AudioFlinger.cpp revision 190a46f7c84e160386610c0c4cecb9767fb5503b
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#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#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <powermanager/PowerManager.h> 61 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63#ifdef DEBUG_CPU_USAGE 64#include <cpustats/CentralTendencyStatistics.h> 65#include <cpustats/ThreadCpuUsage.h> 66#endif 67 68#include <common_time/cc_helper.h> 69#include <common_time/local_clock.h> 70 71// ---------------------------------------------------------------------------- 72 73 74namespace android { 75 76static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 77static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 78 79static const float MAX_GAIN = 4096.0f; 80static const uint32_t MAX_GAIN_INT = 0x1000; 81 82// retry counts for buffer fill timeout 83// 50 * ~20msecs = 1 second 84static const int8_t kMaxTrackRetries = 50; 85static const int8_t kMaxTrackStartupRetries = 50; 86// allow less retry attempts on direct output thread. 87// direct outputs can be a scarce resource in audio hardware and should 88// be released as quickly as possible. 89static const int8_t kMaxTrackRetriesDirect = 2; 90 91static const int kDumpLockRetries = 50; 92static const int kDumpLockSleepUs = 20000; 93 94// don't warn about blocked writes or record buffer overflows more often than this 95static const nsecs_t kWarningThrottleNs = seconds(5); 96 97// RecordThread loop sleep time upon application overrun or audio HAL read error 98static const int kRecordThreadSleepUs = 5000; 99 100// maximum time to wait for setParameters to complete 101static const nsecs_t kSetParametersTimeoutNs = seconds(2); 102 103// minimum sleep time for the mixer thread loop when tracks are active but in underrun 104static const uint32_t kMinThreadSleepTimeUs = 5000; 105// maximum divider applied to the active sleep time in the mixer thread loop 106static const uint32_t kMaxThreadSleepTimeShift = 2; 107 108nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 109 110// ---------------------------------------------------------------------------- 111 112// To collect the amplifier usage 113static void addBatteryData(uint32_t params) { 114 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 115 if (service == NULL) { 116 // it already logged 117 return; 118 } 119 120 service->addBatteryData(params); 121} 122 123static int load_audio_interface(const char *if_name, const hw_module_t **mod, 124 audio_hw_device_t **dev) 125{ 126 int rc; 127 128 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 129 if (rc) 130 goto out; 131 132 rc = audio_hw_device_open(*mod, dev); 133 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 134 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 135 if (rc) 136 goto out; 137 138 return 0; 139 140out: 141 *mod = NULL; 142 *dev = NULL; 143 return rc; 144} 145 146static const char * const audio_interfaces[] = { 147 "primary", 148 "a2dp", 149 "usb", 150}; 151#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 152 153// ---------------------------------------------------------------------------- 154 155AudioFlinger::AudioFlinger() 156 : BnAudioFlinger(), 157 mPrimaryHardwareDev(NULL), 158 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 159 mMasterVolume(1.0f), 160 mMasterVolumeSupportLvl(MVS_NONE), 161 mMasterMute(false), 162 mNextUniqueId(1), 163 mMode(AUDIO_MODE_INVALID), 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 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 176 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 177 uint32_t int_val; 178 if (1 == sscanf(val_str, "%u", &int_val)) { 179 mStandbyTimeInNsecs = milliseconds(int_val); 180 ALOGI("Using %u mSec as standby time.", int_val); 181 } else { 182 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 183 ALOGI("Using default %u mSec as standby time.", 184 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 185 } 186 } 187 188 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 189 const hw_module_t *mod; 190 audio_hw_device_t *dev; 191 192 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 193 if (rc) 194 continue; 195 196 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 197 mod->name, mod->id); 198 mAudioHwDevs.push(dev); 199 200 if (mPrimaryHardwareDev == NULL) { 201 mPrimaryHardwareDev = dev; 202 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 203 mod->name, mod->id, audio_interfaces[i]); 204 } 205 } 206 207 if (mPrimaryHardwareDev == NULL) { 208 ALOGE("Primary audio interface not found"); 209 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 210 } 211 212 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 213 // primary HW dev is selected can change so these conditions might not always be equivalent. 214 // When that happens, re-visit all the code that assumes this. 215 216 AutoMutex lock(mHardwareLock); 217 218 // Determine the level of master volume support the primary audio HAL has, 219 // and set the initial master volume at the same time. 220 float initialVolume = 1.0; 221 mMasterVolumeSupportLvl = MVS_NONE; 222 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 223 audio_hw_device_t *dev = mPrimaryHardwareDev; 224 225 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 226 if ((NULL != dev->get_master_volume) && 227 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 228 mMasterVolumeSupportLvl = MVS_FULL; 229 } else { 230 mMasterVolumeSupportLvl = MVS_SETONLY; 231 initialVolume = 1.0; 232 } 233 234 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 235 if ((NULL == dev->set_master_volume) || 236 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 237 mMasterVolumeSupportLvl = MVS_NONE; 238 } 239 mHardwareStatus = AUDIO_HW_IDLE; 240 } 241 242 // Set the mode for each audio HAL, and try to set the initial volume (if 243 // supported) for all of the non-primary audio HALs. 244 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 245 audio_hw_device_t *dev = mAudioHwDevs[i]; 246 247 mHardwareStatus = AUDIO_HW_INIT; 248 rc = dev->init_check(dev); 249 mHardwareStatus = AUDIO_HW_IDLE; 250 if (rc == 0) { 251 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 252 mHardwareStatus = AUDIO_HW_SET_MODE; 253 dev->set_mode(dev, mMode); 254 255 if ((dev != mPrimaryHardwareDev) && 256 (NULL != dev->set_master_volume)) { 257 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 258 dev->set_master_volume(dev, initialVolume); 259 } 260 261 mHardwareStatus = AUDIO_HW_IDLE; 262 } 263 } 264 265 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 266 ? initialVolume 267 : 1.0; 268 mMasterVolume = initialVolume; 269 mHardwareStatus = AUDIO_HW_IDLE; 270} 271 272AudioFlinger::~AudioFlinger() 273{ 274 275 while (!mRecordThreads.isEmpty()) { 276 // closeInput() will remove first entry from mRecordThreads 277 closeInput(mRecordThreads.keyAt(0)); 278 } 279 while (!mPlaybackThreads.isEmpty()) { 280 // closeOutput() will remove first entry from mPlaybackThreads 281 closeOutput(mPlaybackThreads.keyAt(0)); 282 } 283 284 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 285 // no mHardwareLock needed, as there are no other references to this 286 audio_hw_device_close(mAudioHwDevs[i]); 287 } 288} 289 290audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 291{ 292 /* first matching HW device is returned */ 293 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 294 audio_hw_device_t *dev = mAudioHwDevs[i]; 295 if ((dev->get_supported_devices(dev) & devices) == devices) 296 return dev; 297 } 298 return NULL; 299} 300 301status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 302{ 303 const size_t SIZE = 256; 304 char buffer[SIZE]; 305 String8 result; 306 307 result.append("Clients:\n"); 308 for (size_t i = 0; i < mClients.size(); ++i) { 309 sp<Client> client = mClients.valueAt(i).promote(); 310 if (client != 0) { 311 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 312 result.append(buffer); 313 } 314 } 315 316 result.append("Global session refs:\n"); 317 result.append(" session pid count\n"); 318 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 319 AudioSessionRef *r = mAudioSessionRefs[i]; 320 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 321 result.append(buffer); 322 } 323 write(fd, result.string(), result.size()); 324 return NO_ERROR; 325} 326 327 328status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 329{ 330 const size_t SIZE = 256; 331 char buffer[SIZE]; 332 String8 result; 333 hardware_call_state hardwareStatus = mHardwareStatus; 334 335 snprintf(buffer, SIZE, "Hardware status: %d\n" 336 "Standby Time mSec: %u\n", 337 hardwareStatus, 338 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 339 result.append(buffer); 340 write(fd, result.string(), result.size()); 341 return NO_ERROR; 342} 343 344status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 345{ 346 const size_t SIZE = 256; 347 char buffer[SIZE]; 348 String8 result; 349 snprintf(buffer, SIZE, "Permission Denial: " 350 "can't dump AudioFlinger from pid=%d, uid=%d\n", 351 IPCThreadState::self()->getCallingPid(), 352 IPCThreadState::self()->getCallingUid()); 353 result.append(buffer); 354 write(fd, result.string(), result.size()); 355 return NO_ERROR; 356} 357 358static bool tryLock(Mutex& mutex) 359{ 360 bool locked = false; 361 for (int i = 0; i < kDumpLockRetries; ++i) { 362 if (mutex.tryLock() == NO_ERROR) { 363 locked = true; 364 break; 365 } 366 usleep(kDumpLockSleepUs); 367 } 368 return locked; 369} 370 371status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 372{ 373 if (!dumpAllowed()) { 374 dumpPermissionDenial(fd, args); 375 } else { 376 // get state of hardware lock 377 bool hardwareLocked = tryLock(mHardwareLock); 378 if (!hardwareLocked) { 379 String8 result(kHardwareLockedString); 380 write(fd, result.string(), result.size()); 381 } else { 382 mHardwareLock.unlock(); 383 } 384 385 bool locked = tryLock(mLock); 386 387 // failed to lock - AudioFlinger is probably deadlocked 388 if (!locked) { 389 String8 result(kDeadlockedString); 390 write(fd, result.string(), result.size()); 391 } 392 393 dumpClients(fd, args); 394 dumpInternals(fd, args); 395 396 // dump playback threads 397 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 398 mPlaybackThreads.valueAt(i)->dump(fd, args); 399 } 400 401 // dump record threads 402 for (size_t i = 0; i < mRecordThreads.size(); i++) { 403 mRecordThreads.valueAt(i)->dump(fd, args); 404 } 405 406 // dump all hardware devs 407 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 408 audio_hw_device_t *dev = mAudioHwDevs[i]; 409 dev->dump(dev, fd); 410 } 411 if (locked) mLock.unlock(); 412 } 413 return NO_ERROR; 414} 415 416sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 417{ 418 // If pid is already in the mClients wp<> map, then use that entry 419 // (for which promote() is always != 0), otherwise create a new entry and Client. 420 sp<Client> client = mClients.valueFor(pid).promote(); 421 if (client == 0) { 422 client = new Client(this, pid); 423 mClients.add(pid, client); 424 } 425 426 return client; 427} 428 429// IAudioFlinger interface 430 431 432sp<IAudioTrack> AudioFlinger::createTrack( 433 pid_t pid, 434 audio_stream_type_t streamType, 435 uint32_t sampleRate, 436 audio_format_t format, 437 uint32_t channelMask, 438 int frameCount, 439 // FIXME dead, remove from IAudioFlinger 440 uint32_t flags, 441 const sp<IMemory>& sharedBuffer, 442 audio_io_handle_t output, 443 bool isTimed, 444 int *sessionId, 445 status_t *status) 446{ 447 sp<PlaybackThread::Track> track; 448 sp<TrackHandle> trackHandle; 449 sp<Client> client; 450 status_t lStatus; 451 int lSessionId; 452 453 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 454 // but if someone uses binder directly they could bypass that and cause us to crash 455 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 456 ALOGE("createTrack() invalid stream type %d", streamType); 457 lStatus = BAD_VALUE; 458 goto Exit; 459 } 460 461 { 462 Mutex::Autolock _l(mLock); 463 PlaybackThread *thread = checkPlaybackThread_l(output); 464 PlaybackThread *effectThread = NULL; 465 if (thread == NULL) { 466 ALOGE("unknown output thread"); 467 lStatus = BAD_VALUE; 468 goto Exit; 469 } 470 471 client = registerPid_l(pid); 472 473 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 474 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 475 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 476 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 477 if (mPlaybackThreads.keyAt(i) != output) { 478 // prevent same audio session on different output threads 479 uint32_t sessions = t->hasAudioSession(*sessionId); 480 if (sessions & PlaybackThread::TRACK_SESSION) { 481 ALOGE("createTrack() session ID %d already in use", *sessionId); 482 lStatus = BAD_VALUE; 483 goto Exit; 484 } 485 // check if an effect with same session ID is waiting for a track to be created 486 if (sessions & PlaybackThread::EFFECT_SESSION) { 487 effectThread = t.get(); 488 } 489 } 490 } 491 lSessionId = *sessionId; 492 } else { 493 // if no audio session id is provided, create one here 494 lSessionId = nextUniqueId(); 495 if (sessionId != NULL) { 496 *sessionId = lSessionId; 497 } 498 } 499 ALOGV("createTrack() lSessionId: %d", lSessionId); 500 501 track = thread->createTrack_l(client, streamType, sampleRate, format, 502 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 503 504 // move effect chain to this output thread if an effect on same session was waiting 505 // for a track to be created 506 if (lStatus == NO_ERROR && effectThread != NULL) { 507 Mutex::Autolock _dl(thread->mLock); 508 Mutex::Autolock _sl(effectThread->mLock); 509 moveEffectChain_l(lSessionId, effectThread, thread, true); 510 } 511 } 512 if (lStatus == NO_ERROR) { 513 trackHandle = new TrackHandle(track); 514 } else { 515 // remove local strong reference to Client before deleting the Track so that the Client 516 // destructor is called by the TrackBase destructor with mLock held 517 client.clear(); 518 track.clear(); 519 } 520 521Exit: 522 if (status != NULL) { 523 *status = lStatus; 524 } 525 return trackHandle; 526} 527 528uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 529{ 530 Mutex::Autolock _l(mLock); 531 PlaybackThread *thread = checkPlaybackThread_l(output); 532 if (thread == NULL) { 533 ALOGW("sampleRate() unknown thread %d", output); 534 return 0; 535 } 536 return thread->sampleRate(); 537} 538 539int AudioFlinger::channelCount(audio_io_handle_t output) const 540{ 541 Mutex::Autolock _l(mLock); 542 PlaybackThread *thread = checkPlaybackThread_l(output); 543 if (thread == NULL) { 544 ALOGW("channelCount() unknown thread %d", output); 545 return 0; 546 } 547 return thread->channelCount(); 548} 549 550audio_format_t AudioFlinger::format(audio_io_handle_t output) const 551{ 552 Mutex::Autolock _l(mLock); 553 PlaybackThread *thread = checkPlaybackThread_l(output); 554 if (thread == NULL) { 555 ALOGW("format() unknown thread %d", output); 556 return AUDIO_FORMAT_INVALID; 557 } 558 return thread->format(); 559} 560 561size_t AudioFlinger::frameCount(audio_io_handle_t output) const 562{ 563 Mutex::Autolock _l(mLock); 564 PlaybackThread *thread = checkPlaybackThread_l(output); 565 if (thread == NULL) { 566 ALOGW("frameCount() unknown thread %d", output); 567 return 0; 568 } 569 return thread->frameCount(); 570} 571 572uint32_t AudioFlinger::latency(audio_io_handle_t output) const 573{ 574 Mutex::Autolock _l(mLock); 575 PlaybackThread *thread = checkPlaybackThread_l(output); 576 if (thread == NULL) { 577 ALOGW("latency() unknown thread %d", output); 578 return 0; 579 } 580 return thread->latency(); 581} 582 583status_t AudioFlinger::setMasterVolume(float value) 584{ 585 status_t ret = initCheck(); 586 if (ret != NO_ERROR) { 587 return ret; 588 } 589 590 // check calling permissions 591 if (!settingsAllowed()) { 592 return PERMISSION_DENIED; 593 } 594 595 float swmv = value; 596 597 // when hw supports master volume, don't scale in sw mixer 598 if (MVS_NONE != mMasterVolumeSupportLvl) { 599 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 600 AutoMutex lock(mHardwareLock); 601 audio_hw_device_t *dev = mAudioHwDevs[i]; 602 603 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 604 if (NULL != dev->set_master_volume) { 605 dev->set_master_volume(dev, value); 606 } 607 mHardwareStatus = AUDIO_HW_IDLE; 608 } 609 610 swmv = 1.0; 611 } 612 613 Mutex::Autolock _l(mLock); 614 mMasterVolume = value; 615 mMasterVolumeSW = swmv; 616 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 617 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 618 619 return NO_ERROR; 620} 621 622status_t AudioFlinger::setMode(audio_mode_t mode) 623{ 624 status_t ret = initCheck(); 625 if (ret != NO_ERROR) { 626 return ret; 627 } 628 629 // check calling permissions 630 if (!settingsAllowed()) { 631 return PERMISSION_DENIED; 632 } 633 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 634 ALOGW("Illegal value: setMode(%d)", mode); 635 return BAD_VALUE; 636 } 637 638 { // scope for the lock 639 AutoMutex lock(mHardwareLock); 640 mHardwareStatus = AUDIO_HW_SET_MODE; 641 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 642 mHardwareStatus = AUDIO_HW_IDLE; 643 } 644 645 if (NO_ERROR == ret) { 646 Mutex::Autolock _l(mLock); 647 mMode = mode; 648 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 649 mPlaybackThreads.valueAt(i)->setMode(mode); 650 } 651 652 return ret; 653} 654 655status_t AudioFlinger::setMicMute(bool state) 656{ 657 status_t ret = initCheck(); 658 if (ret != NO_ERROR) { 659 return ret; 660 } 661 662 // check calling permissions 663 if (!settingsAllowed()) { 664 return PERMISSION_DENIED; 665 } 666 667 AutoMutex lock(mHardwareLock); 668 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 669 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 670 mHardwareStatus = AUDIO_HW_IDLE; 671 return ret; 672} 673 674bool AudioFlinger::getMicMute() const 675{ 676 status_t ret = initCheck(); 677 if (ret != NO_ERROR) { 678 return false; 679 } 680 681 bool state = AUDIO_MODE_INVALID; 682 AutoMutex lock(mHardwareLock); 683 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 684 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 685 mHardwareStatus = AUDIO_HW_IDLE; 686 return state; 687} 688 689status_t AudioFlinger::setMasterMute(bool muted) 690{ 691 // check calling permissions 692 if (!settingsAllowed()) { 693 return PERMISSION_DENIED; 694 } 695 696 Mutex::Autolock _l(mLock); 697 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 698 mMasterMute = muted; 699 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 700 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 701 702 return NO_ERROR; 703} 704 705float AudioFlinger::masterVolume() const 706{ 707 Mutex::Autolock _l(mLock); 708 return masterVolume_l(); 709} 710 711float AudioFlinger::masterVolumeSW() const 712{ 713 Mutex::Autolock _l(mLock); 714 return masterVolumeSW_l(); 715} 716 717bool AudioFlinger::masterMute() const 718{ 719 Mutex::Autolock _l(mLock); 720 return masterMute_l(); 721} 722 723float AudioFlinger::masterVolume_l() const 724{ 725 if (MVS_FULL == mMasterVolumeSupportLvl) { 726 float ret_val; 727 AutoMutex lock(mHardwareLock); 728 729 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 730 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 731 (NULL != mPrimaryHardwareDev->get_master_volume), 732 "can't get master volume"); 733 734 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 735 mHardwareStatus = AUDIO_HW_IDLE; 736 return ret_val; 737 } 738 739 return mMasterVolume; 740} 741 742status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 743 audio_io_handle_t output) 744{ 745 // check calling permissions 746 if (!settingsAllowed()) { 747 return PERMISSION_DENIED; 748 } 749 750 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 751 ALOGE("setStreamVolume() invalid stream %d", stream); 752 return BAD_VALUE; 753 } 754 755 AutoMutex lock(mLock); 756 PlaybackThread *thread = NULL; 757 if (output) { 758 thread = checkPlaybackThread_l(output); 759 if (thread == NULL) { 760 return BAD_VALUE; 761 } 762 } 763 764 mStreamTypes[stream].volume = value; 765 766 if (thread == NULL) { 767 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 768 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 769 } 770 } else { 771 thread->setStreamVolume(stream, value); 772 } 773 774 return NO_ERROR; 775} 776 777status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 778{ 779 // check calling permissions 780 if (!settingsAllowed()) { 781 return PERMISSION_DENIED; 782 } 783 784 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 785 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 786 ALOGE("setStreamMute() invalid stream %d", stream); 787 return BAD_VALUE; 788 } 789 790 AutoMutex lock(mLock); 791 mStreamTypes[stream].mute = muted; 792 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 793 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 794 795 return NO_ERROR; 796} 797 798float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 799{ 800 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 801 return 0.0f; 802 } 803 804 AutoMutex lock(mLock); 805 float volume; 806 if (output) { 807 PlaybackThread *thread = checkPlaybackThread_l(output); 808 if (thread == NULL) { 809 return 0.0f; 810 } 811 volume = thread->streamVolume(stream); 812 } else { 813 volume = streamVolume_l(stream); 814 } 815 816 return volume; 817} 818 819bool AudioFlinger::streamMute(audio_stream_type_t stream) const 820{ 821 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 822 return true; 823 } 824 825 AutoMutex lock(mLock); 826 return streamMute_l(stream); 827} 828 829status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 830{ 831 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 832 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 833 // check calling permissions 834 if (!settingsAllowed()) { 835 return PERMISSION_DENIED; 836 } 837 838 // ioHandle == 0 means the parameters are global to the audio hardware interface 839 if (ioHandle == 0) { 840 status_t final_result = NO_ERROR; 841 { 842 AutoMutex lock(mHardwareLock); 843 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 844 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 845 audio_hw_device_t *dev = mAudioHwDevs[i]; 846 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 847 final_result = result ?: final_result; 848 } 849 mHardwareStatus = AUDIO_HW_IDLE; 850 } 851 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 852 AudioParameter param = AudioParameter(keyValuePairs); 853 String8 value; 854 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 855 Mutex::Autolock _l(mLock); 856 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 857 if (mBtNrecIsOff != btNrecIsOff) { 858 for (size_t i = 0; i < mRecordThreads.size(); i++) { 859 sp<RecordThread> thread = mRecordThreads.valueAt(i); 860 RecordThread::RecordTrack *track = thread->track(); 861 if (track != NULL) { 862 audio_devices_t device = (audio_devices_t)( 863 thread->device() & AUDIO_DEVICE_IN_ALL); 864 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 865 thread->setEffectSuspended(FX_IID_AEC, 866 suspend, 867 track->sessionId()); 868 thread->setEffectSuspended(FX_IID_NS, 869 suspend, 870 track->sessionId()); 871 } 872 } 873 mBtNrecIsOff = btNrecIsOff; 874 } 875 } 876 return final_result; 877 } 878 879 // hold a strong ref on thread in case closeOutput() or closeInput() is called 880 // and the thread is exited once the lock is released 881 sp<ThreadBase> thread; 882 { 883 Mutex::Autolock _l(mLock); 884 thread = checkPlaybackThread_l(ioHandle); 885 if (thread == NULL) { 886 thread = checkRecordThread_l(ioHandle); 887 } else if (thread == primaryPlaybackThread_l()) { 888 // indicate output device change to all input threads for pre processing 889 AudioParameter param = AudioParameter(keyValuePairs); 890 int value; 891 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 892 for (size_t i = 0; i < mRecordThreads.size(); i++) { 893 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 894 } 895 } 896 } 897 } 898 if (thread != 0) { 899 return thread->setParameters(keyValuePairs); 900 } 901 return BAD_VALUE; 902} 903 904String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 905{ 906// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 907// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 908 909 if (ioHandle == 0) { 910 String8 out_s8; 911 912 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 913 char *s; 914 { 915 AutoMutex lock(mHardwareLock); 916 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 917 audio_hw_device_t *dev = mAudioHwDevs[i]; 918 s = dev->get_parameters(dev, keys.string()); 919 mHardwareStatus = AUDIO_HW_IDLE; 920 } 921 out_s8 += String8(s ? s : ""); 922 free(s); 923 } 924 return out_s8; 925 } 926 927 Mutex::Autolock _l(mLock); 928 929 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 930 if (playbackThread != NULL) { 931 return playbackThread->getParameters(keys); 932 } 933 RecordThread *recordThread = checkRecordThread_l(ioHandle); 934 if (recordThread != NULL) { 935 return recordThread->getParameters(keys); 936 } 937 return String8(""); 938} 939 940size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 941{ 942 status_t ret = initCheck(); 943 if (ret != NO_ERROR) { 944 return 0; 945 } 946 947 AutoMutex lock(mHardwareLock); 948 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 949 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 950 mHardwareStatus = AUDIO_HW_IDLE; 951 return size; 952} 953 954unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 955{ 956 if (ioHandle == 0) { 957 return 0; 958 } 959 960 Mutex::Autolock _l(mLock); 961 962 RecordThread *recordThread = checkRecordThread_l(ioHandle); 963 if (recordThread != NULL) { 964 return recordThread->getInputFramesLost(); 965 } 966 return 0; 967} 968 969status_t AudioFlinger::setVoiceVolume(float value) 970{ 971 status_t ret = initCheck(); 972 if (ret != NO_ERROR) { 973 return ret; 974 } 975 976 // check calling permissions 977 if (!settingsAllowed()) { 978 return PERMISSION_DENIED; 979 } 980 981 AutoMutex lock(mHardwareLock); 982 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 983 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 984 mHardwareStatus = AUDIO_HW_IDLE; 985 986 return ret; 987} 988 989status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 990 audio_io_handle_t output) const 991{ 992 status_t status; 993 994 Mutex::Autolock _l(mLock); 995 996 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 997 if (playbackThread != NULL) { 998 return playbackThread->getRenderPosition(halFrames, dspFrames); 999 } 1000 1001 return BAD_VALUE; 1002} 1003 1004void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1005{ 1006 1007 Mutex::Autolock _l(mLock); 1008 1009 pid_t pid = IPCThreadState::self()->getCallingPid(); 1010 if (mNotificationClients.indexOfKey(pid) < 0) { 1011 sp<NotificationClient> notificationClient = new NotificationClient(this, 1012 client, 1013 pid); 1014 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1015 1016 mNotificationClients.add(pid, notificationClient); 1017 1018 sp<IBinder> binder = client->asBinder(); 1019 binder->linkToDeath(notificationClient); 1020 1021 // the config change is always sent from playback or record threads to avoid deadlock 1022 // with AudioSystem::gLock 1023 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1024 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1025 } 1026 1027 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1028 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1029 } 1030 } 1031} 1032 1033void