590 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7591 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7592 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7593 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7594 return false; 7595 } 7596 return true; 7597} 7598 7599Vector< sp<AudioFlinger::EffectModule> > AudioFlinger::EffectChain::getSuspendEligibleEffects() 7600{ 7601 Vector< sp<EffectModule> > effects; 7602 for (size_t i = 0; i < mEffects.size(); i++) { 7603 if (!isEffectEligibleForSuspend(mEffects[i]->desc())) { 7604 continue; 7605 } 7606 effects.add(mEffects[i]); 7607 } 7608 return effects; 7609} 7610 7611sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7612 const effect_uuid_t *type) 7613{ 7614 sp<EffectModule> effect; 7615 effect = getEffectFromType_l(type); 7616 if (effect != 0 && !effect->isEnabled()) { 7617 effect.clear(); 7618 } 7619 return effect; 7620} 7621 7622void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 7623 bool enabled) 7624{ 7625 int index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7626 if (enabled) { 7627 if (index < 0) { 7628 // if the effect is not suspend check if all effects are suspended 7629 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7630 if (index < 0) { 7631 return; 7632 } 7633 if (!isEffectEligibleForSuspend(effect->desc())) { 7634 return; 7635 } 7636 setEffectSuspended_l(&effect->desc().type, enabled); 7637 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 7638 if (index < 0) { 7639 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 7640 return; 7641 } 7642 } 7643 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 7644 effect->desc().type.timeLow); 7645 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7646 // if effect is requested to suspended but was not yet enabled, supend it now. 7647 if (desc->mEffect == 0) { 7648 desc->mEffect = effect; 7649 effect->setEnabled(false); 7650 effect->setSuspended(true); 7651 } 7652 } else { 7653 if (index < 0) { 7654 return; 7655 } 7656 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 7657 effect->desc().type.timeLow); 7658 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 7659 desc->mEffect.clear(); 7660 effect->setSuspended(false); 7661 } 7662} 7663 7664#undef LOG_TAG 7665#define LOG_TAG "AudioFlinger" 7666 7667// ---------------------------------------------------------------------------- 7668 7669status_t AudioFlinger::onTransact( 7670 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7671{ 7672 return BnAudioFlinger::onTransact(code, data, reply, flags); 7673} 7674 7675}; // namespace android 7676