AudioFlinger::removeNotificationClient(pid_t pid) 1034{ 1035 Mutex::Autolock _l(mLock); 1036 1037 mNotificationClients.removeItem(pid); 1038 1039 ALOGV("%d died, releasing its sessions", pid); 1040 size_t num = mAudioSessionRefs.size(); 1041 bool removed = false; 1042 for (size_t i = 0; i< num; ) { 1043 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1044 ALOGV(" pid %d @ %d", ref->mPid, i); 1045 if (ref->mPid == pid) { 1046 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1047 mAudioSessionRefs.removeAt(i); 1048 delete ref; 1049 removed = true; 1050 num--; 1051 } else { 1052 i++; 1053 } 1054 } 1055 if (removed) { 1056 purgeStaleEffects_l(); 1057 } 1058} 1059 1060// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1061void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1062{ 1063 size_t size = mNotificationClients.size(); 1064 for (size_t i = 0; i < size; i++) { 1065 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1066 param2); 1067 } 1068} 1069 1070// removeClient_l() must be called with AudioFlinger::mLock held 1071void AudioFlinger::removeClient_l(pid_t pid) 1072{ 1073 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1074 mClients.removeItem(pid); 1075} 1076 1077 1078// ---------------------------------------------------------------------------- 1079 1080AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1081 uint32_t device, type_t type) 1082 : Thread(false), 1083 mType(type), 1084 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1085 // mChannelMask 1086 mChannelCount(0), 1087 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1088 mParamStatus(NO_ERROR), 1089 mStandby(false), mId(id), 1090 mDevice(device), 1091 mDeathRecipient(new PMDeathRecipient(this)) 1092{ 1093} 1094 1095AudioFlinger::ThreadBase::~ThreadBase() 1096{ 1097 mParamCond.broadcast(); 1098 // do not lock the mutex in destructor 1099 releaseWakeLock_l(); 1100 if (mPowerManager != 0) { 1101 sp<IBinder> binder = mPowerManager->asBinder(); 1102 binder->unlinkToDeath(mDeathRecipient); 1103 } 1104} 1105 1106void AudioFlinger::ThreadBase::exit() 1107{ 1108 ALOGV("ThreadBase::exit"); 1109 { 1110 // This lock prevents the following race in thread (uniprocessor for illustration): 1111 // if (!exitPending()) { 1112 // // context switch from here to exit() 1113 // // exit() calls requestExit(), what exitPending() observes 1114 // // exit() calls signal(), which is dropped since no waiters 1115 // // context switch back from exit() to here 1116 // mWaitWorkCV.wait(...); 1117 // // now thread is hung 1118 // } 1119 AutoMutex lock(mLock); 1120 requestExit(); 1121 mWaitWorkCV.signal(); 1122 } 1123 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1124 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1125 requestExitAndWait(); 1126} 1127 1128status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1129{ 1130 status_t status; 1131 1132 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1133 Mutex::Autolock _l(mLock); 1134 1135 mNewParameters.add(keyValuePairs); 1136 mWaitWorkCV.signal(); 1137 // wait condition with timeout in case the thread loop has exited 1138 // before the request could be processed 1139 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1140 status = mParamStatus; 1141 mWaitWorkCV.signal(); 1142 } else { 1143 status = TIMED_OUT; 1144 } 1145 return status; 1146} 1147 1148void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1149{ 1150 Mutex::Autolock _l(mLock); 1151 sendConfigEvent_l(event, param); 1152} 1153 1154// sendConfigEvent_l() must be called with ThreadBase::mLock held 1155void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1156{ 1157 ConfigEvent configEvent; 1158 configEvent.mEvent = event; 1159 configEvent.mParam = param; 1160 mConfigEvents.add(configEvent); 1161 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1162 mWaitWorkCV.signal(); 1163} 1164 1165void AudioFlinger::ThreadBase::processConfigEvents() 1166{ 1167 mLock.lock(); 1168 while (!mConfigEvents.isEmpty()) { 1169 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1170 ConfigEvent configEvent = mConfigEvents[0]; 1171 mConfigEvents.removeAt(0); 1172 // release mLock before locking AudioFlinger mLock: lock order is always 1173 // AudioFlinger then ThreadBase to avoid cross deadlock 1174 mLock.unlock(); 1175 mAudioFlinger->mLock.lock(); 1176 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1177 mAudioFlinger->mLock.unlock(); 1178 mLock.lock(); 1179 } 1180 mLock.unlock(); 1181} 1182 1183status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1184{ 1185 const size_t SIZE = 256; 1186 char buffer[SIZE]; 1187 String8 result; 1188 1189 bool locked = tryLock(mLock); 1190 if (!locked) { 1191 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1192 write(fd, buffer, strlen(buffer)); 1193 } 1194 1195 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1196 result.append(buffer); 1197 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1202 result.append(buffer); 1203 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1204 result.append(buffer); 1205 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1206 result.append(buffer); 1207 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1208 result.append(buffer); 1209 1210 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1211 result.append(buffer); 1212 result.append(" Index Command"); 1213 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1214 snprintf(buffer, SIZE, "\n %02d ", i); 1215 result.append(buffer); 1216 result.append(mNewParameters[i]); 1217 } 1218 1219 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1220 result.append(buffer); 1221 snprintf(buffer, SIZE, " Index event param\n"); 1222 result.append(buffer); 1223 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1224 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1225 result.append(buffer); 1226 } 1227 result.append("\n"); 1228 1229 write(fd, result.string(), result.size()); 1230 1231 if (locked) { 1232 mLock.unlock(); 1233 } 1234 return NO_ERROR; 1235} 1236 1237status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1238{ 1239 const size_t SIZE = 256; 1240 char buffer[SIZE]; 1241 String8 result; 1242 1243 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1244 write(fd, buffer, strlen(buffer)); 1245 1246 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1247 sp<EffectChain> chain = mEffectChains[i]; 1248 if (chain != 0) { 1249 chain->dump(fd, args); 1250 } 1251 } 1252 return NO_ERROR; 1253} 1254 1255void AudioFlinger::ThreadBase::acquireWakeLock() 1256{ 1257 Mutex::Autolock _l(mLock); 1258 acquireWakeLock_l(); 1259} 1260 1261void AudioFlinger::ThreadBase::acquireWakeLock_l() 1262{ 1263 if (mPowerManager == 0) { 1264 // use checkService() to avoid blocking if power service is not up yet 1265 sp<IBinder> binder = 1266 defaultServiceManager()->checkService(String16("power")); 1267 if (binder == 0) { 1268 ALOGW("Thread %s cannot connect to the power manager service", mName); 1269 } else { 1270 mPowerManager = interface_cast<IPowerManager>(binder); 1271 binder->linkToDeath(mDeathRecipient); 1272 } 1273 } 1274 if (mPowerManager != 0) { 1275 sp<IBinder> binder = new BBinder(); 1276 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1277 binder, 1278 String16(mName)); 1279 if (status == NO_ERROR) { 1280 mWakeLockToken = binder; 1281 } 1282 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1283 } 1284} 1285 1286void AudioFlinger::ThreadBase::releaseWakeLock() 1287{ 1288 Mutex::Autolock _l(mLock); 1289 releaseWakeLock_l(); 1290} 1291 1292void AudioFlinger::ThreadBase::releaseWakeLock_l() 1293{ 1294 if (mWakeLockToken != 0) { 1295 ALOGV("releaseWakeLock_l() %s", mName); 1296 if (mPowerManager != 0) { 1297 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1298 } 1299 mWakeLockToken.clear(); 1300 } 1301} 1302 1303void AudioFlinger::ThreadBase::clearPowerManager() 1304{ 1305 Mutex::Autolock _l(mLock); 1306 releaseWakeLock_l(); 1307 mPowerManager.clear(); 1308} 1309 1310void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1311{ 1312 sp<ThreadBase> thread = mThread.promote(); 1313 if (thread != 0) { 1314 thread->clearPowerManager(); 1315 } 1316 ALOGW("power manager service died !!!"); 1317} 1318 1319void AudioFlinger::ThreadBase::setEffectSuspended( 1320 const effect_uuid_t *type, bool suspend, int sessionId) 1321{ 1322 Mutex::Autolock _l(mLock); 1323 setEffectSuspended_l(type, suspend, sessionId); 1324} 1325 1326void AudioFlinger::ThreadBase::setEffectSuspended_l( 1327 const effect_uuid_t *type, bool suspend, int sessionId) 1328{ 1329 sp<EffectChain> chain = getEffectChain_l(sessionId); 1330 if (chain != 0) { 1331 if (type != NULL) { 1332 chain->setEffectSuspended_l(type, suspend); 1333 } else { 1334 chain->setEffectSuspendedAll_l(suspend); 1335 } 1336 } 1337 1338 updateSuspendedSessions_l(type, suspend, sessionId); 1339} 1340 1341void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1342{ 1343 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1344 if (index < 0) { 1345 return; 1346 } 1347 1348 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1349 mSuspendedSessions.editValueAt(index); 1350 1351 for (size_t i = 0; i < sessionEffects.size(); i++) { 1352 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1353 for (int j = 0; j < desc->mRefCount; j++) { 1354 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1355 chain->setEffectSuspendedAll_l(true); 1356 } else { 1357 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1358 desc->mType.timeLow); 1359 chain->setEffectSuspended_l(&desc->mType, true); 1360 } 1361 } 1362 } 1363} 1364 1365void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1366 bool suspend, 1367 int sessionId) 1368{ 1369 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1370 1371 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1372 1373 if (suspend) { 1374 if (index >= 0) { 1375 sessionEffects = mSuspendedSessions.editValueAt(index); 1376 } else { 1377 mSuspendedSessions.add(sessionId, sessionEffects); 1378 } 1379 } else { 1380 if (index < 0) { 1381 return; 1382 } 1383 sessionEffects = mSuspendedSessions.editValueAt(index); 1384 } 1385 1386 1387 int key = EffectChain::kKeyForSuspendAll; 1388 if (type != NULL) { 1389 key = type->timeLow; 1390 } 1391 index = sessionEffects.indexOfKey(key); 1392 1393 sp<SuspendedSessionDesc> desc; 1394 if (suspend) { 1395 if (index >= 0) { 1396 desc = sessionEffects.valueAt(index); 1397 } else { 1398 desc = new SuspendedSessionDesc(); 1399 if (type != NULL) { 1400 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1401 } 1402 sessionEffects.add(key, desc); 1403 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1404 } 1405 desc->mRefCount++; 1406 } else { 1407 if (index < 0) { 1408 return; 1409 } 1410 desc = sessionEffects.valueAt(index); 1411 if (--desc->mRefCount == 0) { 1412 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1413 sessionEffects.removeItemsAt(index); 1414 if (sessionEffects.isEmpty()) { 1415 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1416 sessionId); 1417 mSuspendedSessions.removeItem(sessionId); 1418 } 1419 } 1420 } 1421 if (!sessionEffects.isEmpty()) { 1422 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1423 } 1424} 1425 1426void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1427 bool enabled, 1428 int sessionId) 1429{ 1430 Mutex::Autolock _l(mLock); 1431 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1432} 1433 1434void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1435 bool enabled, 1436 int sessionId) 1437{ 1438 if (mType != RECORD) { 1439 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1440 // another session. This gives the priority to well behaved effect control panels 1441 // and applications not using global effects. 1442 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1443 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1444 } 1445 } 1446 1447 sp<EffectChain> chain = getEffectChain_l(sessionId); 1448 if (chain != 0) { 1449 chain->checkSuspendOnEffectEnabled(effect, enabled); 1450 } 1451} 1452 1453// ---------------------------------------------------------------------------- 1454 1455AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1456 AudioStreamOut* output, 1457 audio_io_handle_t id, 1458 uint32_t device, 1459 type_t type) 1460 : ThreadBase(audioFlinger, id, device, type), 1461 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1462 // Assumes constructor is called by AudioFlinger with it's mLock held, 1463 // but it would be safer to explicitly pass initial masterMute as parameter 1464 mMasterMute(audioFlinger->masterMute_l()), 1465 // mStreamTypes[] initialized in constructor body 1466 mOutput(output), 1467 // Assumes constructor is called by AudioFlinger with it's mLock held, 1468 // but it would be safer to explicitly pass initial masterVolume as parameter 1469 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1470 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1471 mMixerStatus(MIXER_IDLE), 1472 mPrevMixerStatus(MIXER_IDLE), 1473 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1474{ 1475 snprintf(mName, kNameLength, "AudioOut_%X", id); 1476 1477 readOutputParameters(); 1478 1479 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1480 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1481 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1482 stream = (audio_stream_type_t) (stream + 1)) { 1483 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1484 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1485 // initialized by stream_type_t default constructor 1486 // mStreamTypes[stream].valid = true; 1487 } 1488 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1489 // because mAudioFlinger doesn't have one to copy from 1490} 1491 1492AudioFlinger::PlaybackThread::~PlaybackThread() 1493{ 1494 delete [] mMixBuffer; 1495} 1496 1497status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1498{ 1499 dumpInternals(fd, args); 1500 dumpTracks(fd, args); 1501 dumpEffectChains(fd, args); 1502 return NO_ERROR; 1503} 1504 1505status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1506{ 1507 const size_t SIZE = 256; 1508 char buffer[SIZE]; 1509 String8 result; 1510 1511 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1512 result.append(buffer); 1513 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1514 for (size_t i = 0; i < mTracks.size(); ++i) { 1515 sp<Track> track = mTracks[i]; 1516 if (track != 0) { 1517 track->dump(buffer, SIZE); 1518 result.append(buffer); 1519 } 1520 } 1521 1522 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1523 result.append(buffer); 1524 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1525 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1526 sp<Track> track = mActiveTracks[i].promote(); 1527 if (track != 0) { 1528 track->dump(buffer, SIZE); 1529 result.append(buffer); 1530 } 1531 } 1532 write(fd, result.string(), result.size()); 1533 return NO_ERROR; 1534} 1535 1536status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1537{ 1538 const size_t SIZE = 256; 1539 char buffer[SIZE]; 1540 String8 result; 1541 1542 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1545 result.append(buffer); 1546 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1547 result.append(buffer); 1548 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1549 result.append(buffer); 1550 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1551 result.append(buffer); 1552 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1553 result.append(buffer); 1554 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1555 result.append(buffer); 1556 write(fd, result.string(), result.size()); 1557 1558 dumpBase(fd, args); 1559 1560 return NO_ERROR; 1561} 1562 1563// Thread virtuals 1564status_t AudioFlinger::PlaybackThread::readyToRun() 1565{ 1566 status_t status = initCheck(); 1567 if (status == NO_ERROR) { 1568 ALOGI("AudioFlinger's thread %p ready to run", this); 1569 } else { 1570 ALOGE("No working audio driver found."); 1571 } 1572 return status; 1573} 1574 1575void AudioFlinger::PlaybackThread::onFirstRef() 1576{ 1577 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1578} 1579 1580// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1581sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1582 const sp<AudioFlinger::Client>& client, 1583 audio_stream_type_t streamType, 1584 uint32_t sampleRate, 1585 audio_format_t format, 1586 uint32_t channelMask, 1587 int frameCount, 1588 const sp<IMemory>& sharedBuffer, 1589 int sessionId, 1590 bool isTimed, 1591 status_t *status) 1592{ 1593 sp<Track> track; 1594 status_t lStatus; 1595 1596 if (mType == DIRECT) { 1597 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1598 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1599 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1600 "for output %p with format %d", 1601 sampleRate, format, channelMask, mOutput, mFormat); 1602 lStatus = BAD_VALUE; 1603 goto Exit; 1604 } 1605 } 1606 } else { 1607 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1608 if (sampleRate > mSampleRate*2) { 1609 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1610 lStatus = BAD_VALUE; 1611 goto Exit; 1612 } 1613 } 1614 1615 lStatus = initCheck(); 1616 if (lStatus != NO_ERROR) { 1617 ALOGE("Audio driver not initialized."); 1618 goto Exit; 1619 } 1620 1621 { // scope for mLock 1622 Mutex::Autolock _l(mLock); 1623 1624 // all tracks in same audio session must share the same routing strategy otherwise 1625 // conflicts will happen when tracks are moved from one output to another by audio policy 1626 // manager 1627 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1628 for (size_t i = 0; i < mTracks.size(); ++i) { 1629 sp<Track> t = mTracks[i]; 1630 if (t != 0 && !t->isOutputTrack()) { 1631 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1632 if (sessionId == t->sessionId() && strategy != actual) { 1633 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1634 strategy, actual); 1635 lStatus = BAD_VALUE; 1636 goto Exit; 1637 } 1638 } 1639 } 1640 1641 if (!isTimed) { 1642 track = new Track(this, client, streamType, sampleRate, format, 1643 channelMask, frameCount, sharedBuffer, sessionId); 1644 } else { 1645 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1646 channelMask, frameCount, sharedBuffer, sessionId); 1647 } 1648 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1649 lStatus = NO_MEMORY; 1650 goto Exit; 1651 } 1652 mTracks.add(track); 1653 1654 sp<EffectChain> chain = getEffectChain_l(sessionId); 1655 if (chain != 0) { 1656 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1657 track->setMainBuffer(chain->inBuffer()); 1658 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1659 chain->incTrackCnt(); 1660 } 1661 1662 // invalidate track immediately if the stream type was moved to another thread since 1663 // createTrack() was called by the client process. 1664 if (!mStreamTypes[streamType].valid) { 1665 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1666 this, streamType); 1667 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1668 } 1669 } 1670 lStatus = NO_ERROR; 1671 1672Exit: 1673 if (status) { 1674 *status = lStatus; 1675 } 1676 return track; 1677} 1678 1679uint32_t AudioFlinger::PlaybackThread::latency() const 1680{ 1681 Mutex::Autolock _l(mLock); 1682 if (initCheck() == NO_ERROR) { 1683 return mOutput->stream->get_latency(mOutput->stream); 1684 } else { 1685 return 0; 1686 } 1687} 1688 1689void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1690{ 1691 Mutex::Autolock _l(mLock); 1692 mMasterVolume = value; 1693} 1694 1695void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1696{ 1697 Mutex::Autolock _l(mLock); 1698 setMasterMute_l(muted); 1699} 1700 1701void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1702{ 1703 Mutex::Autolock _l(mLock); 1704 mStreamTypes[stream].volume = value; 1705} 1706 1707void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1708{ 1709 Mutex::Autolock _l(mLock); 1710 mStreamTypes[stream].mute = muted; 1711} 1712 1713float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1714{ 1715 Mutex::Autolock _l(mLock); 1716 return mStreamTypes[stream].volume; 1717} 1718 1719// addTrack_l() must be called with ThreadBase::mLock held 1720status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1721{ 1722 status_t status = ALREADY_EXISTS; 1723 1724 // set retry count for buffer fill 1725 track->mRetryCount = kMaxTrackStartupRetries; 1726 if (mActiveTracks.indexOf(track) < 0) { 1727 // the track is newly added, make sure it fills up all its 1728 // buffers before playing. This is to ensure the client will 1729 // effectively get the latency it requested. 1730 track->mFillingUpStatus = Track::FS_FILLING; 1731 track->mResetDone = false; 1732 mActiveTracks.add(track); 1733 if (track->mainBuffer() != mMixBuffer) { 1734 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1735 if (chain != 0) { 1736 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1737 chain->incActiveTrackCnt(); 1738 } 1739 } 1740 1741 status = NO_ERROR; 1742 } 1743 1744 ALOGV("mWaitWorkCV.broadcast"); 1745 mWaitWorkCV.broadcast(); 1746 1747 return status; 1748} 1749 1750// destroyTrack_l() must be called with ThreadBase::mLock held 1751void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1752{ 1753 track->mState = TrackBase::TERMINATED; 1754 if (mActiveTracks.indexOf(track) < 0) { 1755 removeTrack_l(track); 1756 } 1757} 1758 1759void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1760{ 1761 mTracks.remove(track); 1762 deleteTrackName_l(track->name()); 1763 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1764 if (chain != 0) { 1765 chain->decTrackCnt(); 1766 } 1767} 1768 1769String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1770{ 1771 String8 out_s8 = String8(""); 1772 char *s; 1773 1774 Mutex::Autolock _l(mLock); 1775 if (initCheck() != NO_ERROR) { 1776 return out_s8; 1777 } 1778 1779 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1780 out_s8 = String8(s); 1781 free(s); 1782 return out_s8; 1783} 1784 1785// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1786void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1787 AudioSystem::OutputDescriptor desc; 1788 void *param2 = NULL; 1789 1790 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1791 1792 switch (event) { 1793 case AudioSystem::OUTPUT_OPENED: 1794 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1795 desc.channels = mChannelMask; 1796 desc.samplingRate = mSampleRate; 1797 desc.format = mFormat; 1798 desc.frameCount = mFrameCount; 1799 desc.latency = latency(); 1800 param2 = &desc; 1801 break; 1802 1803 case AudioSystem::STREAM_CONFIG_CHANGED: 1804 param2 = ¶m; 1805 case AudioSystem::OUTPUT_CLOSED: 1806 default: 1807 break; 1808 } 1809 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1810} 1811 1812void AudioFlinger::PlaybackThread::readOutputParameters() 1813{ 1814 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1815 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1816 mChannelCount = (uint16_t)popcount(mChannelMask); 1817 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1818 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1819 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1820 1821 // FIXME - Current mixer implementation only supports stereo output: Always 1822 // Allocate a stereo buffer even if HW output is mono. 1823 delete[] mMixBuffer; 1824 mMixBuffer = new int16_t[mFrameCount * 2]; 1825 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1826 1827 // force reconfiguration of effect chains and engines to take new buffer size and audio 1828 // parameters into account 1829 // Note that mLock is not held when readOutputParameters() is called from the constructor 1830 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1831 // matter. 1832 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1833 Vector< sp<EffectChain> > effectChains = mEffectChains; 1834 for (size_t i = 0; i < effectChains.size(); i ++) { 1835 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1836 } 1837} 1838 1839status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1840{ 1841 if (halFrames == NULL || dspFrames == NULL) { 1842 return BAD_VALUE; 1843 } 1844 Mutex::Autolock _l(mLock); 1845 if (initCheck() != NO_ERROR) { 1846 return INVALID_OPERATION; 1847 } 1848 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1849 1850 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1851} 1852 1853uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1854{ 1855 Mutex::Autolock _l(mLock); 1856 uint32_t result = 0; 1857 if (getEffectChain_l(sessionId) != 0) { 1858 result = EFFECT_SESSION; 1859 } 1860 1861 for (size_t i = 0; i < mTracks.size(); ++i) { 1862 sp<Track> track = mTracks[i]; 1863 if (sessionId == track->sessionId() && 1864 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1865 result |= TRACK_SESSION; 1866 break; 1867 } 1868 } 1869 1870 return result; 1871} 1872 1873uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1874{ 1875 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1876 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1877 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1878 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1879 } 1880 for (size_t i = 0; i < mTracks.size(); i++) { 1881 sp<Track> track = mTracks[i]; 1882 if (sessionId == track->sessionId() && 1883 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1884 return AudioSystem::getStrategyForStream(track->streamType()); 1885 } 1886 } 1887 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1888} 1889 1890 1891AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1892{ 1893 Mutex::Autolock _l(mLock); 1894 return mOutput; 1895} 1896 1897AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1898{ 1899 Mutex::Autolock _l(mLock); 1900 AudioStreamOut *output = mOutput; 1901 mOutput = NULL; 1902 return output; 1903} 1904 1905// this method must always be called either with ThreadBase mLock held or inside the thread loop 1906audio_stream_t* AudioFlinger::PlaybackThread::stream() 1907{ 1908 if (mOutput == NULL) { 1909 return NULL; 1910 } 1911 return &mOutput->stream->common; 1912} 1913 1914uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1915{ 1916 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1917 // decoding and transfer time. So sleeping for half of the latency would likely cause 1918 // underruns 1919 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1920 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1921 } else { 1922 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1923 } 1924} 1925 1926// ---------------------------------------------------------------------------- 1927 1928AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1929 audio_io_handle_t id, uint32_t device, type_t type) 1930 : PlaybackThread(audioFlinger, output, id, device, type) 1931{ 1932 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1933 // FIXME - Current mixer implementation only supports stereo output 1934 if (mChannelCount == 1) { 1935 ALOGE("Invalid audio hardware channel count"); 1936 } 1937} 1938 1939AudioFlinger::MixerThread::~MixerThread() 1940{ 1941 delete mAudioMixer; 1942} 1943 1944class CpuStats { 1945public: 1946 CpuStats(); 1947 void sample(const String8 &title); 1948#ifdef DEBUG_CPU_USAGE 1949private: 1950 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 1951 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 1952 1953 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 1954 1955 int mCpuNum; // thread's current CPU number 1956 int mCpukHz; // frequency of thread's current CPU in kHz 1957#endif 1958}; 1959 1960CpuStats::CpuStats() 1961#ifdef DEBUG_CPU_USAGE 1962 : mCpuNum(-1), mCpukHz(-1) 1963#endif 1964{ 1965} 1966 1967void CpuStats::sample(const String8 &title) { 1968#ifdef DEBUG_CPU_USAGE 1969 // get current thread's delta CPU time in wall clock ns 1970 double wcNs; 1971 bool valid = mCpuUsage.sampleAndEnable(wcNs); 1972 1973 // record sample for wall clock statistics 1974 if (valid) { 1975 mWcStats.sample(wcNs); 1976 } 1977 1978 // get the current CPU number 1979 int cpuNum = sched_getcpu(); 1980 1981 // get the current CPU frequency in kHz 1982 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 1983 1984 // check if either CPU number or frequency changed 1985 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 1986 mCpuNum = cpuNum; 1987 mCpukHz = cpukHz; 1988 // ignore sample for purposes of cycles 1989 valid = false; 1990 } 1991 1992 // if no change in CPU number or frequency, then record sample for cycle statistics 1993 if (valid && mCpukHz > 0) { 1994 double cycles = wcNs * cpukHz * 0.000001; 1995 mHzStats.sample(cycles); 1996 } 1997 1998 unsigned n = mWcStats.n(); 1999 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2000 if ((n & 127) == 1) { 2001 long long elapsed = mCpuUsage.elapsed(); 2002 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2003 double perLoop = elapsed / (double) n; 2004 double perLoop100 = perLoop * 0.01; 2005 double perLoop1k = perLoop * 0.001; 2006 double mean = mWcStats.mean(); 2007 double stddev = mWcStats.stddev(); 2008 double minimum = mWcStats.minimum(); 2009 double maximum = mWcStats.maximum(); 2010 double meanCycles = mHzStats.mean(); 2011 double stddevCycles = mHzStats.stddev(); 2012 double minCycles = mHzStats.minimum(); 2013 double maxCycles = mHzStats.maximum(); 2014 mCpuUsage.resetElapsed(); 2015 mWcStats.reset(); 2016 mHzStats.reset(); 2017 ALOGD("CPU usage for %s over past %.1f secs\n" 2018 " (%u mixer loops at %.1f mean ms per loop):\n" 2019 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2020 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2021 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2022 title.string(), 2023 elapsed * .000000001, n, perLoop * .000001, 2024 mean * .001, 2025 stddev * .001, 2026 minimum * .001, 2027 maximum * .001, 2028 mean / perLoop100, 2029 stddev / perLoop100, 2030 minimum / perLoop100, 2031 maximum / perLoop100, 2032 meanCycles / perLoop1k, 2033 stddevCycles / perLoop1k, 2034 minCycles / perLoop1k, 2035 maxCycles / perLoop1k); 2036 2037 } 2038 } 2039#endif 2040}; 2041 2042void AudioFlinger::PlaybackThread::checkSilentMode_l() 2043{ 2044 if (!mMasterMute) { 2045 char value[PROPERTY_VALUE_MAX]; 2046 if (property_get("ro.audio.silent", value, "0") > 0) { 2047 char *endptr; 2048 unsigned long ul = strtoul(value, &endptr, 0); 2049 if (*endptr == '\0' && ul != 0) { 2050 ALOGD("Silence is golden"); 2051 // The setprop command will not allow a property to be changed after 2052 // the first time it is set, so we don't have to worry about un-muting. 2053 setMasterMute_l(true); 2054 } 2055 } 2056 } 2057} 2058 2059bool AudioFlinger::PlaybackThread::threadLoop() 2060{ 2061 Vector< sp<Track> > tracksToRemove; 2062 2063 standbyTime = systemTime(); 2064 2065 // MIXER 2066 nsecs_t lastWarning = 0; 2067if (mType == MIXER) { 2068 longStandbyExit = false; 2069} 2070 2071 // DUPLICATING 2072 // FIXME could this be made local to while loop? 2073 writeFrames = 0; 2074 2075 cacheParameters_l(); 2076 sleepTime = idleSleepTime; 2077 2078if (mType == MIXER) { 2079 sleepTimeShift = 0; 2080} 2081 2082 CpuStats cpuStats; 2083 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2084 2085 acquireWakeLock(); 2086 2087 while (!exitPending()) 2088 { 2089 cpuStats.sample(myName); 2090 2091 Vector< sp<EffectChain> > effectChains; 2092 2093 processConfigEvents(); 2094 2095 { // scope for mLock 2096 2097 Mutex::Autolock _l(mLock); 2098 2099 if (checkForNewParameters_l()) { 2100 cacheParameters_l(); 2101 } 2102 2103 saveOutputTracks(); 2104 2105 // put audio hardware into standby after short delay 2106 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2107 mSuspended > 0)) { 2108 if (!mStandby) { 2109 2110 threadLoop_standby(); 2111 2112 mStandby = true; 2113 mBytesWritten = 0; 2114 } 2115 2116 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2117 // we're about to wait, flush the binder command buffer 2118 IPCThreadState::self()->flushCommands(); 2119 2120 clearOutputTracks(); 2121 2122 if (exitPending()) break; 2123 2124 releaseWakeLock_l(); 2125 // wait until we have something to do... 2126 ALOGV("%s going to sleep", myName.string()); 2127 mWaitWorkCV.wait(mLock); 2128 ALOGV("%s waking up", myName.string()); 2129 acquireWakeLock_l(); 2130 2131 mPrevMixerStatus = MIXER_IDLE; 2132 2133 checkSilentMode_l(); 2134 2135 standbyTime = systemTime() + standbyDelay; 2136 sleepTime = idleSleepTime; 2137 if (mType == MIXER) { 2138 sleepTimeShift = 0; 2139 } 2140 2141 continue; 2142 } 2143 } 2144 2145 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2146 // Shift in the new status; this could be a queue if it's 2147 // useful to filter the mixer status over several cycles. 2148 mPrevMixerStatus = mMixerStatus; 2149 mMixerStatus = newMixerStatus; 2150 2151 // prevent any changes in effect chain list and in each effect chain 2152 // during mixing and effect process as the audio buffers could be deleted 2153 // or modified if an effect is created or deleted 2154 lockEffectChains_l(effectChains); 2155 } 2156 2157 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2158 threadLoop_mix(); 2159 } else { 2160 threadLoop_sleepTime(); 2161 } 2162 2163 if (mSuspended > 0) { 2164 sleepTime = suspendSleepTimeUs(); 2165 } 2166 2167 // only process effects if we're going to write 2168 if (sleepTime == 0) { 2169 for (size_t i = 0; i < effectChains.size(); i ++) { 2170 effectChains[i]->process_l(); 2171 } 2172 } 2173 2174 // enable changes in effect chain 2175 unlockEffectChains(effectChains); 2176 2177 // sleepTime == 0 means we must write to audio hardware 2178 if (sleepTime == 0) { 2179 2180 threadLoop_write(); 2181 2182if (mType == MIXER) { 2183 // write blocked detection 2184 nsecs_t now = systemTime(); 2185 nsecs_t delta = now - mLastWriteTime; 2186 if (!mStandby && delta > maxPeriod) { 2187 mNumDelayedWrites++; 2188 if ((now - lastWarning) > kWarningThrottleNs) { 2189 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2190 ns2ms(delta), mNumDelayedWrites, this); 2191 lastWarning = now; 2192 } 2193 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2194 // a different threshold. Or completely removed for what it is worth anyway... 2195 if (mStandby) { 2196 longStandbyExit = true; 2197 } 2198 } 2199} 2200 2201 mStandby = false; 2202 } else { 2203 usleep(sleepTime); 2204 } 2205 2206 // finally let go of removed track(s), without the lock held 2207 // since we can't guarantee the destructors won't acquire that 2208 // same lock. 2209 tracksToRemove.clear(); 2210 2211 // FIXME I don't understand the need for this here; 2212 // it was in the original code but maybe the 2213 // assignment in saveOutputTracks() makes this unnecessary? 2214 clearOutputTracks(); 2215 2216 // Effect chains will be actually deleted here if they were removed from 2217 // mEffectChains list during mixing or effects processing 2218 effectChains.clear(); 2219 2220 // FIXME Note that the above .clear() is no longer necessary since effectChains 2221 // is now local to this block, but will keep it for now (at least until merge done). 2222 } 2223 2224if (mType == MIXER || mType == DIRECT) { 2225 // put output stream into standby mode 2226 if (!mStandby) { 2227 mOutput->stream->common.standby(&mOutput->stream->common); 2228 } 2229} 2230if (mType == DUPLICATING) { 2231 // for DuplicatingThread, standby mode is handled by the outputTracks 2232} 2233 2234 releaseWakeLock(); 2235 2236 ALOGV("Thread %p type %d exiting", this, mType); 2237 return false; 2238} 2239 2240// shared by MIXER and DIRECT, overridden by DUPLICATING 2241void AudioFlinger::PlaybackThread::threadLoop_write() 2242{ 2243 // FIXME rewrite to reduce number of system calls 2244 mLastWriteTime = systemTime(); 2245 mInWrite = true; 2246 mBytesWritten += mixBufferSize; 2247 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2248 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2249 mNumWrites++; 2250 mInWrite = false; 2251} 2252 2253// shared by MIXER and DIRECT, overridden by DUPLICATING 2254void AudioFlinger::PlaybackThread::threadLoop_standby() 2255{ 2256 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2257 mOutput->stream->common.standby(&mOutput->stream->common); 2258} 2259 2260void AudioFlinger::MixerThread::threadLoop_mix() 2261{ 2262 // obtain the presentation timestamp of the next output buffer 2263 int64_t pts; 2264 status_t status = INVALID_OPERATION; 2265 2266 if (NULL != mOutput->stream->get_next_write_timestamp) { 2267 status = mOutput->stream->get_next_write_timestamp( 2268 mOutput->stream, &pts); 2269 } 2270 2271 if (status != NO_ERROR) { 2272 pts = AudioBufferProvider::kInvalidPTS; 2273 } 2274 2275 // mix buffers... 2276 mAudioMixer->process(pts); 2277 // increase sleep time progressively when application underrun condition clears. 2278 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2279 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2280 // such that we would underrun the audio HAL. 2281 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2282 sleepTimeShift--; 2283 } 2284 sleepTime = 0; 2285 standbyTime = systemTime() + standbyDelay; 2286 //TODO: delay standby when effects have a tail 2287} 2288 2289void AudioFlinger::MixerThread::threadLoop_sleepTime() 2290{ 2291 // If no tracks are ready, sleep once for the duration of an output 2292 // buffer size, then write 0s to the output 2293 if (sleepTime == 0) { 2294 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2295 sleepTime = activeSleepTime >> sleepTimeShift; 2296 if (sleepTime < kMinThreadSleepTimeUs) { 2297 sleepTime = kMinThreadSleepTimeUs; 2298 } 2299 // reduce sleep time in case of consecutive application underruns to avoid 2300 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2301 // duration we would end up writing less data than needed by the audio HAL if 2302 // the condition persists. 2303 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2304 sleepTimeShift++; 2305 } 2306 } else { 2307 sleepTime = idleSleepTime; 2308 } 2309 } else if (mBytesWritten != 0 || 2310 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2311 memset (mMixBuffer, 0, mixBufferSize); 2312 sleepTime = 0; 2313 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2314 } 2315 // TODO add standby time extension fct of effect tail 2316} 2317 2318// prepareTracks_l() must be called with ThreadBase::mLock held 2319AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2320 Vector< sp<Track> > *tracksToRemove) 2321{ 2322 2323 mixer_state mixerStatus = MIXER_IDLE; 2324 // find out which tracks need to be processed 2325 size_t count = mActiveTracks.size(); 2326 size_t mixedTracks = 0; 2327 size_t tracksWithEffect = 0; 2328 2329 float masterVolume = mMasterVolume; 2330 bool masterMute = mMasterMute; 2331 2332 if (masterMute) { 2333 masterVolume = 0; 2334 } 2335 // Delegate master volume control to effect in output mix effect chain if needed 2336 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2337 if (chain != 0) { 2338 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2339 chain->setVolume_l(&v, &v); 2340 masterVolume = (float)((v + (1 << 23)) >> 24); 2341 chain.clear(); 2342 } 2343 2344 for (size_t i=0 ; i<count ; i++) { 2345 sp<Track> t = mActiveTracks[i].promote(); 2346 if (t == 0) continue; 2347 2348 // this const just means the local variable doesn't change 2349 Track* const track = t.get(); 2350 audio_track_cblk_t* cblk = track->cblk(); 2351 2352 // The first time a track is added we wait 2353 // for all its buffers to be filled before processing it 2354 int name = track->name(); 2355 // make sure that we have enough frames to mix one full buffer. 2356 // enforce this condition only once to enable draining the buffer in case the client 2357 // app does not call stop() and relies on underrun to stop: 2358 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2359 // during last round 2360 uint32_t minFrames = 1; 2361 if (!track->isStopped() && !track->isPausing() && 2362 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2363 if (t->sampleRate() == (int)mSampleRate) { 2364 minFrames = mFrameCount; 2365 } else { 2366 // +1 for rounding and +1 for additional sample needed for interpolation 2367 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2368 // add frames already consumed but not yet released by the resampler 2369 // because cblk->framesReady() will include these frames 2370 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2371 // the minimum track buffer size is normally twice the number of frames necessary 2372 // to fill one buffer and the resampler should not leave more than one buffer worth 2373 // of unreleased frames after each pass, but just in case... 2374 ALOG_ASSERT(minFrames <= cblk->frameCount); 2375 } 2376 } 2377 if ((track->framesReady() >= minFrames) && track->isReady() && 2378 !track->isPaused() && !track->isTerminated()) 2379 { 2380 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2381 2382 mixedTracks++; 2383 2384 // track->mainBuffer() != mMixBuffer means there is an effect chain 2385 // connected to the track 2386 chain.clear(); 2387 if (track->mainBuffer() != mMixBuffer) { 2388 chain = getEffectChain_l(track->sessionId()); 2389 // Delegate volume control to effect in track effect chain if needed 2390 if (chain != 0) { 2391 tracksWithEffect++; 2392 } else { 2393 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2394 name, track->sessionId()); 2395 } 2396 } 2397 2398 2399 int param = AudioMixer::VOLUME; 2400 if (track->mFillingUpStatus == Track::FS_FILLED) { 2401 // no ramp for the first volume setting 2402 track->mFillingUpStatus = Track::FS_ACTIVE; 2403 if (track->mState == TrackBase::RESUMING) { 2404 track->mState = TrackBase::ACTIVE; 2405 param = AudioMixer::RAMP_VOLUME; 2406 } 2407 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2408 } else if (cblk->server != 0) { 2409 // If the track is stopped before the first frame was mixed, 2410 // do not apply ramp 2411 param = AudioMixer::RAMP_VOLUME; 2412 } 2413 2414 // compute volume for this track 2415 uint32_t vl, vr, va; 2416 if (track->isMuted() || track->isPausing() || 2417 mStreamTypes[track->streamType()].mute) { 2418 vl = vr = va = 0; 2419 if (track->isPausing()) { 2420 track->setPaused(); 2421 } 2422 } else { 2423 2424 // read original volumes with volume control 2425 float typeVolume = mStreamTypes[track->streamType()].volume; 2426 float v = masterVolume * typeVolume; 2427 uint32_t vlr = cblk->getVolumeLR(); 2428 vl = vlr & 0xFFFF; 2429 vr = vlr >> 16; 2430 // track volumes come from shared memory, so can't be trusted and must be clamped 2431 if (vl > MAX_GAIN_INT) { 2432 ALOGV("Track left volume out of range: %04X", vl); 2433 vl = MAX_GAIN_INT; 2434 } 2435 if (vr > MAX_GAIN_INT) { 2436 ALOGV("Track right volume out of range: %04X", vr); 2437 vr = MAX_GAIN_INT; 2438 } 2439 // now apply the master volume and stream type volume 2440 vl = (uint32_t)(v * vl) << 12; 2441 vr = (uint32_t)(v * vr) << 12; 2442 // assuming master volume and stream type volume each go up to 1.0, 2443 // vl and vr are now in 8.24 format 2444 2445 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2446 // send level comes from shared memory and so may be corrupt 2447 if (sendLevel > MAX_GAIN_INT) { 2448 ALOGV("Track send level out of range: %04X", sendLevel); 2449 sendLevel = MAX_GAIN_INT; 2450 } 2451 va = (uint32_t)(v * sendLevel); 2452 } 2453 // Delegate volume control to effect in track effect chain if needed 2454 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2455 // Do not ramp volume if volume is controlled by effect 2456 param = AudioMixer::VOLUME; 2457 track->mHasVolumeController = true; 2458 } else { 2459 // force no volume ramp when volume controller was just disabled or removed 2460 // from effect chain to avoid volume spike 2461 if (track->mHasVolumeController) { 2462 param = AudioMixer::VOLUME; 2463 } 2464 track->mHasVolumeController = false; 2465 } 2466 2467 // Convert volumes from 8.24 to 4.12 format 2468 // This additional clamping is needed in case chain->setVolume_l() overshot 2469 vl = (vl + (1 << 11)) >> 12; 2470 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2471 vr = (vr + (1 << 11)) >> 12; 2472 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2473 2474 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2475 2476 // XXX: these things DON'T need to be done each time 2477 mAudioMixer->setBufferProvider(name, track); 2478 mAudioMixer->enable(name); 2479 2480 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2481 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2482 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2483 mAudioMixer->setParameter( 2484 name, 2485 AudioMixer::TRACK, 2486 AudioMixer::FORMAT, (void *)track->format()); 2487 mAudioMixer->setParameter( 2488 name, 2489 AudioMixer::TRACK, 2490 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2491 mAudioMixer->setParameter( 2492 name, 2493 AudioMixer::RESAMPLE, 2494 AudioMixer::SAMPLE_RATE, 2495 (void *)(cblk->sampleRate)); 2496 mAudioMixer->setParameter( 2497 name, 2498 AudioMixer::TRACK, 2499 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2500 mAudioMixer->setParameter( 2501 name, 2502 AudioMixer::TRACK, 2503 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2504 2505 // reset retry count 2506 track->mRetryCount = kMaxTrackRetries; 2507 // If one track is ready, set the mixer ready if: 2508 // - the mixer was not ready during previous round OR 2509 // - no other track is not ready 2510 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2511 mixerStatus != MIXER_TRACKS_ENABLED) { 2512 mixerStatus = MIXER_TRACKS_READY; 2513 } 2514 } else { 2515 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2516 if (track->isStopped()) { 2517 track->reset(); 2518 } 2519 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2520 // We have consumed all the buffers of this track. 2521 // Remove it from the list of active tracks. 2522 tracksToRemove->add(track); 2523 } else { 2524 // No buffers for this track. Give it a few chances to 2525 // fill a buffer, then remove it from active list. 2526 if (--(track->mRetryCount) <= 0) { 2527 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2528 tracksToRemove->add(track); 2529 // indicate to client process that the track was disabled because of underrun 2530 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2531 // If one track is not ready, mark the mixer also not ready if: 2532 // - the mixer was ready during previous round OR 2533 // - no other track is ready 2534 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2535 mixerStatus != MIXER_TRACKS_READY) { 2536 mixerStatus = MIXER_TRACKS_ENABLED; 2537 } 2538 } 2539 mAudioMixer->disable(name); 2540 } 2541 } 2542 2543 // remove all the tracks that need to be... 2544 count = tracksToRemove->size(); 2545 if (CC_UNLIKELY(count)) { 2546 for (size_t i=0 ; i<count ; i++) { 2547 const sp<Track>& track = tracksToRemove->itemAt(i); 2548 mActiveTracks.remove(track); 2549 if (track->mainBuffer() != mMixBuffer) { 2550 chain = getEffectChain_l(track->sessionId()); 2551 if (chain != 0) { 2552 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2553 chain->decActiveTrackCnt(); 2554 } 2555 } 2556 if (track->isTerminated()) { 2557 removeTrack_l(track); 2558 } 2559 } 2560 } 2561 2562 // mix buffer must be cleared if all tracks are connected to an 2563 // effect chain as in this case the mixer will not write to 2564 // mix buffer and track effects will accumulate into it 2565 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2566 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2567 } 2568 2569 return mixerStatus; 2570} 2571 2572/* 2573The derived values that are cached: 2574 - mixBufferSize from frame count * frame size 2575 - activeSleepTime from activeSleepTimeUs() 2576 - idleSleepTime from idleSleepTimeUs() 2577 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2578 - maxPeriod from frame count and sample rate (MIXER only) 2579 2580The parameters that affect these derived values are: 2581 - frame count 2582 - frame size 2583 - sample rate 2584 - device type: A2DP or not 2585 - device latency 2586 - format: PCM or not 2587 - active sleep time 2588 - idle sleep time 2589*/ 2590 2591void AudioFlinger::PlaybackThread::cacheParameters_l() 2592{ 2593 mixBufferSize = mFrameCount * mFrameSize; 2594 activeSleepTime = activeSleepTimeUs(); 2595 idleSleepTime = idleSleepTimeUs(); 2596} 2597 2598void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2599{ 2600 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2601 this, streamType, mTracks.size()); 2602 Mutex::Autolock _l(mLock); 2603 2604 size_t size = mTracks.size(); 2605 for (size_t i = 0; i < size; i++) { 2606 sp<Track> t = mTracks[i]; 2607 if (t->streamType() == streamType) { 2608 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2609 t->mCblk->cv.signal(); 2610 } 2611 } 2612} 2613 2614void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2615{ 2616 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2617 this, streamType, valid); 2618 Mutex::Autolock _l(mLock); 2619 2620 mStreamTypes[streamType].valid = valid; 2621} 2622 2623// getTrackName_l() must be called with ThreadBase::mLock held 2624int AudioFlinger::MixerThread::getTrackName_l() 2625{ 2626 return mAudioMixer->getTrackName(); 2627} 2628 2629// deleteTrackName_l() must be called with ThreadBase::mLock held 2630void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2631{ 2632 ALOGV("remove track (%d) and delete from mixer", name); 2633 mAudioMixer->deleteTrackName(name); 2634} 2635 2636// checkForNewParameters_l() must be called with ThreadBase::mLock held 2637bool AudioFlinger::MixerThread::checkForNewParameters_l() 2638{ 2639 bool reconfig = false; 2640 2641 while (!mNewParameters.isEmpty()) { 2642 status_t status = NO_ERROR; 2643 String8 keyValuePair = mNewParameters[0]; 2644 AudioParameter param = AudioParameter(keyValuePair); 2645 int value; 2646 2647 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2648 reconfig = true; 2649 } 2650 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2651 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2652 status = BAD_VALUE; 2653 } else { 2654 reconfig = true; 2655 } 2656 } 2657 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2658 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2659 status = BAD_VALUE; 2660 } else { 2661 reconfig = true; 2662 } 2663 } 2664 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2665 // do not accept frame count changes if tracks are open as the track buffer 2666 // size depends on frame count and correct behavior would not be guaranteed 2667 // if frame count is changed after track creation 2668 if (!mTracks.isEmpty()) { 2669 status = INVALID_OPERATION; 2670 } else { 2671 reconfig = true; 2672 } 2673 } 2674 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2675 // when changing the audio output device, call addBatteryData to notify 2676 // the change 2677 if ((int)mDevice != value) { 2678 uint32_t params = 0; 2679 // check whether speaker is on 2680 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2681 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2682 } 2683 2684 int deviceWithoutSpeaker 2685 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2686 // check if any other device (except speaker) is on 2687 if (value & deviceWithoutSpeaker ) { 2688 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2689 } 2690 2691 if (params != 0) { 2692 addBatteryData(params); 2693 } 2694 } 2695 2696 // forward device change to effects that have requested to be 2697 // aware of attached audio device. 2698 mDevice = (uint32_t)value; 2699 for (size_t i = 0; i < mEffectChains.size(); i++) { 2700 mEffectChains[i]->setDevice_l(mDevice); 2701 } 2702 } 2703 2704 if (status == NO_ERROR) { 2705 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2706 keyValuePair.string()); 2707 if (!mStandby && status == INVALID_OPERATION) { 2708 mOutput->stream->common.standby(&mOutput->stream->common); 2709 mStandby = true; 2710 mBytesWritten = 0; 2711 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2712 keyValuePair.string()); 2713 } 2714 if (status == NO_ERROR && reconfig) { 2715 delete mAudioMixer; 2716 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2717 mAudioMixer = NULL; 2718 readOutputParameters(); 2719 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2720 for (size_t i = 0; i < mTracks.size() ; i++) { 2721 int name = getTrackName_l(); 2722 if (name < 0) break; 2723 mTracks[i]->mName = name; 2724 // limit track sample rate to 2 x new output sample rate 2725 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2726 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2727 } 2728 } 2729 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2730 } 2731 } 2732 2733 mNewParameters.removeAt(0); 2734 2735 mParamStatus = status; 2736 mParamCond.signal(); 2737 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2738 // already timed out waiting for the status and will never signal the condition. 2739 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2740 } 2741 return reconfig; 2742} 2743 2744status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2745{ 2746 const size_t SIZE = 256; 2747 char buffer[SIZE]; 2748 String8 result; 2749 2750 PlaybackThread::dumpInternals(fd, args); 2751 2752 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2753 result.append(buffer); 2754 write(fd, result.string(), result.size()); 2755 return NO_ERROR; 2756} 2757 2758uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2759{ 2760 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2761} 2762 2763uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2764{ 2765 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2766} 2767 2768void AudioFlinger::MixerThread::cacheParameters_l() 2769{ 2770 PlaybackThread::cacheParameters_l(); 2771 2772 // FIXME: Relaxed timing because of a certain device that can't meet latency 2773 // Should be reduced to 2x after the vendor fixes the driver issue 2774 // increase threshold again due to low power audio mode. The way this warning 2775 // threshold is calculated and its usefulness should be reconsidered anyway. 2776 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2777} 2778 2779// ---------------------------------------------------------------------------- 2780AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2781 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2782 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2783 // mLeftVolFloat, mRightVolFloat 2784 // mLeftVolShort, mRightVolShort 2785{ 2786} 2787 2788AudioFlinger::DirectOutputThread::~DirectOutputThread() 2789{ 2790} 2791 2792AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2793 Vector< sp<Track> > *tracksToRemove 2794) 2795{ 2796 sp<Track> trackToRemove; 2797 2798 mixer_state mixerStatus = MIXER_IDLE; 2799 2800 // find out which tracks need to be processed 2801 if (mActiveTracks.size() != 0) { 2802 sp<Track> t = mActiveTracks[0].promote(); 2803 // The track died recently 2804 if (t == 0) return MIXER_IDLE; 2805 2806 Track* const track = t.get(); 2807 audio_track_cblk_t* cblk = track->cblk(); 2808 2809 // The first time a track is added we wait 2810 // for all its buffers to be filled before processing it 2811 if (cblk->framesReady() && track->isReady() && 2812 !track->isPaused() && !track->isTerminated()) 2813 { 2814 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2815 2816 if (track->mFillingUpStatus == Track::FS_FILLED) { 2817 track->mFillingUpStatus = Track::FS_ACTIVE; 2818 mLeftVolFloat = mRightVolFloat = 0; 2819 mLeftVolShort = mRightVolShort = 0; 2820 if (track->mState == TrackBase::RESUMING) { 2821 track->mState = TrackBase::ACTIVE; 2822 rampVolume = true; 2823 } 2824 } else if (cblk->server != 0) { 2825 // If the track is stopped before the first frame was mixed, 2826 // do not apply ramp 2827 rampVolume = true; 2828 } 2829 // compute volume for this track 2830 float left, right; 2831 if (track->isMuted() || mMasterMute || track->isPausing() || 2832 mStreamTypes[track->streamType()].mute) { 2833 left = right = 0; 2834 if (track->isPausing()) { 2835 track->setPaused(); 2836 } 2837 } else { 2838 float typeVolume = mStreamTypes[track->streamType()].volume; 2839 float v = mMasterVolume * typeVolume; 2840 uint32_t vlr = cblk->getVolumeLR(); 2841 float v_clamped = v * (vlr & 0xFFFF); 2842 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2843 left = v_clamped/MAX_GAIN; 2844 v_clamped = v * (vlr >> 16); 2845 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2846 right = v_clamped/MAX_GAIN; 2847 } 2848 2849 if (left != mLeftVolFloat || right != mRightVolFloat) { 2850 mLeftVolFloat = left; 2851 mRightVolFloat = right; 2852 2853 // If audio HAL implements volume control, 2854 // force software volume to nominal value 2855 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2856 left = 1.0f; 2857 right = 1.0f; 2858 } 2859 2860 // Convert volumes from float to 8.24 2861 uint32_t vl = (uint32_t)(left * (1 << 24)); 2862 uint32_t vr = (uint32_t)(right * (1 << 24)); 2863 2864 // Delegate volume control to effect in track effect chain if needed 2865 // only one effect chain can be present on DirectOutputThread, so if 2866 // there is one, the track is connected to it 2867 if (!mEffectChains.isEmpty()) { 2868 // Do not ramp volume if volume is controlled by effect 2869 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2870 rampVolume = false; 2871 } 2872 } 2873 2874 // Convert volumes from 8.24 to 4.12 format 2875 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2876 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2877 leftVol = (uint16_t)v_clamped; 2878 v_clamped = (vr + (1 << 11)) >> 12; 2879 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2880 rightVol = (uint16_t)v_clamped; 2881 } else { 2882 leftVol = mLeftVolShort; 2883 rightVol = mRightVolShort; 2884 rampVolume = false; 2885 } 2886 2887 // reset retry count 2888 track->mRetryCount = kMaxTrackRetriesDirect; 2889 mActiveTrack = t; 2890 mixerStatus = MIXER_TRACKS_READY; 2891 } else { 2892 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2893 if (track->isStopped()) { 2894 track->reset(); 2895 } 2896 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2897 // We have consumed all the buffers of this track. 2898 // Remove it from the list of active tracks. 2899 trackToRemove = track; 2900 } else { 2901 // No buffers for this track. Give it a few chances to 2902 // fill a buffer, then remove it from active list. 2903 if (--(track->mRetryCount) <= 0) { 2904 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2905 trackToRemove = track; 2906 } else { 2907 mixerStatus = MIXER_TRACKS_ENABLED; 2908 } 2909 } 2910 } 2911 } 2912 2913 // FIXME merge this with similar code for removing multiple tracks 2914 // remove all the tracks that need to be... 2915 if (CC_UNLIKELY(trackToRemove != 0)) { 2916 tracksToRemove->add(trackToRemove); 2917 mActiveTracks.remove(trackToRemove); 2918 if (!mEffectChains.isEmpty()) { 2919 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2920 trackToRemove->sessionId()); 2921 mEffectChains[0]->decActiveTrackCnt(); 2922 } 2923 if (trackToRemove->isTerminated()) { 2924 removeTrack_l(trackToRemove); 2925 } 2926 } 2927 2928 return mixerStatus; 2929} 2930 2931void AudioFlinger::DirectOutputThread::threadLoop_mix() 2932{ 2933 AudioBufferProvider::Buffer buffer; 2934 size_t frameCount = mFrameCount; 2935 int8_t *curBuf = (int8_t *)mMixBuffer; 2936 // output audio to hardware 2937 while (frameCount) { 2938 buffer.frameCount = frameCount; 2939 mActiveTrack->getNextBuffer(&buffer); 2940 if (CC_UNLIKELY(buffer.raw == NULL)) { 2941 memset(curBuf, 0, frameCount * mFrameSize); 2942 break; 2943 } 2944 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2945 frameCount -= buffer.frameCount; 2946 curBuf += buffer.frameCount * mFrameSize; 2947 mActiveTrack->releaseBuffer(&buffer); 2948 } 2949 sleepTime = 0; 2950 standbyTime = systemTime() + standbyDelay; 2951 mActiveTrack.clear(); 2952 2953 // apply volume 2954 2955 // Do not apply volume on compressed audio 2956 if (!audio_is_linear_pcm(mFormat)) { 2957 return; 2958 } 2959 2960 // convert to signed 16 bit before volume calculation 2961 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2962 size_t count = mFrameCount * mChannelCount; 2963 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2964 int16_t *dst = mMixBuffer + count-1; 2965 while (count--) { 2966 *dst-- = (int16_t)(*src--^0x80) << 8; 2967 } 2968 } 2969 2970 frameCount = mFrameCount; 2971 int16_t *out = mMixBuffer; 2972 if (rampVolume) { 2973 if (mChannelCount == 1) { 2974 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2975 int32_t vlInc = d / (int32_t)frameCount; 2976 int32_t vl = ((int32_t)mLeftVolShort << 16); 2977 do { 2978 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2979 out++; 2980 vl += vlInc; 2981 } while (--frameCount); 2982 2983 } else { 2984 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2985 int32_t vlInc = d / (int32_t)frameCount; 2986 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2987 int32_t vrInc = d / (int32_t)frameCount; 2988 int32_t vl = ((int32_t)mLeftVolShort << 16); 2989 int32_t vr = ((int32_t)mRightVolShort << 16); 2990 do { 2991 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2992 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2993 out += 2; 2994 vl += vlInc; 2995 vr += vrInc; 2996 } while (--frameCount); 2997 } 2998 } else { 2999 if (mChannelCount == 1) { 3000 do { 3001 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3002 out++; 3003 } while (--frameCount); 3004 } else { 3005 do { 3006 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3007 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3008 out += 2; 3009 } while (--frameCount); 3010 } 3011 } 3012 3013 // convert back to unsigned 8 bit after volume calculation 3014 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3015 size_t count = mFrameCount * mChannelCount; 3016 int16_t *src = mMixBuffer; 3017 uint8_t *dst = (uint8_t *)mMixBuffer; 3018 while (count--) { 3019 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3020 } 3021 } 3022 3023 mLeftVolShort = leftVol; 3024 mRightVolShort = rightVol; 3025} 3026 3027void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3028{ 3029 if (sleepTime == 0) { 3030 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3031 sleepTime = activeSleepTime; 3032 } else { 3033 sleepTime = idleSleepTime; 3034 } 3035 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3036 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3037 sleepTime = 0; 3038 } 3039} 3040 3041// getTrackName_l() must be called with ThreadBase::mLock held 3042int AudioFlinger::DirectOutputThread::getTrackName_l() 3043{ 3044 return 0; 3045} 3046 3047// deleteTrackName_l() must be called with ThreadBase::mLock held 3048void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3049{ 3050} 3051 3052// checkForNewParameters_l() must be called with ThreadBase::mLock held 3053bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3054{ 3055 bool reconfig = false; 3056 3057 while (!mNewParameters.isEmpty()) { 3058 status_t status = NO_ERROR; 3059 String8 keyValuePair = mNewParameters[0]; 3060 AudioParameter param = AudioParameter(keyValuePair); 3061 int value; 3062 3063 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3064 // do not accept frame count changes if tracks are open as the track buffer 3065 // size depends on frame count and correct behavior would not be garantied 3066 // if frame count is changed after track creation 3067 if (!mTracks.isEmpty()) { 3068 status = INVALID_OPERATION; 3069 } else { 3070 reconfig = true; 3071 } 3072 } 3073 if (status == NO_ERROR) { 3074 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3075 keyValuePair.string()); 3076 if (!mStandby && status == INVALID_OPERATION) { 3077 mOutput->stream->common.standby(&mOutput->stream->common); 3078 mStandby = true; 3079 mBytesWritten = 0; 3080 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3081 keyValuePair.string()); 3082 } 3083 if (status == NO_ERROR && reconfig) { 3084 readOutputParameters(); 3085 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3086 } 3087 } 3088 3089 mNewParameters.removeAt(0); 3090 3091 mParamStatus = status; 3092 mParamCond.signal(); 3093 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3094 // already timed out waiting for the status and will never signal the condition. 3095 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3096 } 3097 return reconfig; 3098} 3099 3100uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3101{ 3102 uint32_t time; 3103 if (audio_is_linear_pcm(mFormat)) { 3104 time = PlaybackThread::activeSleepTimeUs(); 3105 } else { 3106 time = 10000; 3107 } 3108 return time; 3109} 3110 3111uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3112{ 3113 uint32_t time; 3114 if (audio_is_linear_pcm(mFormat)) { 3115 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3116 } else { 3117 time = 10000; 3118 } 3119 return time; 3120} 3121 3122uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3123{ 3124 uint32_t time; 3125 if (audio_is_linear_pcm(mFormat)) { 3126 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3127 } else { 3128 time = 10000; 3129 } 3130 return time; 3131} 3132 3133void AudioFlinger::DirectOutputThread::cacheParameters_l() 3134{ 3135 PlaybackThread::cacheParameters_l(); 3136 3137 // use shorter standby delay as on normal output to release 3138 // hardware resources as soon as possible 3139 standbyDelay = microseconds(activeSleepTime*2); 3140} 3141 3142// ---------------------------------------------------------------------------- 3143 3144AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3145 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3146 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3147 mWaitTimeMs(UINT_MAX) 3148{ 3149 addOutputTrack(mainThread); 3150} 3151 3152AudioFlinger::DuplicatingThread::~DuplicatingThread() 3153{ 3154 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3155 mOutputTracks[i]->destroy(); 3156 } 3157} 3158 3159void AudioFlinger::DuplicatingThread::threadLoop_mix() 3160{ 3161 // mix buffers... 3162 if (outputsReady(outputTracks)) { 3163 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3164 } else { 3165 memset(mMixBuffer, 0, mixBufferSize); 3166 } 3167 sleepTime = 0; 3168 writeFrames = mFrameCount; 3169} 3170 3171void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3172{ 3173 if (sleepTime == 0) { 3174 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3175 sleepTime = activeSleepTime; 3176 } else { 3177 sleepTime = idleSleepTime; 3178 } 3179 } else if (mBytesWritten != 0) { 3180 // flush remaining overflow buffers in output tracks 3181 for (size_t i = 0; i < outputTracks.size(); i++) { 3182 if (outputTracks[i]->isActive()) { 3183 sleepTime = 0; 3184 writeFrames = 0; 3185 memset(mMixBuffer, 0, mixBufferSize); 3186 break; 3187 } 3188 } 3189 } 3190} 3191 3192void AudioFlinger::DuplicatingThread::threadLoop_write() 3193{ 3194 standbyTime = systemTime() + standbyDelay; 3195 for (size_t i = 0; i < outputTracks.size(); i++) { 3196 outputTracks[i]->write(mMixBuffer, writeFrames); 3197 } 3198 mBytesWritten += mixBufferSize; 3199} 3200 3201void AudioFlinger::DuplicatingThread::threadLoop_standby() 3202{ 3203 // DuplicatingThread implements standby by stopping all tracks 3204 for (size_t i = 0; i < outputTracks.size(); i++) { 3205 outputTracks[i]->stop(); 3206 } 3207} 3208 3209void AudioFlinger::DuplicatingThread::saveOutputTracks() 3210{ 3211 outputTracks = mOutputTracks; 3212} 3213 3214void AudioFlinger::DuplicatingThread::clearOutputTracks() 3215{ 3216 outputTracks.clear(); 3217} 3218 3219void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3220{ 3221 Mutex::Autolock _l(mLock); 3222 // FIXME explain this formula 3223 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3224 OutputTrack *outputTrack = new OutputTrack(thread, 3225 this, 3226 mSampleRate, 3227 mFormat, 3228 mChannelMask, 3229 frameCount); 3230 if (outputTrack->cblk() != NULL) { 3231 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3232 mOutputTracks.add(outputTrack); 3233 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3234 updateWaitTime_l(); 3235 } 3236} 3237 3238void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3239{ 3240 Mutex::Autolock _l(mLock); 3241 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3242 if (mOutputTracks[i]->thread() == thread) { 3243 mOutputTracks[i]->destroy(); 3244 mOutputTracks.removeAt(i); 3245 updateWaitTime_l(); 3246 return; 3247 } 3248 } 3249 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3250} 3251 3252// caller must hold mLock 3253void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3254{ 3255 mWaitTimeMs = UINT_MAX; 3256 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3257 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3258 if (strong != 0) { 3259 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3260 if (waitTimeMs < mWaitTimeMs) { 3261 mWaitTimeMs = waitTimeMs; 3262 } 3263 } 3264 } 3265} 3266 3267 3268bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3269{ 3270 for (size_t i = 0; i < outputTracks.size(); i++) { 3271 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3272 if (thread == 0) { 3273 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3274 return false; 3275 } 3276 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3277 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3278 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3279 return false; 3280 } 3281 } 3282 return true; 3283} 3284 3285uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3286{ 3287 return (mWaitTimeMs * 1000) / 2; 3288} 3289 3290void AudioFlinger::DuplicatingThread::cacheParameters_l() 3291{ 3292 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3293 updateWaitTime_l(); 3294 3295 MixerThread::cacheParameters_l(); 3296} 3297 3298// ---------------------------------------------------------------------------- 3299 3300// TrackBase constructor must be called with AudioFlinger::mLock held 3301AudioFlinger::ThreadBase::TrackBase::TrackBase( 3302 ThreadBase *thread, 3303 const sp<Client>& client, 3304 uint32_t sampleRate, 3305 audio_format_t format, 3306 uint32_t channelMask, 3307 int frameCount, 3308 const sp<IMemory>& sharedBuffer, 3309 int sessionId) 3310 : RefBase(), 3311 mThread(thread), 3312 mClient(client), 3313 mCblk(NULL), 3314 // mBuffer 3315 // mBufferEnd 3316 mFrameCount(0), 3317 mState(IDLE), 3318 mFormat(format), 3319 mStepServerFailed(false), 3320 mSessionId(sessionId) 3321 // mChannelCount 3322 // mChannelMask 3323{ 3324 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3325 3326 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3327 size_t size = sizeof(audio_track_cblk_t); 3328 uint8_t channelCount = popcount(channelMask); 3329 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3330 if (sharedBuffer == 0) { 3331 size += bufferSize; 3332 } 3333 3334 if (client != NULL) { 3335 mCblkMemory = client->heap()->allocate(size); 3336 if (mCblkMemory != 0) { 3337 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3338 if (mCblk != NULL) { // construct the shared structure in-place. 3339 new(mCblk) audio_track_cblk_t(); 3340 // clear all buffers 3341 mCblk->frameCount = frameCount; 3342 mCblk->sampleRate = sampleRate; 3343 mChannelCount = channelCount; 3344 mChannelMask = channelMask; 3345 if (sharedBuffer == 0) { 3346 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3347 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3348 // Force underrun condition to avoid false underrun callback until first data is 3349 // written to buffer (other flags are cleared) 3350 mCblk->flags = CBLK_UNDERRUN_ON; 3351 } else { 3352 mBuffer = sharedBuffer->pointer(); 3353 } 3354 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3355 } 3356 } else { 3357 ALOGE("not enough memory for AudioTrack size=%u", size); 3358 client->heap()->dump("AudioTrack"); 3359 return; 3360 } 3361 } else { 3362 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3363 // construct the shared structure in-place. 3364 new(mCblk) audio_track_cblk_t(); 3365 // clear all buffers 3366 mCblk->frameCount = frameCount; 3367 mCblk->sampleRate = sampleRate; 3368 mChannelCount = channelCount; 3369 mChannelMask = channelMask; 3370 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3371 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3372 // Force underrun condition to avoid false underrun callback until first data is 3373 // written to buffer (other flags are cleared) 3374 mCblk->flags = CBLK_UNDERRUN_ON; 3375 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3376 } 3377} 3378 3379AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3380{ 3381 if (mCblk != NULL) { 3382 if (mClient == 0) { 3383 delete mCblk; 3384 } else { 3385 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3386 } 3387 } 3388 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3389 if (mClient != 0) { 3390 // Client destructor must run with AudioFlinger mutex locked 3391 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3392 // If the client's reference count drops to zero, the associated destructor 3393 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3394 // relying on the automatic clear() at end of scope. 3395 mClient.clear(); 3396 } 3397} 3398 3399// AudioBufferProvider interface 3400// getNextBuffer() = 0; 3401// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3402void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3403{ 3404 buffer->raw = NULL; 3405 mFrameCount = buffer->frameCount; 3406 (void) step(); // ignore return value of step() 3407 buffer->frameCount = 0; 3408} 3409 3410bool AudioFlinger::ThreadBase::TrackBase::step() { 3411 bool result; 3412 audio_track_cblk_t* cblk = this->cblk(); 3413 3414 result = cblk->stepServer(mFrameCount); 3415 if (!result) { 3416 ALOGV("stepServer failed acquiring cblk mutex"); 3417 mStepServerFailed = true; 3418 } 3419 return result; 3420} 3421 3422void AudioFlinger::ThreadBase::TrackBase::reset() { 3423 audio_track_cblk_t* cblk = this->cblk(); 3424 3425 cblk->user = 0; 3426 cblk->server = 0; 3427 cblk->userBase = 0; 3428 cblk->serverBase = 0; 3429 mStepServerFailed = false; 3430 ALOGV("TrackBase::reset"); 3431} 3432 3433int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3434 return (int)mCblk->sampleRate; 3435} 3436 3437void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3438 audio_track_cblk_t* cblk = this->cblk(); 3439 size_t frameSize = cblk->frameSize; 3440 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3441 int8_t *bufferEnd = bufferStart + frames * frameSize; 3442 3443 // Check validity of returned pointer in case the track control block would have been corrupted. 3444 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3445 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3446 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3447 server %d, serverBase %d, user %d, userBase %d", 3448 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3449 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3450 return NULL; 3451 } 3452 3453 return bufferStart; 3454} 3455 3456// ---------------------------------------------------------------------------- 3457 3458// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3459AudioFlinger::PlaybackThread::Track::Track( 3460 PlaybackThread *thread, 3461 const sp<Client>& client, 3462 audio_stream_type_t streamType, 3463 uint32_t sampleRate, 3464 audio_format_t format, 3465 uint32_t channelMask, 3466 int frameCount, 3467 const sp<IMemory>& sharedBuffer, 3468 int sessionId) 3469 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3470 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3471 mAuxEffectId(0), mHasVolumeController(false) 3472{ 3473 if (mCblk != NULL) { 3474 if (thread != NULL) { 3475 mName = thread->getTrackName_l(); 3476 mMainBuffer = thread->mixBuffer(); 3477 } 3478 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3479 if (mName < 0) { 3480 ALOGE("no more track names available"); 3481 } 3482 mStreamType = streamType; 3483 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3484 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3485 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3486 } 3487} 3488 3489AudioFlinger::PlaybackThread::Track::~Track() 3490{ 3491 ALOGV("PlaybackThread::Track destructor"); 3492 sp<ThreadBase> thread = mThread.promote(); 3493 if (thread != 0) { 3494 Mutex::Autolock _l(thread->mLock); 3495 mState = TERMINATED; 3496 } 3497} 3498 3499void AudioFlinger::PlaybackThread::Track::destroy() 3500{ 3501 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3502 // by removing it from mTracks vector, so there is a risk that this Tracks's 3503 // destructor is called. As the destructor needs to lock mLock, 3504 // we must acquire a strong reference on this Track before locking mLock 3505 // here so that the destructor is called only when exiting this function. 3506 // On the other hand, as long as Track::destroy() is only called by 3507 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3508 // this Track with its member mTrack. 3509 sp<Track> keep(this); 3510 { // scope for mLock 3511 sp<ThreadBase> thread = mThread.promote(); 3512 if (thread != 0) { 3513 if (!isOutputTrack()) { 3514 if (mState == ACTIVE || mState == RESUMING) { 3515 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3516 3517 // to track the speaker usage 3518 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3519 } 3520 AudioSystem::releaseOutput(thread->id()); 3521 } 3522 Mutex::Autolock _l(thread->mLock); 3523 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3524 playbackThread->destroyTrack_l(this); 3525 } 3526 } 3527} 3528 3529void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3530{ 3531 uint32_t vlr = mCblk->getVolumeLR(); 3532 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", 3533 mName - AudioMixer::TRACK0, 3534 (mClient == 0) ? getpid_cached : mClient->pid(), 3535 mStreamType, 3536 mFormat, 3537 mChannelMask, 3538 mSessionId, 3539 mFrameCount, 3540 mState, 3541 mMute, 3542 mFillingUpStatus, 3543 mCblk->sampleRate, 3544 vlr & 0xFFFF, 3545 vlr >> 16, 3546 mCblk->server, 3547 mCblk->user, 3548 (int)mMainBuffer, 3549 (int)mAuxBuffer); 3550} 3551 3552// AudioBufferProvider interface 3553status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3554 AudioBufferProvider::Buffer* buffer, int64_t pts) 3555{ 3556 audio_track_cblk_t* cblk = this->cblk(); 3557 uint32_t framesReady; 3558 uint32_t framesReq = buffer->frameCount; 3559 3560 // Check if last stepServer failed, try to step now 3561 if (mStepServerFailed) { 3562 if (!step()) goto getNextBuffer_exit; 3563 ALOGV("stepServer recovered"); 3564 mStepServerFailed = false; 3565 } 3566 3567 framesReady = cblk->framesReady(); 3568 3569 if (CC_LIKELY(framesReady)) { 3570 uint32_t s = cblk->server; 3571 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3572 3573 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3574 if (framesReq > framesReady) { 3575 framesReq = framesReady; 3576 } 3577 if (s + framesReq > bufferEnd) { 3578 framesReq = bufferEnd - s; 3579 } 3580 3581 buffer->raw = getBuffer(s, framesReq); 3582 if (buffer->raw == NULL) goto getNextBuffer_exit; 3583 3584 buffer->frameCount = framesReq; 3585 return NO_ERROR; 3586 } 3587 3588getNextBuffer_exit: 3589 buffer->raw = NULL; 3590 buffer->frameCount = 0; 3591 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3592 return NOT_ENOUGH_DATA; 3593} 3594 3595uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3596 return mCblk->framesReady(); 3597} 3598 3599bool AudioFlinger::PlaybackThread::Track::isReady() const { 3600 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3601 3602 if (framesReady() >= mCblk->frameCount || 3603 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3604 mFillingUpStatus = FS_FILLED; 3605 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3606 return true; 3607 } 3608 return false; 3609} 3610 3611status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3612{ 3613 status_t status = NO_ERROR; 3614 ALOGV("start(%d), calling pid %d session %d tid %d", 3615 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3616 sp<ThreadBase> thread = mThread.promote(); 3617 if (thread != 0) { 3618 Mutex::Autolock _l(thread->mLock); 3619 track_state state = mState; 3620 // here the track could be either new, or restarted 3621 // in both cases "unstop" the track 3622 if (mState == PAUSED) { 3623 mState = TrackBase::RESUMING; 3624 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3625 } else { 3626 mState = TrackBase::ACTIVE; 3627 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3628 } 3629 3630 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3631 thread->mLock.unlock(); 3632 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3633 thread->mLock.lock(); 3634 3635 // to track the speaker usage 3636 if (status == NO_ERROR) { 3637 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3638 } 3639 } 3640 if (status == NO_ERROR) { 3641 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3642 playbackThread->addTrack_l(this); 3643 } else { 3644 mState = state; 3645 } 3646 } else { 3647 status = BAD_VALUE; 3648 } 3649 return status; 3650} 3651 3652void AudioFlinger::PlaybackThread::Track::stop() 3653{ 3654 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3655 sp<ThreadBase> thread = mThread.promote(); 3656 if (thread != 0) { 3657 Mutex::Autolock _l(thread->mLock); 3658 track_state state = mState; 3659 if (mState > STOPPED) { 3660 mState = STOPPED; 3661 // If the track is not active (PAUSED and buffers full), flush buffers 3662 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3663 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3664 reset(); 3665 } 3666 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3667 } 3668 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3669 thread->mLock.unlock(); 3670 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3671 thread->mLock.lock(); 3672 3673 // to track the speaker usage 3674 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3675 } 3676 } 3677} 3678 3679void AudioFlinger::PlaybackThread::Track::pause() 3680{ 3681 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3682 sp<ThreadBase> thread = mThread.promote(); 3683 if (thread != 0) { 3684 Mutex::Autolock _l(thread->mLock); 3685 if (mState == ACTIVE || mState == RESUMING) { 3686 mState = PAUSING; 3687 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3688 if (!isOutputTrack()) { 3689 thread->mLock.unlock(); 3690 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3691 thread->mLock.lock(); 3692 3693 // to track the speaker usage 3694 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3695 } 3696 } 3697 } 3698} 3699 3700void AudioFlinger::PlaybackThread::Track::flush() 3701{ 3702 ALOGV("flush(%d)", mName); 3703 sp<ThreadBase> thread = mThread.promote(); 3704 if (thread != 0) { 3705 Mutex::Autolock _l(thread->mLock); 3706 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3707 return; 3708 } 3709 // No point remaining in PAUSED state after a flush => go to 3710 // STOPPED state 3711 mState = STOPPED; 3712 3713 // do not reset the track if it is still in the process of being stopped or paused. 3714 // this will be done by prepareTracks_l() when the track is stopped. 3715 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3716 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3717 reset(); 3718 } 3719 } 3720} 3721 3722void AudioFlinger::PlaybackThread::Track::reset() 3723{ 3724 // Do not reset twice to avoid discarding data written just after a flush and before 3725 // the audioflinger thread detects the track is stopped. 3726 if (!mResetDone) { 3727 TrackBase::reset(); 3728 // Force underrun condition to avoid false underrun callback until first data is 3729 // written to buffer 3730 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3731 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3732 mFillingUpStatus = FS_FILLING; 3733 mResetDone = true; 3734 } 3735} 3736 3737void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3738{ 3739 mMute = muted; 3740} 3741 3742status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3743{ 3744 status_t status = DEAD_OBJECT; 3745 sp<ThreadBase> thread = mThread.promote(); 3746 if (thread != 0) { 3747 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3748 status = playbackThread->attachAuxEffect(this, EffectId); 3749 } 3750 return status; 3751} 3752 3753void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3754{ 3755 mAuxEffectId = EffectId; 3756 mAuxBuffer = buffer; 3757} 3758 3759// timed audio tracks 3760 3761sp<AudioFlinger::PlaybackThread::TimedTrack> 3762AudioFlinger::PlaybackThread::TimedTrack::create( 3763 PlaybackThread *thread, 3764 const sp<Client>& client, 3765 audio_stream_type_t streamType, 3766 uint32_t sampleRate, 3767 audio_format_t format, 3768 uint32_t channelMask, 3769 int frameCount, 3770 const sp<IMemory>& sharedBuffer, 3771 int sessionId) { 3772 if (!client->reserveTimedTrack()) 3773 return NULL; 3774 3775 sp<TimedTrack> track = new TimedTrack( 3776 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3777 sharedBuffer, sessionId); 3778 3779 if (track == NULL) { 3780 client->releaseTimedTrack(); 3781 return NULL; 3782 } 3783 3784 return track; 3785} 3786 3787AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3788 PlaybackThread *thread, 3789 const sp<Client>& client, 3790 audio_stream_type_t streamType, 3791 uint32_t sampleRate, 3792 audio_format_t format, 3793 uint32_t channelMask, 3794 int frameCount, 3795 const sp<IMemory>& sharedBuffer, 3796 int sessionId) 3797 : Track(thread, client, streamType, sampleRate, format, channelMask, 3798 frameCount, sharedBuffer, sessionId), 3799 mTimedSilenceBuffer(NULL), 3800 mTimedSilenceBufferSize(0), 3801 mTimedAudioOutputOnTime(false), 3802 mMediaTimeTransformValid(false) 3803{ 3804 LocalClock lc; 3805 mLocalTimeFreq = lc.getLocalFreq(); 3806 3807 mLocalTimeToSampleTransform.a_zero = 0; 3808 mLocalTimeToSampleTransform.b_zero = 0; 3809 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3810 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3811 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3812 &mLocalTimeToSampleTransform.a_to_b_denom); 3813} 3814 3815AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3816 mClient->releaseTimedTrack(); 3817 delete [] mTimedSilenceBuffer; 3818} 3819 3820status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3821 size_t size, sp<IMemory>* buffer) { 3822 3823 Mutex::Autolock _l(mTimedBufferQueueLock); 3824 3825 trimTimedBufferQueue_l(); 3826 3827 // lazily initialize the shared memory heap for timed buffers 3828 if (mTimedMemoryDealer == NULL) { 3829 const int kTimedBufferHeapSize = 512 << 10; 3830 3831 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3832 "AudioFlingerTimed"); 3833 if (mTimedMemoryDealer == NULL) 3834 return NO_MEMORY; 3835 } 3836 3837 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3838 if (newBuffer == NULL) { 3839 newBuffer = mTimedMemoryDealer->allocate(size); 3840 if (newBuffer == NULL) 3841 return NO_MEMORY; 3842 } 3843 3844 *buffer = newBuffer; 3845 return NO_ERROR; 3846} 3847 3848// caller must hold mTimedBufferQueueLock 3849void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3850 int64_t mediaTimeNow; 3851 { 3852 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3853 if (!mMediaTimeTransformValid) 3854 return; 3855 3856 int64_t targetTimeNow; 3857 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3858 ? mCCHelper.getCommonTime(&targetTimeNow) 3859 : mCCHelper.getLocalTime(&targetTimeNow); 3860 3861 if (OK != res) 3862 return; 3863 3864 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3865 &mediaTimeNow)) { 3866 return; 3867 } 3868 } 3869 3870 size_t trimIndex; 3871 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3872 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3873 break; 3874 } 3875 3876 if (trimIndex) { 3877 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3878 } 3879} 3880 3881status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3882 const sp<IMemory>& buffer, int64_t pts) { 3883 3884 { 3885 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3886 if (!mMediaTimeTransformValid) 3887 return INVALID_OPERATION; 3888 } 3889 3890 Mutex::Autolock _l(mTimedBufferQueueLock); 3891 3892 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3893 3894 return NO_ERROR; 3895} 3896 3897status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3898 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3899 3900 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3901 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3902 target); 3903 3904 if (!(target == TimedAudioTrack::LOCAL_TIME || 3905 target == TimedAudioTrack::COMMON_TIME)) { 3906 return BAD_VALUE; 3907 } 3908 3909 Mutex::Autolock lock(mMediaTimeTransformLock); 3910 mMediaTimeTransform = xform; 3911 mMediaTimeTransformTarget = target; 3912 mMediaTimeTransformValid = true; 3913 3914 return NO_ERROR; 3915} 3916 3917#define min(a, b) ((a) < (b) ? (a) : (b)) 3918 3919// implementation of getNextBuffer for tracks whose buffers have timestamps 3920status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3921 AudioBufferProvider::Buffer* buffer, int64_t pts) 3922{ 3923 if (pts == AudioBufferProvider::kInvalidPTS) { 3924 buffer->raw = 0; 3925 buffer->frameCount = 0; 3926 return INVALID_OPERATION; 3927 } 3928 3929 Mutex::Autolock _l(mTimedBufferQueueLock); 3930 3931 while (true) { 3932 3933 // if we have no timed buffers, then fail 3934 if (mTimedBufferQueue.isEmpty()) { 3935 buffer->raw = 0; 3936 buffer->frameCount = 0; 3937 return NOT_ENOUGH_DATA; 3938 } 3939 3940 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3941 3942 // calculate the PTS of the head of the timed buffer queue expressed in 3943 // local time 3944 int64_t headLocalPTS; 3945 { 3946 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3947 3948 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 3949 3950 if (mMediaTimeTransform.a_to_b_denom == 0) { 3951 // the transform represents a pause, so yield silence 3952 timedYieldSilence(buffer->frameCount, buffer); 3953 return NO_ERROR; 3954 } 3955 3956 int64_t transformedPTS; 3957 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3958 &transformedPTS)) { 3959 // the transform failed. this shouldn't happen, but if it does 3960 // then just drop this buffer 3961 ALOGW("timedGetNextBuffer transform failed"); 3962 buffer->raw = 0; 3963 buffer->frameCount = 0; 3964 mTimedBufferQueue.removeAt(0); 3965 return NO_ERROR; 3966 } 3967 3968 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3969 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3970 &headLocalPTS)) { 3971 buffer->raw = 0; 3972 buffer->frameCount = 0; 3973 return INVALID_OPERATION; 3974 } 3975 } else { 3976 headLocalPTS = transformedPTS; 3977 } 3978 } 3979 3980 // adjust the head buffer's PTS to reflect the portion of the head buffer 3981 // that has already been consumed 3982 int64_t effectivePTS = headLocalPTS + 3983 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3984 3985 // Calculate the delta in samples between the head of the input buffer 3986 // queue and the start of the next output buffer that will be written. 3987 // If the transformation fails because of over or underflow, it means 3988 // that the sample's position in the output stream is so far out of 3989 // whack that it should just be dropped. 3990 int64_t sampleDelta; 3991 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3992 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3993 mTimedBufferQueue.removeAt(0); 3994 continue; 3995 } 3996 if (!mLocalTimeToSampleTransform.doForwardTransform( 3997 (effectivePTS - pts) << 32, &sampleDelta)) { 3998 ALOGV("*** too late during sample rate transform: dropped buffer"); 3999 mTimedBufferQueue.removeAt(0); 4000 continue; 4001 } 4002 4003 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4004 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4005 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4006 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4007 4008 // if the delta between the ideal placement for the next input sample and 4009 // the current output position is within this threshold, then we will 4010 // concatenate the next input samples to the previous output 4011 const int64_t kSampleContinuityThreshold = 4012 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4013 4014 // if this is the first buffer of audio that we're emitting from this track 4015 // then it should be almost exactly on time. 4016 const int64_t kSampleStartupThreshold = 1LL << 32; 4017 4018 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4019 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4020 // the next input is close enough to being on time, so concatenate it 4021 // with the last output 4022 timedYieldSamples(buffer); 4023 4024 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4025 return NO_ERROR; 4026 } else if (sampleDelta > 0) { 4027 // the gap between the current output position and the proper start of 4028 // the next input sample is too big, so fill it with silence 4029 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4030 4031 timedYieldSilence(framesUntilNextInput, buffer); 4032 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4033 return NO_ERROR; 4034 } else { 4035 // the next input sample is late 4036 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4037 size_t onTimeSamplePosition = 4038 head.position() + lateFrames * mCblk->frameSize; 4039 4040 if (onTimeSamplePosition > head.buffer()->size()) { 4041 // all the remaining samples in the head are too late, so 4042 // drop it and move on 4043 ALOGV("*** too late: dropped buffer"); 4044 mTimedBufferQueue.removeAt(0); 4045 continue; 4046 } else { 4047 // skip over the late samples 4048 head.setPosition(onTimeSamplePosition); 4049 4050 // yield the available samples 4051 timedYieldSamples(buffer); 4052 4053 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4054 return NO_ERROR; 4055 } 4056 } 4057 } 4058} 4059 4060// Yield samples from the timed buffer queue head up to the given output 4061// buffer's capacity. 4062// 4063// Caller must hold mTimedBufferQueueLock 4064void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4065 AudioBufferProvider::Buffer* buffer) { 4066 4067 const TimedBuffer& head = mTimedBufferQueue[0]; 4068 4069 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4070 head.position()); 4071 4072 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4073 mCblk->frameSize); 4074 size_t framesRequested = buffer->frameCount; 4075 buffer->frameCount = min(framesLeftInHead, framesRequested); 4076 4077 mTimedAudioOutputOnTime = true; 4078} 4079 4080// Yield samples of silence up to the given output buffer's capacity 4081// 4082// Caller must hold mTimedBufferQueueLock 4083void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4084 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4085 4086 // lazily allocate a buffer filled with silence 4087 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4088 delete [] mTimedSilenceBuffer; 4089 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4090 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4091 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4092 } 4093 4094 buffer->raw = mTimedSilenceBuffer; 4095 size_t framesRequested = buffer->frameCount; 4096 buffer->frameCount = min(numFrames, framesRequested); 4097 4098 mTimedAudioOutputOnTime = false; 4099} 4100 4101// AudioBufferProvider interface 4102void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4103 AudioBufferProvider::Buffer* buffer) { 4104 4105 Mutex::Autolock _l(mTimedBufferQueueLock); 4106 4107 // If the buffer which was just released is part of the buffer at the head 4108 // of the queue, be sure to update the amt of the buffer which has been 4109 // consumed. If the buffer being returned is not part of the head of the 4110 // queue, its either because the buffer is part of the silence buffer, or 4111 // because the head of the timed queue was trimmed after the mixer called 4112 // getNextBuffer but before the mixer called releaseBuffer. 4113 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4114 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4115 4116 void* start = head.buffer()->pointer(); 4117 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4118 4119 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4120 head.setPosition(head.position() + 4121 (buffer->frameCount * mCblk->frameSize)); 4122 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4123 mTimedBufferQueue.removeAt(0); 4124 } 4125 } 4126 } 4127 4128 buffer->raw = 0; 4129 buffer->frameCount = 0; 4130} 4131 4132uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4133 Mutex::Autolock _l(mTimedBufferQueueLock); 4134 4135 uint32_t frames = 0; 4136 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4137 const TimedBuffer& tb = mTimedBufferQueue[i]; 4138 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4139 } 4140 4141 return frames; 4142} 4143 4144AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4145 : mPTS(0), mPosition(0) {} 4146 4147AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4148 const sp<IMemory>& buffer, int64_t pts) 4149 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4150 4151// ---------------------------------------------------------------------------- 4152 4153// RecordTrack constructor must be called with AudioFlinger::mLock held 4154AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4155 RecordThread *thread, 4156 const sp<Client>& client, 4157 uint32_t sampleRate, 4158 audio_format_t format, 4159 uint32_t channelMask, 4160 int frameCount, 4161 int sessionId) 4162 : TrackBase(thread, client, sampleRate, format, 4163 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4164 mOverflow(false) 4165{ 4166 if (mCblk != NULL) { 4167 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4168 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4169 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4170 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4171 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4172 } else { 4173 mCblk->frameSize = sizeof(int8_t); 4174 } 4175 } 4176} 4177 4178AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4179{ 4180 sp<ThreadBase> thread = mThread.promote(); 4181 if (thread != 0) { 4182 AudioSystem::releaseInput(thread->id()); 4183 } 4184} 4185 4186// AudioBufferProvider interface 4187status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4188{ 4189 audio_track_cblk_t* cblk = this->cblk(); 4190 uint32_t framesAvail; 4191 uint32_t framesReq = buffer->frameCount; 4192 4193 // Check if last stepServer failed, try to step now 4194 if (mStepServerFailed) { 4195 if (!step()) goto getNextBuffer_exit; 4196 ALOGV("stepServer recovered"); 4197 mStepServerFailed = false; 4198 } 4199 4200 framesAvail = cblk->framesAvailable_l(); 4201 4202 if (CC_LIKELY(framesAvail)) { 4203 uint32_t s = cblk->server; 4204 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4205 4206 if (framesReq > framesAvail) { 4207 framesReq = framesAvail; 4208 } 4209 if (s + framesReq > bufferEnd) { 4210 framesReq = bufferEnd - s; 4211 } 4212 4213 buffer->raw = getBuffer(s, framesReq); 4214 if (buffer->raw == NULL) goto getNextBuffer_exit; 4215 4216 buffer->frameCount = framesReq; 4217 return NO_ERROR; 4218 } 4219 4220getNextBuffer_exit: 4221 buffer->raw = NULL; 4222 buffer->frameCount = 0; 4223 return NOT_ENOUGH_DATA; 4224} 4225 4226status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4227{ 4228 sp<ThreadBase> thread = mThread.promote(); 4229 if (thread != 0) { 4230 RecordThread *recordThread = (RecordThread *)thread.get(); 4231 return recordThread->start(this, tid); 4232 } else { 4233 return BAD_VALUE; 4234 } 4235} 4236 4237void AudioFlinger::RecordThread::RecordTrack::stop() 4238{ 4239 sp<ThreadBase> thread = mThread.promote(); 4240 if (thread != 0) { 4241 RecordThread *recordThread = (RecordThread *)thread.get(); 4242 recordThread->stop(this); 4243 TrackBase::reset(); 4244 // Force overerrun condition to avoid false overrun callback until first data is 4245 // read from buffer 4246 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4247 } 4248} 4249 4250void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4251{ 4252 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4253 (mClient == 0) ? getpid_cached : mClient->pid(), 4254 mFormat, 4255 mChannelMask, 4256 mSessionId, 4257 mFrameCount, 4258 mState, 4259 mCblk->sampleRate, 4260 mCblk->server, 4261 mCblk->user); 4262} 4263 4264 4265// ---------------------------------------------------------------------------- 4266 4267AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4268 PlaybackThread *playbackThread, 4269 DuplicatingThread *sourceThread, 4270 uint32_t sampleRate, 4271 audio_format_t format, 4272 uint32_t channelMask, 4273 int frameCount) 4274 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4275 mActive(false), mSourceThread(sourceThread) 4276{ 4277 4278 if (mCblk != NULL) { 4279 mCblk->flags |= CBLK_DIRECTION_OUT; 4280 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4281 mOutBuffer.frameCount = 0; 4282 playbackThread->mTracks.add(this); 4283 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4284 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4285 mCblk, mBuffer, mCblk->buffers, 4286 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4287 } else { 4288 ALOGW("Error creating output track on thread %p", playbackThread); 4289 } 4290} 4291 4292AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4293{ 4294 clearBufferQueue(); 4295} 4296 4297status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4298{ 4299 status_t status = Track::start(tid); 4300 if (status != NO_ERROR) { 4301 return status; 4302 } 4303 4304 mActive = true; 4305 mRetryCount = 127; 4306 return status; 4307} 4308 4309void AudioFlinger::PlaybackThread::OutputTrack::stop() 4310{ 4311 Track::stop(); 4312 clearBufferQueue(); 4313 mOutBuffer.frameCount = 0; 4314 mActive = false; 4315} 4316 4317bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4318{ 4319 Buffer *pInBuffer; 4320 Buffer inBuffer; 4321 uint32_t channelCount = mChannelCount; 4322 bool outputBufferFull = false; 4323 inBuffer.frameCount = frames; 4324 inBuffer.i16 = data; 4325 4326 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4327 4328 if (!mActive && frames != 0) { 4329 start(0); 4330 sp<ThreadBase> thread = mThread.promote(); 4331 if (thread != 0) { 4332 MixerThread *mixerThread = (MixerThread *)thread.get(); 4333 if (mCblk->frameCount > frames){ 4334 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4335 uint32_t startFrames = (mCblk->frameCount - frames); 4336 pInBuffer = new Buffer; 4337 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4338 pInBuffer->frameCount = startFrames; 4339 pInBuffer->i16 = pInBuffer->mBuffer; 4340 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4341 mBufferQueue.add(pInBuffer); 4342 } else { 4343 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4344 } 4345 } 4346 } 4347 } 4348 4349 while (waitTimeLeftMs) { 4350 // First write pending buffers, then new data 4351 if (mBufferQueue.size()) { 4352 pInBuffer = mBufferQueue.itemAt(0); 4353 } else { 4354 pInBuffer = &inBuffer; 4355 } 4356 4357 if (pInBuffer->frameCount == 0) { 4358 break; 4359 } 4360 4361 if (mOutBuffer.frameCount == 0) { 4362 mOutBuffer.frameCount = pInBuffer->frameCount; 4363 nsecs_t startTime = systemTime(); 4364 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4365 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4366 outputBufferFull = true; 4367 break; 4368 } 4369 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4370 if (waitTimeLeftMs >= waitTimeMs) { 4371 waitTimeLeftMs -= waitTimeMs; 4372 } else { 4373 waitTimeLeftMs = 0; 4374 } 4375 } 4376 4377 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4378 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4379 mCblk->stepUser(outFrames); 4380 pInBuffer->frameCount -= outFrames; 4381 pInBuffer->i16 += outFrames * channelCount; 4382 mOutBuffer.frameCount -= outFrames; 4383 mOutBuffer.i16 += outFrames * channelCount; 4384 4385 if (pInBuffer->frameCount == 0) { 4386 if (mBufferQueue.size()) { 4387 mBufferQueue.removeAt(0); 4388 delete [] pInBuffer->mBuffer; 4389 delete pInBuffer; 4390 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4391 } else { 4392 break; 4393 } 4394 } 4395 } 4396 4397 // If we could not write all frames, allocate a buffer and queue it for next time. 4398 if (inBuffer.frameCount) { 4399 sp<ThreadBase> thread = mThread.promote(); 4400 if (thread != 0 && !thread->standby()) { 4401 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4402 pInBuffer = new Buffer; 4403 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4404 pInBuffer->frameCount = inBuffer.frameCount; 4405 pInBuffer->i16 = pInBuffer->mBuffer; 4406 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4407 mBufferQueue.add(pInBuffer); 4408 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4409 } else { 4410 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4411 } 4412 } 4413 } 4414 4415 // Calling write() with a 0 length buffer, means that no more data will be written: 4416 // If no more buffers are pending, fill output track buffer to make sure it is started 4417 // by output mixer. 4418 if (frames == 0 && mBufferQueue.size() == 0) { 4419 if (mCblk->user < mCblk->frameCount) { 4420 frames = mCblk->frameCount - mCblk->user; 4421 pInBuffer = new Buffer; 4422 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4423 pInBuffer->frameCount = frames; 4424 pInBuffer->i16 = pInBuffer->mBuffer; 4425 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4426 mBufferQueue.add(pInBuffer); 4427 } else if (mActive) { 4428 stop(); 4429 } 4430 } 4431 4432 return outputBufferFull; 4433} 4434 4435status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4436{ 4437 int active; 4438 status_t result; 4439 audio_track_cblk_t* cblk = mCblk; 4440 uint32_t framesReq = buffer->frameCount; 4441 4442// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4443 buffer->frameCount = 0; 4444 4445 uint32_t framesAvail = cblk->framesAvailable(); 4446 4447 4448 if (framesAvail == 0) { 4449 Mutex::Autolock _l(cblk->lock); 4450 goto start_loop_here; 4451 while (framesAvail == 0) { 4452 active = mActive; 4453 if (CC_UNLIKELY(!active)) { 4454 ALOGV("Not active and NO_MORE_BUFFERS"); 4455 return NO_MORE_BUFFERS; 4456 } 4457 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4458 if (result != NO_ERROR) { 4459 return NO_MORE_BUFFERS; 4460 } 4461 // read the server count again 4462 start_loop_here: 4463 framesAvail = cblk->framesAvailable_l(); 4464 } 4465 } 4466 4467// if (framesAvail < framesReq) { 4468// return NO_MORE_BUFFERS; 4469// } 4470 4471 if (framesReq > framesAvail) { 4472 framesReq = framesAvail; 4473 } 4474 4475 uint32_t u = cblk->user; 4476 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4477 4478 if (u + framesReq > bufferEnd) { 4479 framesReq = bufferEnd - u; 4480 } 4481 4482 buffer->frameCount = framesReq; 4483 buffer->raw = (void *)cblk->buffer(u); 4484 return NO_ERROR; 4485} 4486 4487 4488void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4489{ 4490 size_t size = mBufferQueue.size(); 4491 4492 for (size_t i = 0; i < size; i++) { 4493 Buffer *pBuffer = mBufferQueue.itemAt(i); 4494 delete [] pBuffer->mBuffer; 4495 delete pBuffer; 4496 } 4497 mBufferQueue.clear(); 4498} 4499 4500// ---------------------------------------------------------------------------- 4501 4502AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4503 : RefBase(), 4504 mAudioFlinger(audioFlinger), 4505 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4506 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4507 mPid(pid), 4508 mTimedTrackCount(0) 4509{ 4510 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4511} 4512 4513// Client destructor must be called with AudioFlinger::mLock held 4514AudioFlinger::Client::~Client() 4515{ 4516 mAudioFlinger->removeClient_l(mPid); 4517} 4518 4519sp<MemoryDealer> AudioFlinger::Client::heap() const 4520{ 4521 return mMemoryDealer; 4522} 4523 4524// Reserve one of the limited slots for a timed audio track associated 4525// with this client 4526bool AudioFlinger::Client::reserveTimedTrack() 4527{ 4528 const int kMaxTimedTracksPerClient = 4; 4529 4530 Mutex::Autolock _l(mTimedTrackLock); 4531 4532 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4533 ALOGW("can not create timed track - pid %d has exceeded the limit", 4534 mPid); 4535 return false; 4536 } 4537 4538 mTimedTrackCount++; 4539 return true; 4540} 4541 4542// Release a slot for a timed audio track 4543void AudioFlinger::Client::releaseTimedTrack() 4544{ 4545 Mutex::Autolock _l(mTimedTrackLock); 4546 mTimedTrackCount--; 4547} 4548 4549// ---------------------------------------------------------------------------- 4550 4551AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4552 const sp<IAudioFlingerClient>& client, 4553 pid_t pid) 4554 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4555{ 4556} 4557 4558AudioFlinger::NotificationClient::~NotificationClient() 4559{ 4560} 4561 4562void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4563{ 4564 sp<NotificationClient> keep(this); 4565 mAudioFlinger->removeNotificationClient(mPid); 4566} 4567 4568// ---------------------------------------------------------------------------- 4569 4570AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4571 : BnAudioTrack(), 4572 mTrack(track) 4573{ 4574} 4575 4576AudioFlinger::TrackHandle::~TrackHandle() { 4577 // just stop the track on deletion, associated resources 4578 // will be freed from the main thread once all pending buffers have 4579 // been played. Unless it's not in the active track list, in which 4580 // case we free everything now... 4581 mTrack->destroy(); 4582} 4583 4584sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4585 return mTrack->getCblk(); 4586} 4587 4588status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4589 return mTrack->start(tid); 4590} 4591 4592void AudioFlinger::TrackHandle::stop() { 4593 mTrack->stop(); 4594} 4595 4596void AudioFlinger::TrackHandle::flush() { 4597 mTrack->flush(); 4598} 4599 4600void AudioFlinger::TrackHandle::mute(bool e) { 4601 mTrack->mute(e); 4602} 4603 4604void AudioFlinger::TrackHandle::pause() { 4605 mTrack->pause(); 4606} 4607 4608status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4609{ 4610 return mTrack->attachAuxEffect(EffectId); 4611} 4612 4613status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4614 sp<IMemory>* buffer) { 4615 if (!mTrack->isTimedTrack()) 4616 return INVALID_OPERATION; 4617 4618 PlaybackThread::TimedTrack* tt = 4619 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4620 return tt->allocateTimedBuffer(size, buffer); 4621} 4622 4623status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4624 int64_t pts) { 4625 if (!mTrack->isTimedTrack()) 4626 return INVALID_OPERATION; 4627 4628 PlaybackThread::TimedTrack* tt = 4629 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4630 return tt->queueTimedBuffer(buffer, pts); 4631} 4632 4633status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4634 const LinearTransform& xform, int target) { 4635 4636 if (!mTrack->isTimedTrack()) 4637 return INVALID_OPERATION; 4638 4639 PlaybackThread::TimedTrack* tt = 4640 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4641 return tt->setMediaTimeTransform( 4642 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4643} 4644 4645status_t AudioFlinger::TrackHandle::onTransact( 4646 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4647{ 4648 return BnAudioTrack::onTransact(code, data, reply, flags); 4649} 4650 4651// ---------------------------------------------------------------------------- 4652 4653sp<IAudioRecord> AudioFlinger::openRecord( 4654 pid_t pid, 4655 audio_io_handle_t input, 4656 uint32_t sampleRate, 4657 audio_format_t format, 4658 uint32_t channelMask, 4659 int frameCount, 4660 // FIXME dead, remove from IAudioFlinger 4661 uint32_t flags, 4662 int *sessionId, 4663 status_t *status) 4664{ 4665 sp<RecordThread::RecordTrack> recordTrack; 4666 sp<RecordHandle> recordHandle; 4667 sp<Client> client; 4668 status_t lStatus; 4669 RecordThread *thread; 4670 size_t inFrameCount; 4671 int lSessionId; 4672 4673 // check calling permissions 4674 if (!recordingAllowed()) { 4675 lStatus = PERMISSION_DENIED; 4676 goto Exit; 4677 } 4678 4679 // add client to list 4680 { // scope for mLock 4681 Mutex::Autolock _l(mLock); 4682 thread = checkRecordThread_l(input); 4683 if (thread == NULL) { 4684 lStatus = BAD_VALUE; 4685 goto Exit; 4686 } 4687 4688 client = registerPid_l(pid); 4689 4690 // If no audio session id is provided, create one here 4691 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4692 lSessionId = *sessionId; 4693 } else { 4694 lSessionId = nextUniqueId(); 4695 if (sessionId != NULL) { 4696 *sessionId = lSessionId; 4697 } 4698 } 4699 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4700 recordTrack = thread->createRecordTrack_l(client, 4701 sampleRate, 4702 format, 4703 channelMask, 4704 frameCount, 4705 lSessionId, 4706 &lStatus); 4707 } 4708 if (lStatus != NO_ERROR) { 4709 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4710 // destructor is called by the TrackBase destructor with mLock held 4711 client.clear(); 4712 recordTrack.clear(); 4713 goto Exit; 4714 } 4715 4716 // return to handle to client 4717 recordHandle = new RecordHandle(recordTrack); 4718 lStatus = NO_ERROR; 4719 4720Exit: 4721 if (status) { 4722 *status = lStatus; 4723 } 4724 return recordHandle; 4725} 4726 4727// ---------------------------------------------------------------------------- 4728 4729AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4730 : BnAudioRecord(), 4731 mRecordTrack(recordTrack) 4732{ 4733} 4734 4735AudioFlinger::RecordHandle::~RecordHandle() { 4736 stop(); 4737} 4738 4739sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4740 return mRecordTrack->getCblk(); 4741} 4742 4743status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4744 ALOGV("RecordHandle::start()"); 4745 return mRecordTrack->start(tid); 4746} 4747 4748void AudioFlinger::RecordHandle::stop() { 4749 ALOGV("RecordHandle::stop()"); 4750 mRecordTrack->stop(); 4751} 4752 4753status_t AudioFlinger::RecordHandle::onTransact( 4754 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4755{ 4756 return BnAudioRecord::onTransact(code, data, reply, flags); 4757} 4758 4759// ---------------------------------------------------------------------------- 4760 4761AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4762 AudioStreamIn *input, 4763 uint32_t sampleRate, 4764 uint32_t channels, 4765 audio_io_handle_t id, 4766 uint32_t device) : 4767 ThreadBase(audioFlinger, id, device, RECORD), 4768 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4769 // mRsmpInIndex and mInputBytes set by readInputParameters() 4770 mReqChannelCount(popcount(channels)), 4771 mReqSampleRate(sampleRate) 4772 // mBytesRead is only meaningful while active, and so is cleared in start() 4773 // (but might be better to also clear here for dump?) 4774{ 4775 snprintf(mName, kNameLength, "AudioIn_%X", id); 4776 4777 readInputParameters(); 4778} 4779 4780 4781AudioFlinger::RecordThread::~RecordThread() 4782{ 4783 delete[] mRsmpInBuffer; 4784 delete mResampler; 4785 delete[] mRsmpOutBuffer; 4786} 4787 4788void AudioFlinger::RecordThread::onFirstRef() 4789{ 4790 run(mName, PRIORITY_URGENT_AUDIO); 4791} 4792 4793status_t AudioFlinger::RecordThread::readyToRun() 4794{ 4795 status_t status = initCheck(); 4796 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4797 return status; 4798} 4799 4800bool AudioFlinger::RecordThread::threadLoop() 4801{ 4802 AudioBufferProvider::Buffer buffer; 4803 sp<RecordTrack> activeTrack; 4804 Vector< sp<EffectChain> > effectChains; 4805 4806 nsecs_t lastWarning = 0; 4807 4808 acquireWakeLock(); 4809 4810 // start recording 4811 while (!exitPending()) { 4812 4813 processConfigEvents(); 4814 4815 { // scope for mLock 4816 Mutex::Autolock _l(mLock); 4817 checkForNewParameters_l(); 4818 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4819 if (!mStandby) { 4820 mInput->stream->common.standby(&mInput->stream->common); 4821 mStandby = true; 4822 } 4823 4824 if (exitPending()) break; 4825 4826 releaseWakeLock_l(); 4827 ALOGV("RecordThread: loop stopping"); 4828 // go to sleep 4829 mWaitWorkCV.wait(mLock); 4830 ALOGV("RecordThread: loop starting"); 4831 acquireWakeLock_l(); 4832 continue; 4833 } 4834 if (mActiveTrack != 0) { 4835 if (mActiveTrack->mState == TrackBase::PAUSING) { 4836 if (!mStandby) { 4837 mInput->stream->common.standby(&mInput->stream->common); 4838 mStandby = true; 4839 } 4840 mActiveTrack.clear(); 4841 mStartStopCond.broadcast(); 4842 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4843 if (mReqChannelCount != mActiveTrack->channelCount()) { 4844 mActiveTrack.clear(); 4845 mStartStopCond.broadcast(); 4846 } else if (mBytesRead != 0) { 4847 // record start succeeds only if first read from audio input 4848 // succeeds 4849 if (mBytesRead > 0) { 4850 mActiveTrack->mState = TrackBase::ACTIVE; 4851 } else { 4852 mActiveTrack.clear(); 4853 } 4854 mStartStopCond.broadcast(); 4855 } 4856 mStandby = false; 4857 } 4858 } 4859 lockEffectChains_l(effectChains); 4860 } 4861 4862 if (mActiveTrack != 0) { 4863 if (mActiveTrack->mState != TrackBase::ACTIVE && 4864 mActiveTrack->mState != TrackBase::RESUMING) { 4865 unlockEffectChains(effectChains); 4866 usleep(kRecordThreadSleepUs); 4867 continue; 4868 } 4869 for (size_t i = 0; i < effectChains.size(); i ++) { 4870 effectChains[i]->process_l(); 4871 } 4872 4873 buffer.frameCount = mFrameCount; 4874 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4875 size_t framesOut = buffer.frameCount; 4876 if (mResampler == NULL) { 4877 // no resampling 4878 while (framesOut) { 4879 size_t framesIn = mFrameCount - mRsmpInIndex; 4880 if (framesIn) { 4881 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4882 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4883 if (framesIn > framesOut) 4884 framesIn = framesOut; 4885 mRsmpInIndex += framesIn; 4886 framesOut -= framesIn; 4887 if ((int)mChannelCount == mReqChannelCount || 4888 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4889 memcpy(dst, src, framesIn * mFrameSize); 4890 } else { 4891 int16_t *src16 = (int16_t *)src; 4892 int16_t *dst16 = (int16_t *)dst; 4893 if (mChannelCount == 1) { 4894 while (framesIn--) { 4895 *dst16++ = *src16; 4896 *dst16++ = *src16++; 4897 } 4898 } else { 4899 while (framesIn--) { 4900 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4901 src16 += 2; 4902 } 4903 } 4904 } 4905 } 4906 if (framesOut && mFrameCount == mRsmpInIndex) { 4907 if (framesOut == mFrameCount && 4908 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4909 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4910 framesOut = 0; 4911 } else { 4912 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4913 mRsmpInIndex = 0; 4914 } 4915 if (mBytesRead < 0) { 4916 ALOGE("Error reading audio input"); 4917 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4918 // Force input into standby so that it tries to 4919 // recover at next read attempt 4920 mInput->stream->common.standby(&mInput->stream->common); 4921 usleep(kRecordThreadSleepUs); 4922 } 4923 mRsmpInIndex = mFrameCount; 4924 framesOut = 0; 4925 buffer.frameCount = 0; 4926 } 4927 } 4928 } 4929 } else { 4930 // resampling 4931 4932 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4933 // alter output frame count as if we were expecting stereo samples 4934 if (mChannelCount == 1 && mReqChannelCount == 1) { 4935 framesOut >>= 1; 4936 } 4937 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4938 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4939 // are 32 bit aligned which should be always true. 4940 if (mChannelCount == 2 && mReqChannelCount == 1) { 4941 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4942 // the resampler always outputs stereo samples: do post stereo to mono conversion 4943 int16_t *src = (int16_t *)mRsmpOutBuffer; 4944 int16_t *dst = buffer.i16; 4945 while (framesOut--) { 4946 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4947 src += 2; 4948 } 4949 } else { 4950 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4951 } 4952 4953 } 4954 mActiveTrack->releaseBuffer(&buffer); 4955 mActiveTrack->overflow(); 4956 } 4957 // client isn't retrieving buffers fast enough 4958 else { 4959 if (!mActiveTrack->setOverflow()) { 4960 nsecs_t now = systemTime(); 4961 if ((now - lastWarning) > kWarningThrottleNs) { 4962 ALOGW("RecordThread: buffer overflow"); 4963 lastWarning = now; 4964 } 4965 } 4966 // Release the processor for a while before asking for a new buffer. 4967 // This will give the application more chance to read from the buffer and 4968 // clear the overflow. 4969 usleep(kRecordThreadSleepUs); 4970 } 4971 } 4972 // enable changes in effect chain 4973 unlockEffectChains(effectChains); 4974 effectChains.clear(); 4975 } 4976 4977 if (!mStandby) { 4978 mInput->stream->common.standby(&mInput->stream->common); 4979 } 4980 mActiveTrack.clear(); 4981 4982 mStartStopCond.broadcast(); 4983 4984 releaseWakeLock(); 4985 4986 ALOGV("RecordThread %p exiting", this); 4987 return false; 4988} 4989 4990 4991sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4992 const sp<AudioFlinger::Client>& client, 4993 uint32_t sampleRate, 4994 audio_format_t format, 4995 int channelMask, 4996 int frameCount, 4997 int sessionId, 4998 status_t *status) 4999{ 5000 sp<RecordTrack> track; 5001 status_t lStatus; 5002 5003 lStatus = initCheck(); 5004 if (lStatus != NO_ERROR) { 5005 ALOGE("Audio driver not initialized."); 5006 goto Exit; 5007 } 5008 5009 { // scope for mLock 5010 Mutex::Autolock _l(mLock); 5011 5012 track = new RecordTrack(this, client, sampleRate, 5013 format, channelMask, frameCount, sessionId); 5014 5015 if (track->getCblk() == 0) { 5016 lStatus = NO_MEMORY; 5017 goto Exit; 5018 } 5019 5020 mTrack = track.get(); 5021 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5022 bool suspend = audio_is_bluetooth_sco_device( 5023 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5024 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5025 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5026 } 5027 lStatus = NO_ERROR; 5028 5029Exit: 5030 if (status) { 5031 *status = lStatus; 5032 } 5033 return track; 5034} 5035 5036status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5037{ 5038 ALOGV("RecordThread::start tid=%d", tid); 5039 sp<ThreadBase> strongMe = this; 5040 status_t status = NO_ERROR; 5041 { 5042 AutoMutex lock(mLock); 5043 if (mActiveTrack != 0) { 5044 if (recordTrack != mActiveTrack.get()) { 5045 status = -EBUSY; 5046 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5047 mActiveTrack->mState = TrackBase::ACTIVE; 5048 } 5049 return status; 5050 } 5051 5052 recordTrack->mState = TrackBase::IDLE; 5053 mActiveTrack = recordTrack; 5054 mLock.unlock(); 5055 status_t status = AudioSystem::startInput(mId); 5056 mLock.lock(); 5057 if (status != NO_ERROR) { 5058 mActiveTrack.clear(); 5059 return status; 5060 } 5061 mRsmpInIndex = mFrameCount; 5062 mBytesRead = 0; 5063 if (mResampler != NULL) { 5064 mResampler->reset(); 5065 } 5066 mActiveTrack->mState = TrackBase::RESUMING; 5067 // signal thread to start 5068 ALOGV("Signal record thread"); 5069 mWaitWorkCV.signal(); 5070 // do not wait for mStartStopCond if exiting 5071 if (exitPending()) { 5072 mActiveTrack.clear(); 5073 status = INVALID_OPERATION; 5074 goto startError; 5075 } 5076 mStartStopCond.wait(mLock); 5077 if (mActiveTrack == 0) { 5078 ALOGV("Record failed to start"); 5079 status = BAD_VALUE; 5080 goto startError; 5081 } 5082 ALOGV("Record started OK"); 5083 return status; 5084 } 5085startError: 5086 AudioSystem::stopInput(mId); 5087 return status; 5088} 5089 5090void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5091 ALOGV("RecordThread::stop"); 5092 sp<ThreadBase> strongMe = this; 5093 { 5094 AutoMutex lock(mLock); 5095 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5096 mActiveTrack->mState = TrackBase::PAUSING; 5097 // do not wait for mStartStopCond if exiting 5098 if (exitPending()) { 5099 return; 5100 } 5101 mStartStopCond.wait(mLock); 5102 // if we have been restarted, recordTrack == mActiveTrack.get() here 5103 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5104 mLock.unlock(); 5105 AudioSystem::stopInput(mId); 5106 mLock.lock(); 5107 ALOGV("Record stopped OK"); 5108 } 5109 } 5110 } 5111} 5112 5113status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5114{ 5115 const size_t SIZE = 256; 5116 char buffer[SIZE]; 5117 String8 result; 5118 5119 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5120 result.append(buffer); 5121 5122 if (mActiveTrack != 0) { 5123 result.append("Active Track:\n"); 5124 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5125 mActiveTrack->dump(buffer, SIZE); 5126 result.append(buffer); 5127 5128 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5129 result.append(buffer); 5130 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5131 result.append(buffer); 5132 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5133 result.append(buffer); 5134 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5135 result.append(buffer); 5136 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5137 result.append(buffer); 5138 5139 5140 } else { 5141 result.append("No record client\n"); 5142 } 5143 write(fd, result.string(), result.size()); 5144 5145 dumpBase(fd, args); 5146 dumpEffectChains(fd, args); 5147 5148 return NO_ERROR; 5149} 5150 5151// AudioBufferProvider interface 5152status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5153{ 5154 size_t framesReq = buffer->frameCount; 5155 size_t framesReady = mFrameCount - mRsmpInIndex; 5156 int channelCount; 5157 5158 if (framesReady == 0) { 5159 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5160 if (mBytesRead < 0) { 5161 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5162 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5163 // Force input into standby so that it tries to 5164 // recover at next read attempt 5165 mInput->stream->common.standby(&mInput->stream->common); 5166 usleep(kRecordThreadSleepUs); 5167 } 5168 buffer->raw = NULL; 5169 buffer->frameCount = 0; 5170 return NOT_ENOUGH_DATA; 5171 } 5172 mRsmpInIndex = 0; 5173 framesReady = mFrameCount; 5174 } 5175 5176 if (framesReq > framesReady) { 5177 framesReq = framesReady; 5178 } 5179 5180 if (mChannelCount == 1 && mReqChannelCount == 2) { 5181 channelCount = 1; 5182 } else { 5183 channelCount = 2; 5184 } 5185 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5186 buffer->frameCount = framesReq; 5187 return NO_ERROR; 5188} 5189 5190// AudioBufferProvider interface 5191void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5192{ 5193 mRsmpInIndex += buffer->frameCount; 5194 buffer->frameCount = 0; 5195} 5196 5197bool AudioFlinger::RecordThread::checkForNewParameters_l() 5198{ 5199 bool reconfig = false; 5200 5201 while (!mNewParameters.isEmpty()) { 5202 status_t status = NO_ERROR; 5203 String8 keyValuePair = mNewParameters[0]; 5204 AudioParameter param = AudioParameter(keyValuePair); 5205 int value; 5206 audio_format_t reqFormat = mFormat; 5207 int reqSamplingRate = mReqSampleRate; 5208 int reqChannelCount = mReqChannelCount; 5209 5210 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5211 reqSamplingRate = value; 5212 reconfig = true; 5213 } 5214 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5215 reqFormat = (audio_format_t) value; 5216 reconfig = true; 5217 } 5218 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5219 reqChannelCount = popcount(value); 5220 reconfig = true; 5221 } 5222 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5223 // do not accept frame count changes if tracks are open as the track buffer 5224 // size depends on frame count and correct behavior would not be guaranteed 5225 // if frame count is changed after track creation 5226 if (mActiveTrack != 0) { 5227 status = INVALID_OPERATION; 5228 } else { 5229 reconfig = true; 5230 } 5231 } 5232 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5233 // forward device change to effects that have requested to be 5234 // aware of attached audio device. 5235 for (size_t i = 0; i < mEffectChains.size(); i++) { 5236 mEffectChains[i]->setDevice_l(value); 5237 } 5238 // store input device and output device but do not forward output device to audio HAL. 5239 // Note that status is ignored by the caller for output device 5240 // (see AudioFlinger::setParameters() 5241 if (value & AUDIO_DEVICE_OUT_ALL) { 5242 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5243 status = BAD_VALUE; 5244 } else { 5245 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5246 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5247 if (mTrack != NULL) { 5248 bool suspend = audio_is_bluetooth_sco_device( 5249 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5250 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5251 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5252 } 5253 } 5254 mDevice |= (uint32_t)value; 5255 } 5256 if (status == NO_ERROR) { 5257 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5258 if (status == INVALID_OPERATION) { 5259 mInput->stream->common.standby(&mInput->stream->common); 5260 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5261 keyValuePair.string()); 5262 } 5263 if (reconfig) { 5264 if (status == BAD_VALUE && 5265 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5266 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5267 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5268 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5269 (reqChannelCount <= FCC_2)) { 5270 status = NO_ERROR; 5271 } 5272 if (status == NO_ERROR) { 5273 readInputParameters(); 5274 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5275 } 5276 } 5277 } 5278 5279 mNewParameters.removeAt(0); 5280 5281 mParamStatus = status; 5282 mParamCond.signal(); 5283 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5284 // already timed out waiting for the status and will never signal the condition. 5285 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5286 } 5287 return reconfig; 5288} 5289 5290String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5291{ 5292 char *s; 5293 String8 out_s8 = String8(); 5294 5295 Mutex::Autolock _l(mLock); 5296 if (initCheck() != NO_ERROR) { 5297 return out_s8; 5298 } 5299 5300 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5301 out_s8 = String8(s); 5302 free(s); 5303 return out_s8; 5304} 5305 5306void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5307 AudioSystem::OutputDescriptor desc; 5308 void *param2 = NULL; 5309 5310 switch (event) { 5311 case AudioSystem::INPUT_OPENED: 5312 case AudioSystem::INPUT_CONFIG_CHANGED: 5313 desc.channels = mChannelMask; 5314 desc.samplingRate = mSampleRate; 5315 desc.format = mFormat; 5316 desc.frameCount = mFrameCount; 5317 desc.latency = 0; 5318 param2 = &desc; 5319 break; 5320 5321 case AudioSystem::INPUT_CLOSED: 5322 default: 5323 break; 5324 } 5325 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5326} 5327 5328void AudioFlinger::RecordThread::readInputParameters() 5329{ 5330 delete mRsmpInBuffer; 5331 // mRsmpInBuffer is always assigned a new[] below 5332 delete mRsmpOutBuffer; 5333 mRsmpOutBuffer = NULL; 5334 delete mResampler; 5335 mResampler = NULL; 5336 5337 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5338 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5339 mChannelCount = (uint16_t)popcount(mChannelMask); 5340 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5341 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5342 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5343 mFrameCount = mInputBytes / mFrameSize; 5344 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5345 5346 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5347 { 5348 int channelCount; 5349 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5350 // stereo to mono post process as the resampler always outputs stereo. 5351 if (mChannelCount == 1 && mReqChannelCount == 2) { 5352 channelCount = 1; 5353 } else { 5354 channelCount = 2; 5355 } 5356 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5357 mResampler->setSampleRate(mSampleRate); 5358 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5359 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5360 5361 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5362 if (mChannelCount == 1 && mReqChannelCount == 1) { 5363 mFrameCount >>= 1; 5364 } 5365 5366 } 5367 mRsmpInIndex = mFrameCount; 5368} 5369 5370unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5371{ 5372 Mutex::Autolock _l(mLock); 5373 if (initCheck() != NO_ERROR) { 5374 return 0; 5375 } 5376 5377 return mInput->stream->get_input_frames_lost(mInput->stream); 5378} 5379 5380uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5381{ 5382 Mutex::Autolock _l(mLock); 5383 uint32_t result = 0; 5384 if (getEffectChain_l(sessionId) != 0) { 5385 result = EFFECT_SESSION; 5386 } 5387 5388 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5389 result |= TRACK_SESSION; 5390 } 5391 5392 return result; 5393} 5394 5395AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5396{ 5397 Mutex::Autolock _l(mLock); 5398 return mTrack; 5399} 5400 5401AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5402{ 5403 Mutex::Autolock _l(mLock); 5404 return mInput; 5405} 5406 5407AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5408{ 5409 Mutex::Autolock _l(mLock); 5410 AudioStreamIn *input = mInput; 5411 mInput = NULL; 5412 return input; 5413} 5414 5415// this method must always be called either with ThreadBase mLock held or inside the thread loop 5416audio_stream_t* AudioFlinger::RecordThread::stream() 5417{ 5418 if (mInput == NULL) { 5419 return NULL; 5420 } 5421 return &mInput->stream->common; 5422} 5423 5424 5425// ---------------------------------------------------------------------------- 5426 5427audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5428 uint32_t *pSamplingRate, 5429 audio_format_t *pFormat, 5430 uint32_t *pChannels, 5431 uint32_t *pLatencyMs, 5432 audio_policy_output_flags_t flags) 5433{ 5434 status_t status; 5435 PlaybackThread *thread = NULL; 5436 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5437 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5438 uint32_t channels = pChannels ? *pChannels : 0; 5439 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5440 audio_stream_out_t *outStream; 5441 audio_hw_device_t *outHwDev; 5442 5443 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5444 pDevices ? *pDevices : 0, 5445 samplingRate, 5446 format, 5447 channels, 5448 flags); 5449 5450 if (pDevices == NULL || *pDevices == 0) { 5451 return 0; 5452 } 5453 5454 Mutex::Autolock _l(mLock); 5455 5456 outHwDev = findSuitableHwDev_l(*pDevices); 5457 if (outHwDev == NULL) 5458 return 0; 5459 5460 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5461 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5462 &channels, &samplingRate, &outStream); 5463 mHardwareStatus = AUDIO_HW_IDLE; 5464 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5465 outStream, 5466 samplingRate, 5467 format, 5468 channels, 5469 status); 5470 5471 if (outStream != NULL) { 5472 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5473 audio_io_handle_t id = nextUniqueId(); 5474 5475 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5476 (format != AUDIO_FORMAT_PCM_16_BIT) || 5477 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5478 thread = new DirectOutputThread(this, output, id, *pDevices); 5479 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5480 } else { 5481 thread = new MixerThread(this, output, id, *pDevices); 5482 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5483 } 5484 mPlaybackThreads.add(id, thread); 5485 5486 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5487 if (pFormat != NULL) *pFormat = format; 5488 if (pChannels != NULL) *pChannels = channels; 5489 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5490 5491 // notify client processes of the new output creation 5492 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5493 return id; 5494 } 5495 5496 return 0; 5497} 5498 5499audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5500 audio_io_handle_t output2) 5501{ 5502 Mutex::Autolock _l(mLock); 5503 MixerThread *thread1 = checkMixerThread_l(output1); 5504 MixerThread *thread2 = checkMixerThread_l(output2); 5505 5506 if (thread1 == NULL || thread2 == NULL) { 5507 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5508 return 0; 5509 } 5510 5511 audio_io_handle_t id = nextUniqueId(); 5512 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5513 thread->addOutputTrack(thread2); 5514 mPlaybackThreads.add(id, thread); 5515 // notify client processes of the new output creation 5516 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5517 return id; 5518} 5519 5520status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5521{ 5522 // keep strong reference on the playback thread so that 5523 // it is not destroyed while exit() is executed 5524 sp<PlaybackThread> thread; 5525 { 5526 Mutex::Autolock _l(mLock); 5527 thread = checkPlaybackThread_l(output); 5528 if (thread == NULL) { 5529 return BAD_VALUE; 5530 } 5531 5532 ALOGV("closeOutput() %d", output); 5533 5534 if (thread->type() == ThreadBase::MIXER) { 5535 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5536 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5537 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5538 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5539 } 5540 } 5541 } 5542 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5543 mPlaybackThreads.removeItem(output); 5544 } 5545 thread->exit(); 5546 // The thread entity (active unit of execution) is no longer running here, 5547 // but the ThreadBase container still exists. 5548 5549 if (thread->type() != ThreadBase::DUPLICATING) { 5550 AudioStreamOut *out = thread->clearOutput(); 5551 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5552 // from now on thread->mOutput is NULL 5553 out->hwDev->close_output_stream(out->hwDev, out->stream); 5554 delete out; 5555 } 5556 return NO_ERROR; 5557} 5558 5559status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5560{ 5561 Mutex::Autolock _l(mLock); 5562 PlaybackThread *thread = checkPlaybackThread_l(output); 5563 5564 if (thread == NULL) { 5565 return BAD_VALUE; 5566 } 5567 5568 ALOGV("suspendOutput() %d", output); 5569 thread->suspend(); 5570 5571 return NO_ERROR; 5572} 5573 5574status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5575{ 5576 Mutex::Autolock _l(mLock); 5577 PlaybackThread *thread = checkPlaybackThread_l(output); 5578 5579 if (thread == NULL) { 5580 return BAD_VALUE; 5581 } 5582 5583 ALOGV("restoreOutput() %d", output); 5584 5585 thread->restore(); 5586 5587 return NO_ERROR; 5588} 5589 5590audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5591 uint32_t *pSamplingRate, 5592 audio_format_t *pFormat, 5593 uint32_t *pChannels, 5594 audio_in_acoustics_t acoustics) 5595{ 5596 status_t status; 5597 RecordThread *thread = NULL; 5598 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5599 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5600 uint32_t channels = pChannels ? *pChannels : 0; 5601 uint32_t reqSamplingRate = samplingRate; 5602 audio_format_t reqFormat = format; 5603 uint32_t reqChannels = channels; 5604 audio_stream_in_t *inStream; 5605 audio_hw_device_t *inHwDev; 5606 5607 if (pDevices == NULL || *pDevices == 0) { 5608 return 0; 5609 } 5610 5611 Mutex::Autolock _l(mLock); 5612 5613 inHwDev = findSuitableHwDev_l(*pDevices); 5614 if (inHwDev == NULL) 5615 return 0; 5616 5617 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5618 &channels, &samplingRate, 5619 acoustics, 5620 &inStream); 5621 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5622 inStream, 5623 samplingRate, 5624 format, 5625 channels, 5626 acoustics, 5627 status); 5628 5629 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5630 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5631 // or stereo to mono conversions on 16 bit PCM inputs. 5632 if (inStream == NULL && status == BAD_VALUE && 5633 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5634 (samplingRate <= 2 * reqSamplingRate) && 5635 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5636 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5637 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5638 &channels, &samplingRate, 5639 acoustics, 5640 &inStream); 5641 } 5642 5643 if (inStream != NULL) { 5644 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5645 5646 audio_io_handle_t id = nextUniqueId(); 5647 // Start record thread 5648 // RecorThread require both input and output device indication to forward to audio 5649 // pre processing modules 5650 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5651 thread = new RecordThread(this, 5652 input, 5653 reqSamplingRate, 5654 reqChannels, 5655 id, 5656 device); 5657 mRecordThreads.add(id, thread); 5658 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5659 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5660 if (pFormat != NULL) *pFormat = format; 5661 if (pChannels != NULL) *pChannels = reqChannels; 5662 5663 input->stream->common.standby(&input->stream->common); 5664 5665 // notify client processes of the new input creation 5666 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5667 return id; 5668 } 5669 5670 return 0; 5671} 5672 5673status_t AudioFlinger::closeInput(audio_io_handle_t input) 5674{ 5675 // keep strong reference on the record thread so that 5676 // it is not destroyed while exit() is executed 5677 sp<RecordThread> thread; 5678 { 5679 Mutex::Autolock _l(mLock); 5680 thread = checkRecordThread_l(input); 5681 if (thread == NULL) { 5682 return BAD_VALUE; 5683 } 5684 5685 ALOGV("closeInput() %d", input); 5686 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5687 mRecordThreads.removeItem(input); 5688 } 5689 thread->exit(); 5690 // The thread entity (active unit of execution) is no longer running here, 5691 // but the ThreadBase container still exists. 5692 5693 AudioStreamIn *in = thread->clearInput(); 5694 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5695 // from now on thread->mInput is NULL 5696 in->hwDev->close_input_stream(in->hwDev, in->stream); 5697 delete in; 5698 5699 return NO_ERROR; 5700} 5701 5702status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5703{ 5704 Mutex::Autolock _l(mLock); 5705 MixerThread *dstThread = checkMixerThread_l(output); 5706 if (dstThread == NULL) { 5707 ALOGW("setStreamOutput() bad output id %d", output); 5708 return BAD_VALUE; 5709 } 5710 5711 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5712 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5713 5714 dstThread->setStreamValid(stream, true); 5715 5716 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5717 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5718 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5719 MixerThread *srcThread = (MixerThread *)thread; 5720 srcThread->setStreamValid(stream, false); 5721 srcThread->invalidateTracks(stream); 5722 } 5723 } 5724 5725 return NO_ERROR; 5726} 5727 5728 5729int AudioFlinger::newAudioSessionId() 5730{ 5731 return nextUniqueId(); 5732} 5733 5734void AudioFlinger::acquireAudioSessionId(int audioSession) 5735{ 5736 Mutex::Autolock _l(mLock); 5737 pid_t caller = IPCThreadState::self()->getCallingPid(); 5738 ALOGV("acquiring %d from %d", audioSession, caller); 5739 size_t num = mAudioSessionRefs.size(); 5740 for (size_t i = 0; i< num; i++) { 5741 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5742 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5743 ref->mCnt++; 5744 ALOGV(" incremented refcount to %d", ref->mCnt); 5745 return; 5746 } 5747 } 5748 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5749 ALOGV(" added new entry for %d", audioSession); 5750} 5751 5752void AudioFlinger::releaseAudioSessionId(int audioSession) 5753{ 5754 Mutex::Autolock _l(mLock); 5755 pid_t caller = IPCThreadState::self()->getCallingPid(); 5756 ALOGV("releasing %d from %d", audioSession, caller); 5757 size_t num = mAudioSessionRefs.size(); 5758 for (size_t i = 0; i< num; i++) { 5759 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5760 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5761 ref->mCnt--; 5762 ALOGV(" decremented refcount to %d", ref->mCnt); 5763 if (ref->mCnt == 0) { 5764 mAudioSessionRefs.removeAt(i); 5765 delete ref; 5766 purgeStaleEffects_l(); 5767 } 5768 return; 5769 } 5770 } 5771 ALOGW("session id %d not found for pid %d", audioSession, caller); 5772} 5773 5774void AudioFlinger::purgeStaleEffects_l() { 5775 5776 ALOGV("purging stale effects"); 5777 5778 Vector< sp<EffectChain> > chains; 5779 5780 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5781 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5782 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5783 sp<EffectChain> ec = t->mEffectChains[j]; 5784 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5785 chains.push(ec); 5786 } 5787 } 5788 } 5789 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5790 sp<RecordThread> t = mRecordThreads.valueAt(i); 5791 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5792 sp<EffectChain> ec = t->mEffectChains[j]; 5793 chains.push(ec); 5794 } 5795 } 5796 5797 for (size_t i = 0; i < chains.size(); i++) { 5798 sp<EffectChain> ec = chains[i]; 5799 int sessionid = ec->sessionId(); 5800 sp<ThreadBase> t = ec->mThread.promote(); 5801 if (t == 0) { 5802 continue; 5803 } 5804 size_t numsessionrefs = mAudioSessionRefs.size(); 5805 bool found = false; 5806 for (size_t k = 0; k < numsessionrefs; k++) { 5807 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5808 if (ref->mSessionid == sessionid) { 5809 ALOGV(" session %d still exists for %d with %d refs", 5810 sessionid, ref->mPid, ref->mCnt); 5811 found = true; 5812 break; 5813 } 5814 } 5815 if (!found) { 5816 // remove all effects from the chain 5817 while (ec->mEffects.size()) { 5818 sp<EffectModule> effect = ec->mEffects[0]; 5819 effect->unPin(); 5820 Mutex::Autolock _l (t->mLock); 5821 t->removeEffect_l(effect); 5822 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5823 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5824 if (handle != 0) { 5825 handle->mEffect.clear(); 5826 if (handle->mHasControl && handle->mEnabled) { 5827 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5828 } 5829 } 5830 } 5831 AudioSystem::unregisterEffect(effect->id()); 5832 } 5833 } 5834 } 5835 return; 5836} 5837 5838// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5839AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5840{ 5841 return mPlaybackThreads.valueFor(output).get(); 5842} 5843 5844// checkMixerThread_l() must be called with AudioFlinger::mLock held 5845AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5846{ 5847 PlaybackThread *thread = checkPlaybackThread_l(output); 5848 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5849} 5850 5851// checkRecordThread_l() must be called with AudioFlinger::mLock held 5852AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5853{ 5854 return mRecordThreads.valueFor(input).get(); 5855} 5856 5857uint32_t AudioFlinger::nextUniqueId() 5858{ 5859 return android_atomic_inc(&mNextUniqueId); 5860} 5861 5862AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5863{ 5864 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5865 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5866 AudioStreamOut *output = thread->getOutput(); 5867 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5868 return thread; 5869 } 5870 } 5871 return NULL; 5872} 5873 5874uint32_t AudioFlinger::primaryOutputDevice_l() const 5875{ 5876 PlaybackThread *thread = primaryPlaybackThread_l(); 5877 5878 if (thread == NULL) { 5879 return 0; 5880 } 5881 5882 return thread->device(); 5883} 5884 5885 5886// ---------------------------------------------------------------------------- 5887// Effect management 5888// ---------------------------------------------------------------------------- 5889 5890 5891status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5892{ 5893 Mutex::Autolock _l(mLock); 5894 return EffectQueryNumberEffects(numEffects); 5895} 5896 5897status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5898{ 5899 Mutex::Autolock _l(mLock); 5900 return EffectQueryEffect(index, descriptor); 5901} 5902 5903status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5904 effect_descriptor_t *descriptor) const 5905{ 5906 Mutex::Autolock _l(mLock); 5907 return EffectGetDescriptor(pUuid, descriptor); 5908} 5909 5910 5911sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5912 effect_descriptor_t *pDesc, 5913 const sp<IEffectClient>& effectClient, 5914 int32_t priority, 5915 audio_io_handle_t io, 5916 int sessionId, 5917 status_t *status, 5918 int *id, 5919 int *enabled) 5920{ 5921 status_t lStatus = NO_ERROR; 5922 sp<EffectHandle> handle; 5923 effect_descriptor_t desc; 5924 5925 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5926 pid, effectClient.get(), priority, sessionId, io); 5927 5928 if (pDesc == NULL) { 5929 lStatus = BAD_VALUE; 5930 goto Exit; 5931 } 5932 5933 // check audio settings permission for global effects 5934 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5935 lStatus = PERMISSION_DENIED; 5936 goto Exit; 5937 } 5938 5939 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5940 // that can only be created by audio policy manager (running in same process) 5941 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5942 lStatus = PERMISSION_DENIED; 5943 goto Exit; 5944 } 5945 5946 if (io == 0) { 5947 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5948 // output must be specified by AudioPolicyManager when using session 5949 // AUDIO_SESSION_OUTPUT_STAGE 5950 lStatus = BAD_VALUE; 5951 goto Exit; 5952 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5953 // if the output returned by getOutputForEffect() is removed before we lock the 5954 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5955 // and we will exit safely 5956 io = AudioSystem::getOutputForEffect(&desc); 5957 } 5958 } 5959 5960 { 5961 Mutex::Autolock _l(mLock); 5962 5963 5964 if (!EffectIsNullUuid(&pDesc->uuid)) { 5965 // if uuid is specified, request effect descriptor 5966 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5967 if (lStatus < 0) { 5968 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5969 goto Exit; 5970 } 5971 } else { 5972 // if uuid is not specified, look for an available implementation 5973 // of the required type in effect factory 5974 if (EffectIsNullUuid(&pDesc->type)) { 5975 ALOGW("createEffect() no effect type"); 5976 lStatus = BAD_VALUE; 5977 goto Exit; 5978 } 5979 uint32_t numEffects = 0; 5980 effect_descriptor_t d; 5981 d.flags = 0; // prevent compiler warning 5982 bool found = false; 5983 5984 lStatus = EffectQueryNumberEffects(&numEffects); 5985 if (lStatus < 0) { 5986 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5987 goto Exit; 5988 } 5989 for (uint32_t i = 0; i < numEffects; i++) { 5990 lStatus = EffectQueryEffect(i, &desc); 5991 if (lStatus < 0) { 5992 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5993 continue; 5994 } 5995 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5996 // If matching type found save effect descriptor. If the session is 5997 // 0 and the effect is not auxiliary, continue enumeration in case 5998 // an auxiliary version of this effect type is available 5999 found = true; 6000 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6001 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6002 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6003 break; 6004 } 6005 } 6006 } 6007 if (!found) { 6008 lStatus = BAD_VALUE; 6009 ALOGW("createEffect() effect not found"); 6010 goto Exit; 6011 } 6012 // For same effect type, chose auxiliary version over insert version if 6013 // connect to output mix (Compliance to OpenSL ES) 6014 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6015 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6016 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6017 } 6018 } 6019 6020 // Do not allow auxiliary effects on a session different from 0 (output mix) 6021 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6022 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6023 lStatus = INVALID_OPERATION; 6024 goto Exit; 6025 } 6026 6027 // check recording permission for visualizer 6028 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6029 !recordingAllowed()) { 6030 lStatus = PERMISSION_DENIED; 6031 goto Exit; 6032 } 6033 6034 // return effect descriptor 6035 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6036 6037 // If output is not specified try to find a matching audio session ID in one of the 6038 // output threads. 6039 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6040 // because of code checking output when entering the function. 6041 // Note: io is never 0 when creating an effect on an input 6042 if (io == 0) { 6043 // look for the thread where the specified audio session is present 6044 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6045 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6046 io = mPlaybackThreads.keyAt(i); 6047 break; 6048 } 6049 } 6050 if (io == 0) { 6051 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6052 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6053 io = mRecordThreads.keyAt(i); 6054 break; 6055 } 6056 } 6057 } 6058 // If no output thread contains the requested session ID, default to 6059 // first output. The effect chain will be moved to the correct output 6060 // thread when a track with the same session ID is created 6061 if (io == 0 && mPlaybackThreads.size()) { 6062 io = mPlaybackThreads.keyAt(0); 6063 } 6064 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6065 } 6066 ThreadBase *thread = checkRecordThread_l(io); 6067 if (thread == NULL) { 6068 thread = checkPlaybackThread_l(io); 6069 if (thread == NULL) { 6070 ALOGE("createEffect() unknown output thread"); 6071 lStatus = BAD_VALUE; 6072 goto Exit; 6073 } 6074 } 6075 6076 sp<Client> client = registerPid_l(pid); 6077 6078 // create effect on selected output thread 6079 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6080 &desc, enabled, &lStatus); 6081 if (handle != 0 && id != NULL) { 6082 *id = handle->id(); 6083 } 6084 } 6085 6086Exit: 6087 if (status != NULL) { 6088 *status = lStatus; 6089 } 6090 return handle; 6091} 6092 6093status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6094 audio_io_handle_t dstOutput) 6095{ 6096 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6097 sessionId, srcOutput, dstOutput); 6098 Mutex::Autolock _l(mLock); 6099 if (srcOutput == dstOutput) { 6100 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6101 return NO_ERROR; 6102 } 6103 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6104 if (srcThread == NULL) { 6105 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6106 return BAD_VALUE; 6107 } 6108 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6109 if (dstThread == NULL) { 6110 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6111 return BAD_VALUE; 6112 } 6113 6114 Mutex::Autolock _dl(dstThread->mLock); 6115 Mutex::Autolock _sl(srcThread->mLock); 6116 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6117 6118 return NO_ERROR; 6119} 6120 6121// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6122status_t AudioFlinger::moveEffectChain_l(int sessionId, 6123 AudioFlinger::PlaybackThread *srcThread, 6124 AudioFlinger::PlaybackThread *dstThread, 6125 bool reRegister) 6126{ 6127 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6128 sessionId, srcThread, dstThread); 6129 6130 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6131 if (chain == 0) { 6132 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6133 sessionId, srcThread); 6134 return INVALID_OPERATION; 6135 } 6136 6137 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6138 // so that a new chain is created with correct parameters when first effect is added. This is 6139 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6140 // removed. 6141 srcThread->removeEffectChain_l(chain); 6142 6143 // transfer all effects one by one so that new effect chain is created on new thread with 6144 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6145 audio_io_handle_t dstOutput = dstThread->id(); 6146 sp<EffectChain> dstChain; 6147 uint32_t strategy = 0; // prevent compiler warning 6148 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6149 while (effect != 0) { 6150 srcThread->removeEffect_l(effect); 6151 dstThread->addEffect_l(effect); 6152 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6153 if (effect->state() == EffectModule::ACTIVE || 6154 effect->state() == EffectModule::STOPPING) { 6155 effect->start(); 6156 } 6157 // if the move request is not received from audio policy manager, the effect must be 6158 // re-registered with the new strategy and output 6159 if (dstChain == 0) { 6160 dstChain = effect->chain().promote(); 6161 if (dstChain == 0) { 6162 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6163 srcThread->addEffect_l(effect); 6164 return NO_INIT; 6165 } 6166 strategy = dstChain->strategy(); 6167 } 6168 if (reRegister) { 6169 AudioSystem::unregisterEffect(effect->id()); 6170 AudioSystem::registerEffect(&effect->desc(), 6171 dstOutput, 6172 strategy, 6173 sessionId, 6174 effect->id()); 6175 } 6176 effect = chain->getEffectFromId_l(0); 6177 } 6178 6179 return NO_ERROR; 6180} 6181 6182 6183// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6184sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6185 const sp<AudioFlinger::Client>& client, 6186 const sp<IEffectClient>& effectClient, 6187 int32_t priority, 6188 int sessionId, 6189 effect_descriptor_t *desc, 6190 int *enabled, 6191 status_t *status 6192 ) 6193{ 6194 sp<EffectModule> effect; 6195 sp<EffectHandle> handle; 6196 status_t lStatus; 6197 sp<EffectChain> chain; 6198 bool chainCreated = false; 6199 bool effectCreated = false; 6200 bool effectRegistered = false; 6201 6202 lStatus = initCheck(); 6203 if (lStatus != NO_ERROR) { 6204 ALOGW("createEffect_l() Audio driver not initialized."); 6205 goto Exit; 6206 } 6207 6208 // Do not allow effects with session ID 0 on direct output or duplicating threads 6209 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6210 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6211 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6212 desc->name, sessionId); 6213 lStatus = BAD_VALUE; 6214 goto Exit; 6215 } 6216 // Only Pre processor effects are allowed on input threads and only on input threads 6217 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6218 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6219 desc->name, desc->flags, mType); 6220 lStatus = BAD_VALUE; 6221 goto Exit; 6222 } 6223 6224 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6225 6226 { // scope for mLock 6227 Mutex::Autolock _l(mLock); 6228 6229 // check for existing effect chain with the requested audio session 6230 chain = getEffectChain_l(sessionId); 6231 if (chain == 0) { 6232 // create a new chain for this session 6233 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6234 chain = new EffectChain(this, sessionId); 6235 addEffectChain_l(chain); 6236 chain->setStrategy(getStrategyForSession_l(sessionId)); 6237 chainCreated = true; 6238 } else { 6239 effect = chain->getEffectFromDesc_l(desc); 6240 } 6241 6242 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6243 6244 if (effect == 0) { 6245 int id = mAudioFlinger->nextUniqueId(); 6246 // Check CPU and memory usage 6247 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6248 if (lStatus != NO_ERROR) { 6249 goto Exit; 6250 } 6251 effectRegistered = true; 6252 // create a new effect module if none present in the chain 6253 effect = new EffectModule(this, chain, desc, id, sessionId); 6254 lStatus = effect->status(); 6255 if (lStatus != NO_ERROR) { 6256 goto Exit; 6257 } 6258 lStatus = chain->addEffect_l(effect); 6259 if (lStatus != NO_ERROR) { 6260 goto Exit; 6261 } 6262 effectCreated = true; 6263 6264 effect->setDevice(mDevice); 6265 effect->setMode(mAudioFlinger->getMode()); 6266 } 6267 // create effect handle and connect it to effect module 6268 handle = new EffectHandle(effect, client, effectClient, priority); 6269 lStatus = effect->addHandle(handle); 6270 if (enabled != NULL) { 6271 *enabled = (int)effect->isEnabled(); 6272 } 6273 } 6274 6275Exit: 6276 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6277 Mutex::Autolock _l(mLock); 6278 if (effectCreated) { 6279 chain->removeEffect_l(effect); 6280 } 6281 if (effectRegistered) { 6282 AudioSystem::unregisterEffect(effect->id()); 6283 } 6284 if (chainCreated) { 6285 removeEffectChain_l(chain); 6286 } 6287 handle.clear(); 6288 } 6289 6290 if (status != NULL) { 6291 *status = lStatus; 6292 } 6293 return handle; 6294} 6295 6296sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6297{ 6298 sp<EffectChain> chain = getEffectChain_l(sessionId); 6299 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6300} 6301 6302// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6303// PlaybackThread::mLock held 6304status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6305{ 6306 // check for existing effect chain with the requested audio session 6307 int sessionId = effect->sessionId(); 6308 sp<EffectChain> chain = getEffectChain_l(sessionId); 6309 bool chainCreated = false; 6310 6311 if (chain == 0) { 6312 // create a new chain for this session 6313 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6314 chain = new EffectChain(this, sessionId); 6315 addEffectChain_l(chain); 6316 chain->setStrategy(getStrategyForSession_l(sessionId)); 6317 chainCreated = true; 6318 } 6319 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6320 6321 if (chain->getEffectFromId_l(effect->id()) != 0) { 6322 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6323 this, effect->desc().name, chain.get()); 6324 return BAD_VALUE; 6325 } 6326 6327 status_t status = chain->addEffect_l(effect); 6328 if (status != NO_ERROR) { 6329 if (chainCreated) { 6330 removeEffectChain_l(chain); 6331 } 6332 return status; 6333 } 6334 6335 effect->setDevice(mDevice); 6336 effect->setMode(mAudioFlinger->getMode()); 6337 return NO_ERROR; 6338} 6339 6340void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6341 6342 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6343 effect_descriptor_t desc = effect->desc(); 6344 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6345 detachAuxEffect_l(effect->id()); 6346 } 6347 6348 sp<EffectChain> chain = effect->chain().promote(); 6349 if (chain != 0) { 6350 // remove effect chain if removing last effect 6351 if (chain->removeEffect_l(effect) == 0) { 6352 removeEffectChain_l(chain); 6353 } 6354 } else { 6355 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6356 } 6357} 6358 6359void AudioFlinger::ThreadBase::lockEffectChains_l( 6360 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6361{ 6362 effectChains = mEffectChains; 6363 for (size_t i = 0; i < mEffectChains.size(); i++) { 6364 mEffectChains[i]->lock(); 6365 } 6366} 6367 6368void AudioFlinger::ThreadBase::unlockEffectChains( 6369 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6370{ 6371 for (size_t i = 0; i < effectChains.size(); i++) { 6372 effectChains[i]->unlock(); 6373 } 6374} 6375 6376sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6377{ 6378 Mutex::Autolock _l(mLock); 6379 return getEffectChain_l(sessionId); 6380} 6381 6382sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6383{ 6384 size_t size = mEffectChains.size(); 6385 for (size_t i = 0; i < size; i++) { 6386 if (mEffectChains[i]->sessionId() == sessionId) { 6387 return mEffectChains[i]; 6388 } 6389 } 6390 return 0; 6391} 6392 6393void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6394{ 6395 Mutex::Autolock _l(mLock); 6396 size_t size = mEffectChains.size(); 6397 for (size_t i = 0; i < size; i++) { 6398 mEffectChains[i]->setMode_l(mode); 6399 } 6400} 6401 6402void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6403 const wp<EffectHandle>& handle, 6404 bool unpinIfLast) { 6405 6406 Mutex::Autolock _l(mLock); 6407 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6408 // delete the effect module if removing last handle on it 6409 if (effect->removeHandle(handle) == 0) { 6410 if (!effect->isPinned() || unpinIfLast) { 6411 removeEffect_l(effect); 6412 AudioSystem::unregisterEffect(effect->id()); 6413 } 6414 } 6415} 6416 6417status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6418{ 6419 int session = chain->sessionId(); 6420 int16_t *buffer = mMixBuffer; 6421 bool ownsBuffer = false; 6422 6423 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6424 if (session > 0) { 6425 // Only one effect chain can be present in direct output thread and it uses 6426 // the mix buffer as input 6427 if (mType != DIRECT) { 6428 size_t numSamples = mFrameCount * mChannelCount; 6429 buffer = new int16_t[numSamples]; 6430 memset(buffer, 0, numSamples * sizeof(int16_t)); 6431 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6432 ownsBuffer = true; 6433 } 6434 6435 // Attach all tracks with same session ID to this chain. 6436 for (size_t i = 0; i < mTracks.size(); ++i) { 6437 sp<Track> track = mTracks[i]; 6438 if (session == track->sessionId()) { 6439 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6440 track->setMainBuffer(buffer); 6441 chain->incTrackCnt(); 6442 } 6443 } 6444 6445 // indicate all active tracks in the chain 6446 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6447 sp<Track> track = mActiveTracks[i].promote(); 6448 if (track == 0) continue; 6449 if (session == track->sessionId()) { 6450 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6451 chain->incActiveTrackCnt(); 6452 } 6453 } 6454 } 6455 6456 chain->setInBuffer(buffer, ownsBuffer); 6457 chain->setOutBuffer(mMixBuffer); 6458 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6459 // chains list in order to be processed last as it contains output stage effects 6460 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6461 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6462 // after track specific effects and before output stage 6463 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6464 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6465 // Effect chain for other sessions are inserted at beginning of effect 6466 // chains list to be processed before output mix effects. Relative order between other 6467 // sessions is not important 6468 size_t size = mEffectChains.size(); 6469 size_t i = 0; 6470 for (i = 0; i < size; i++) { 6471 if (mEffectChains[i]->sessionId() < session) break; 6472 } 6473 mEffectChains.insertAt(chain, i); 6474 checkSuspendOnAddEffectChain_l(chain); 6475 6476 return NO_ERROR; 6477} 6478 6479size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6480{ 6481 int session = chain->sessionId(); 6482 6483 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6484 6485 for (size_t i = 0; i < mEffectChains.size(); i++) { 6486 if (chain == mEffectChains[i]) { 6487 mEffectChains.removeAt(i); 6488 // detach all active tracks from the chain 6489 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6490 sp<Track> track = mActiveTracks[i].promote(); 6491 if (track == 0) continue; 6492 if (session == track->sessionId()) { 6493 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6494 chain.get(), session); 6495 chain->decActiveTrackCnt(); 6496 } 6497 } 6498 6499 // detach all tracks with same session ID from this chain 6500 for (size_t i = 0; i < mTracks.size(); ++i) { 6501 sp<Track> track = mTracks[i]; 6502 if (session == track->sessionId()) { 6503 track->setMainBuffer(mMixBuffer); 6504 chain->decTrackCnt(); 6505 } 6506 } 6507 break; 6508 } 6509 } 6510 return mEffectChains.size(); 6511} 6512 6513status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6514 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6515{ 6516 Mutex::Autolock _l(mLock); 6517 return attachAuxEffect_l(track, EffectId); 6518} 6519 6520status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6521 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6522{ 6523 status_t status = NO_ERROR; 6524 6525 if (EffectId == 0) { 6526 track->setAuxBuffer(0, NULL); 6527 } else { 6528 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6529 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6530 if (effect != 0) { 6531 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6532 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6533 } else { 6534 status = INVALID_OPERATION; 6535 } 6536 } else { 6537 status = BAD_VALUE; 6538 } 6539 } 6540 return status; 6541} 6542 6543void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6544{ 6545 for (size_t i = 0; i < mTracks.size(); ++i) { 6546 sp<Track> track = mTracks[i]; 6547 if (track->auxEffectId() == effectId) { 6548 attachAuxEffect_l(track, 0); 6549 } 6550 } 6551} 6552 6553status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6554{ 6555 // only one chain per input thread 6556 if (mEffectChains.size() != 0) { 6557 return INVALID_OPERATION; 6558 } 6559 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6560 6561 chain->setInBuffer(NULL); 6562 chain->setOutBuffer(NULL); 6563 6564 checkSuspendOnAddEffectChain_l(chain); 6565 6566 mEffectChains.add(chain); 6567 6568 return NO_ERROR; 6569} 6570 6571size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6572{ 6573 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6574 ALOGW_IF(mEffectChains.size() != 1, 6575 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6576 chain.get(), mEffectChains.size(), this); 6577 if (mEffectChains.size() == 1) { 6578 mEffectChains.removeAt(0); 6579 } 6580 return 0; 6581} 6582 6583// ---------------------------------------------------------------------------- 6584// EffectModule implementation 6585// ---------------------------------------------------------------------------- 6586 6587#undef LOG_TAG 6588#define LOG_TAG "AudioFlinger::EffectModule" 6589 6590AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6591 const wp<AudioFlinger::EffectChain>& chain, 6592 effect_descriptor_t *desc, 6593 int id, 6594 int sessionId) 6595 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6596 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6597{ 6598 ALOGV("Constructor %p", this); 6599 int lStatus; 6600 if (thread == NULL) { 6601 return; 6602 } 6603 6604 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6605 6606 // create effect engine from effect factory 6607 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6608 6609 if (mStatus != NO_ERROR) { 6610 return; 6611 } 6612 lStatus = init(); 6613 if (lStatus < 0) { 6614 mStatus = lStatus; 6615 goto Error; 6616 } 6617 6618 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6619 mPinned = true; 6620 } 6621 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6622 return; 6623Error: 6624 EffectRelease(mEffectInterface); 6625 mEffectInterface = NULL; 6626 ALOGV("Constructor Error %d", mStatus); 6627} 6628 6629AudioFlinger::EffectModule::~EffectModule() 6630{ 6631 ALOGV("Destructor %p", this); 6632 if (mEffectInterface != NULL) { 6633 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6634 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6635 sp<ThreadBase> thread = mThread.promote(); 6636 if (thread != 0) { 6637 audio_stream_t *stream = thread->stream(); 6638 if (stream != NULL) { 6639 stream->remove_audio_effect(stream, mEffectInterface); 6640 } 6641 } 6642 } 6643 // release effect engine 6644 EffectRelease(mEffectInterface); 6645 } 6646} 6647 6648status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6649{ 6650 status_t status; 6651 6652 Mutex::Autolock _l(mLock); 6653 int priority = handle->priority(); 6654 size_t size = mHandles.size(); 6655 sp<EffectHandle> h; 6656 size_t i; 6657 for (i = 0; i < size; i++) { 6658 h = mHandles[i].promote(); 6659 if (h == 0) continue; 6660 if (h->priority() <= priority) break; 6661 } 6662 // if inserted in first place, move effect control from previous owner to this handle 6663 if (i == 0) { 6664 bool enabled = false; 6665 if (h != 0) { 6666 enabled = h->enabled(); 6667 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6668 } 6669 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6670 status = NO_ERROR; 6671 } else { 6672 status = ALREADY_EXISTS; 6673 } 6674 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6675 mHandles.insertAt(handle, i); 6676 return status; 6677} 6678 6679size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6680{ 6681 Mutex::Autolock _l(mLock); 6682 size_t size = mHandles.size(); 6683 size_t i; 6684 for (i = 0; i < size; i++) { 6685 if (mHandles[i] == handle) break; 6686 } 6687 if (i == size) { 6688 return size; 6689 } 6690 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6691 6692 bool enabled = false; 6693 EffectHandle *hdl = handle.unsafe_get(); 6694 if (hdl != NULL) { 6695 ALOGV("removeHandle() unsafe_get OK"); 6696 enabled = hdl->enabled(); 6697 } 6698 mHandles.removeAt(i); 6699 size = mHandles.size(); 6700 // if removed from first place, move effect control from this handle to next in line 6701 if (i == 0 && size != 0) { 6702 sp<EffectHandle> h = mHandles[0].promote(); 6703 if (h != 0) { 6704 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6705 } 6706 } 6707 6708 // Prevent calls to process() and other functions on effect interface from now on. 6709 // The effect engine will be released by the destructor when the last strong reference on 6710 // this object is released which can happen after next process is called. 6711 if (size == 0 && !mPinned) { 6712 mState = DESTROYED; 6713 } 6714 6715 return size; 6716} 6717 6718sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6719{ 6720 Mutex::Autolock _l(mLock); 6721 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6722} 6723 6724void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6725{ 6726 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6727 // keep a strong reference on this EffectModule to avoid calling the 6728 // destructor before we exit 6729 sp<EffectModule> keep(this); 6730 { 6731 sp<ThreadBase> thread = mThread.promote(); 6732 if (thread != 0) { 6733 thread->disconnectEffect(keep, handle, unpinIfLast); 6734 } 6735 } 6736} 6737 6738void AudioFlinger::EffectModule::updateState() { 6739 Mutex::Autolock _l(mLock); 6740 6741 switch (mState) { 6742 case RESTART: 6743 reset_l(); 6744 // FALL THROUGH 6745 6746 case STARTING: 6747 // clear auxiliary effect input buffer for next accumulation 6748 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6749 memset(mConfig.inputCfg.buffer.raw, 6750 0, 6751 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6752 } 6753 start_l(); 6754 mState = ACTIVE; 6755 break; 6756 case STOPPING: 6757 stop_l(); 6758 mDisableWaitCnt = mMaxDisableWaitCnt; 6759 mState = STOPPED; 6760 break; 6761 case STOPPED: 6762 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6763 // turn off sequence. 6764 if (--mDisableWaitCnt == 0) { 6765 reset_l(); 6766 mState = IDLE; 6767 } 6768 break; 6769 default: //IDLE , ACTIVE, DESTROYED 6770 break; 6771 } 6772} 6773 6774void AudioFlinger::EffectModule::process() 6775{ 6776 Mutex::Autolock _l(mLock); 6777 6778 if (mState == DESTROYED || mEffectInterface == NULL || 6779 mConfig.inputCfg.buffer.raw == NULL || 6780 mConfig.outputCfg.buffer.raw == NULL) { 6781 return; 6782 } 6783 6784 if (isProcessEnabled()) { 6785 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6786 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6787 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6788 mConfig.inputCfg.buffer.s32, 6789 mConfig.inputCfg.buffer.frameCount/2); 6790 } 6791 6792 // do the actual processing in the effect engine 6793 int ret = (*mEffectInterface)->process(mEffectInterface, 6794 &mConfig.inputCfg.buffer, 6795 &mConfig.outputCfg.buffer); 6796 6797 // force transition to IDLE state when engine is ready 6798 if (mState == STOPPED && ret == -ENODATA) { 6799 mDisableWaitCnt = 1; 6800 } 6801 6802 // clear auxiliary effect input buffer for next accumulation 6803 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6804 memset(mConfig.inputCfg.buffer.raw, 0, 6805 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6806 } 6807 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6808 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6809 // If an insert effect is idle and input buffer is different from output buffer, 6810 // accumulate input onto output 6811 sp<EffectChain> chain = mChain.promote(); 6812 if (chain != 0 && chain->activeTrackCnt() != 0) { 6813 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6814 int16_t *in = mConfig.inputCfg.buffer.s16; 6815 int16_t *out = mConfig.outputCfg.buffer.s16; 6816 for (size_t i = 0; i < frameCnt; i++) { 6817 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6818 } 6819 } 6820 } 6821} 6822 6823void AudioFlinger::EffectModule::reset_l() 6824{ 6825 if (mEffectInterface == NULL) { 6826 return; 6827 } 6828 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6829} 6830 6831status_t AudioFlinger::EffectModule::configure() 6832{ 6833 uint32_t channels; 6834 if (mEffectInterface == NULL) { 6835 return NO_INIT; 6836 } 6837 6838 sp<ThreadBase> thread = mThread.promote(); 6839 if (thread == 0) { 6840 return DEAD_OBJECT; 6841 } 6842 6843 // TODO: handle configuration of effects replacing track process 6844 if (thread->channelCount() == 1) { 6845 channels = AUDIO_CHANNEL_OUT_MONO; 6846 } else { 6847 channels = AUDIO_CHANNEL_OUT_STEREO; 6848 } 6849 6850 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6851 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6852 } else { 6853 mConfig.inputCfg.channels = channels; 6854 } 6855 mConfig.outputCfg.channels = channels; 6856 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6857 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6858 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6859 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6860 mConfig.inputCfg.bufferProvider.cookie = NULL; 6861 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6862 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6863 mConfig.outputCfg.bufferProvider.cookie = NULL; 6864 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6865 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6866 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6867 // Insert effect: 6868 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6869 // always overwrites output buffer: input buffer == output buffer 6870 // - in other sessions: 6871 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6872 // other effect: overwrites output buffer: input buffer == output buffer 6873 // Auxiliary effect: 6874 // accumulates in output buffer: input buffer != output buffer 6875 // Therefore: accumulate <=> input buffer != output buffer 6876 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6877 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6878 } else { 6879 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6880 } 6881 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6882 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6883 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6884 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6885 6886 ALOGV("configure() %p thread %p buffer %p framecount %d", 6887 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6888 6889 status_t cmdStatus; 6890 uint32_t size = sizeof(int); 6891 status_t status = (*mEffectInterface)->command(mEffectInterface, 6892 EFFECT_CMD_SET_CONFIG, 6893 sizeof(effect_config_t), 6894 &mConfig, 6895 &size, 6896 &cmdStatus); 6897 if (status == 0) { 6898 status = cmdStatus; 6899 } 6900 6901 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6902 (1000 * mConfig.outputCfg.buffer.frameCount); 6903 6904 return status; 6905} 6906 6907status_t AudioFlinger::EffectModule::init() 6908{ 6909 Mutex::Autolock _l(mLock); 6910 if (mEffectInterface == NULL) { 6911 return NO_INIT; 6912 } 6913 status_t cmdStatus; 6914 uint32_t size = sizeof(status_t); 6915 status_t status = (*mEffectInterface)->command(mEffectInterface, 6916 EFFECT_CMD_INIT, 6917 0, 6918 NULL, 6919 &size, 6920 &cmdStatus); 6921 if (status == 0) { 6922 status = cmdStatus; 6923 } 6924 return status; 6925} 6926 6927status_t AudioFlinger::EffectModule::start() 6928{ 6929 Mutex::Autolock _l(mLock); 6930 return start_l(); 6931} 6932 6933status_t AudioFlinger::EffectModule::start_l() 6934{ 6935 if (mEffectInterface == NULL) { 6936 return NO_INIT; 6937 } 6938 status_t cmdStatus; 6939 uint32_t size = sizeof(status_t); 6940 status_t status = (*mEffectInterface)->command(mEffectInterface, 6941 EFFECT_CMD_ENABLE, 6942 0, 6943 NULL, 6944 &size, 6945 &cmdStatus); 6946 if (status == 0) { 6947 status = cmdStatus; 6948 } 6949 if (status == 0 && 6950 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6951 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6952 sp<ThreadBase> thread = mThread.promote(); 6953 if (thread != 0) { 6954 audio_stream_t *stream = thread->stream(); 6955 if (stream != NULL) { 6956 stream->add_audio_effect(stream, mEffectInterface); 6957 } 6958 } 6959 } 6960 return status; 6961} 6962 6963status_t AudioFlinger::EffectModule::stop() 6964{ 6965 Mutex::Autolock _l(mLock); 6966 return stop_l(); 6967} 6968 6969status_t AudioFlinger::EffectModule::stop_l() 6970{ 6971 if (mEffectInterface == NULL) { 6972 return NO_INIT; 6973 } 6974 status_t cmdStatus; 6975 uint32_t size = sizeof(status_t); 6976 status_t status = (*mEffectInterface)->command(mEffectInterface, 6977 EFFECT_CMD_DISABLE, 6978 0, 6979 NULL, 6980 &size, 6981 &cmdStatus); 6982 if (status == 0) { 6983 status = cmdStatus; 6984 } 6985 if (status == 0 && 6986 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6987 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6988 sp<ThreadBase> thread = mThread.promote(); 6989 if (thread != 0) { 6990 audio_stream_t *stream = thread->stream(); 6991 if (stream != NULL) { 6992 stream->remove_audio_effect(stream, mEffectInterface); 6993 } 6994 } 6995 } 6996 return status; 6997} 6998 6999status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7000 uint32_t cmdSize, 7001 void *pCmdData, 7002 uint32_t *replySize, 7003 void *pReplyData) 7004{ 7005 Mutex::Autolock _l(mLock); 7006// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7007 7008 if (mState == DESTROYED || mEffectInterface == NULL) { 7009 return NO_INIT; 7010 } 7011 status_t status = (*mEffectInterface)->command(mEffectInterface, 7012 cmdCode, 7013 cmdSize, 7014 pCmdData, 7015 replySize, 7016 pReplyData); 7017 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7018 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7019 for (size_t i = 1; i < mHandles.size(); i++) { 7020 sp<EffectHandle> h = mHandles[i].promote(); 7021 if (h != 0) { 7022 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7023 } 7024 } 7025 } 7026 return status; 7027} 7028 7029status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7030{ 7031 7032 Mutex::Autolock _l(mLock); 7033 ALOGV("setEnabled %p enabled %d", this, enabled); 7034 7035 if (enabled != isEnabled()) { 7036 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7037 if (enabled && status != NO_ERROR) { 7038 return status; 7039 } 7040 7041 switch (mState) { 7042 // going from disabled to enabled 7043 case IDLE: 7044 mState = STARTING; 7045 break; 7046 case STOPPED: 7047 mState = RESTART; 7048 break; 7049 case STOPPING: 7050 mState = ACTIVE; 7051 break; 7052 7053 // going from enabled to disabled 7054 case RESTART: 7055 mState = STOPPED; 7056 break; 7057 case STARTING: 7058 mState = IDLE; 7059 break; 7060 case ACTIVE: 7061 mState = STOPPING; 7062 break; 7063 case DESTROYED: 7064 return NO_ERROR; // simply ignore as we are being destroyed 7065 } 7066 for (size_t i = 1; i < mHandles.size(); i++) { 7067 sp<EffectHandle> h = mHandles[i].promote(); 7068 if (h != 0) { 7069 h->setEnabled(enabled); 7070 } 7071 } 7072 } 7073 return NO_ERROR; 7074} 7075 7076bool AudioFlinger::EffectModule::isEnabled() const 7077{ 7078 switch (mState) { 7079 case RESTART: 7080 case STARTING: 7081 case ACTIVE: 7082 return true; 7083 case IDLE: 7084 case STOPPING: 7085 case STOPPED: 7086 case DESTROYED: 7087 default: 7088 return false; 7089 } 7090} 7091 7092bool AudioFlinger::EffectModule::isProcessEnabled() const 7093{ 7094 switch (mState) { 7095 case RESTART: 7096 case ACTIVE: 7097 case STOPPING: 7098 case STOPPED: 7099 return true; 7100 case IDLE: 7101 case STARTING: 7102 case DESTROYED: 7103 default: 7104 return false; 7105 } 7106} 7107 7108status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7109{ 7110 Mutex::Autolock _l(mLock); 7111 status_t status = NO_ERROR; 7112 7113 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7114 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7115 if (isProcessEnabled() && 7116 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7117 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7118 status_t cmdStatus; 7119 uint32_t volume[2]; 7120 uint32_t *pVolume = NULL; 7121 uint32_t size = sizeof(volume); 7122 volume[0] = *left; 7123 volume[1] = *right; 7124 if (controller) { 7125 pVolume = volume; 7126 } 7127 status = (*mEffectInterface)->command(mEffectInterface, 7128 EFFECT_CMD_SET_VOLUME, 7129 size, 7130 volume, 7131 &size, 7132 pVolume); 7133 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7134 *left = volume[0]; 7135 *right = volume[1]; 7136 } 7137 } 7138 return status; 7139} 7140 7141status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7142{ 7143 Mutex::Autolock _l(mLock); 7144 status_t status = NO_ERROR; 7145 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7146 // audio pre processing modules on RecordThread can receive both output and 7147 // input device indication in the same call 7148 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7149 if (dev) { 7150 status_t cmdStatus; 7151 uint32_t size = sizeof(status_t); 7152 7153 status = (*mEffectInterface)->command(mEffectInterface, 7154 EFFECT_CMD_SET_DEVICE, 7155 sizeof(uint32_t), 7156 &dev, 7157 &size, 7158 &cmdStatus); 7159 if (status == NO_ERROR) { 7160 status = cmdStatus; 7161 } 7162 } 7163 dev = device & AUDIO_DEVICE_IN_ALL; 7164 if (dev) { 7165 status_t cmdStatus; 7166 uint32_t size = sizeof(status_t); 7167 7168 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7169 EFFECT_CMD_SET_INPUT_DEVICE, 7170 sizeof(uint32_t), 7171 &dev, 7172 &size, 7173 &cmdStatus); 7174 if (status2 == NO_ERROR) { 7175 status2 = cmdStatus; 7176 } 7177 if (status == NO_ERROR) { 7178 status = status2; 7179 } 7180 } 7181 } 7182 return status; 7183} 7184 7185status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7186{ 7187 Mutex::Autolock _l(mLock); 7188 status_t status = NO_ERROR; 7189 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7190 status_t cmdStatus; 7191 uint32_t size = sizeof(status_t); 7192 status = (*mEffectInterface)->command(mEffectInterface, 7193 EFFECT_CMD_SET_AUDIO_MODE, 7194 sizeof(audio_mode_t), 7195 &mode, 7196 &size, 7197 &cmdStatus); 7198 if (status == NO_ERROR) { 7199 status = cmdStatus; 7200 } 7201 } 7202 return status; 7203} 7204 7205void AudioFlinger::EffectModule::setSuspended(bool suspended) 7206{ 7207 Mutex::Autolock _l(mLock); 7208 mSuspended = suspended; 7209} 7210 7211bool AudioFlinger::EffectModule::suspended() const 7212{ 7213 Mutex::Autolock _l(mLock); 7214 return mSuspended; 7215} 7216 7217status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7218{ 7219 const size_t SIZE = 256; 7220 char buffer[SIZE]; 7221 String8 result; 7222 7223 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7224 result.append(buffer); 7225 7226 bool locked = tryLock(mLock); 7227 // failed to lock - AudioFlinger is probably deadlocked 7228 if (!locked) { 7229 result.append("\t\tCould not lock Fx mutex:\n"); 7230 } 7231 7232 result.append("\t\tSession Status State Engine:\n"); 7233 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7234 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7235 result.append(buffer); 7236 7237 result.append("\t\tDescriptor:\n"); 7238 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7239 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7240 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7241 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7242 result.append(buffer); 7243 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7244 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7245 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7246 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7247 result.append(buffer); 7248 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7249 mDescriptor.apiVersion, 7250 mDescriptor.flags); 7251 result.append(buffer); 7252 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7253 mDescriptor.name); 7254 result.append(buffer); 7255 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7256 mDescriptor.implementor); 7257 result.append(buffer); 7258 7259 result.append("\t\t- Input configuration:\n"); 7260 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7261 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7262 (uint32_t)mConfig.inputCfg.buffer.raw, 7263 mConfig.inputCfg.buffer.frameCount, 7264 mConfig.inputCfg.samplingRate, 7265 mConfig.inputCfg.channels, 7266 mConfig.inputCfg.format); 7267 result.append(buffer); 7268 7269 result.append("\t\t- Output configuration:\n"); 7270 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7271 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7272 (uint32_t)mConfig.outputCfg.buffer.raw, 7273 mConfig.outputCfg.buffer.frameCount, 7274 mConfig.outputCfg.samplingRate, 7275 mConfig.outputCfg.channels, 7276 mConfig.outputCfg.format); 7277 result.append(buffer); 7278 7279 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7280 result.append(buffer); 7281 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7282 for (size_t i = 0; i < mHandles.size(); ++i) { 7283 sp<EffectHandle> handle = mHandles[i].promote(); 7284 if (handle != 0) { 7285 handle->dump(buffer, SIZE); 7286 result.append(buffer); 7287 } 7288 } 7289 7290 result.append("\n"); 7291 7292 write(fd, result.string(), result.length()); 7293 7294 if (locked) { 7295 mLock.unlock(); 7296 } 7297 7298 return NO_ERROR; 7299} 7300 7301// ---------------------------------------------------------------------------- 7302// EffectHandle implementation 7303// ---------------------------------------------------------------------------- 7304 7305#undef LOG_TAG 7306#define LOG_TAG "AudioFlinger::EffectHandle" 7307 7308AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7309 const sp<AudioFlinger::Client>& client, 7310 const sp<IEffectClient>& effectClient, 7311 int32_t priority) 7312 : BnEffect(), 7313 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7314 mPriority(priority), mHasControl(false), mEnabled(false) 7315{ 7316 ALOGV("constructor %p", this); 7317 7318 if (client == 0) { 7319 return; 7320 } 7321 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7322 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7323 if (mCblkMemory != 0) { 7324 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7325 7326 if (mCblk != NULL) { 7327 new(mCblk) effect_param_cblk_t(); 7328 mBuffer = (uint8_t *)mCblk + bufOffset; 7329 } 7330 } else { 7331 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7332 return; 7333 } 7334} 7335 7336AudioFlinger::EffectHandle::~EffectHandle() 7337{ 7338 ALOGV("Destructor %p", this); 7339 disconnect(false); 7340 ALOGV("Destructor DONE %p", this); 7341} 7342 7343status_t AudioFlinger::EffectHandle::enable() 7344{ 7345 ALOGV("enable %p", this); 7346 if (!mHasControl) return INVALID_OPERATION; 7347 if (mEffect == 0) return DEAD_OBJECT; 7348 7349 if (mEnabled) { 7350 return NO_ERROR; 7351 } 7352 7353 mEnabled = true; 7354 7355 sp<ThreadBase> thread = mEffect->thread().promote(); 7356 if (thread != 0) { 7357 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7358 } 7359 7360 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7361 if (mEffect->suspended()) { 7362 return NO_ERROR; 7363 } 7364 7365 status_t status = mEffect->setEnabled(true); 7366 if (status != NO_ERROR) { 7367 if (thread != 0) { 7368 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7369 } 7370 mEnabled = false; 7371 } 7372 return status; 7373} 7374 7375status_t AudioFlinger::EffectHandle::disable() 7376{ 7377 ALOGV("disable %p", this); 7378 if (!mHasControl) return INVALID_OPERATION; 7379 if (mEffect == 0) return DEAD_OBJECT; 7380 7381 if (!mEnabled) { 7382 return NO_ERROR; 7383 } 7384 mEnabled = false; 7385 7386 if (mEffect->suspended()) { 7387 return NO_ERROR; 7388 } 7389 7390 status_t status = mEffect->setEnabled(false); 7391 7392 sp<ThreadBase> thread = mEffect->thread().promote(); 7393 if (thread != 0) { 7394 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7395 } 7396 7397 return status; 7398} 7399 7400void AudioFlinger::EffectHandle::disconnect() 7401{ 7402 disconnect(true); 7403} 7404 7405void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7406{ 7407 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7408 if (mEffect == 0) { 7409 return; 7410 } 7411 mEffect->disconnect(this, unpinIfLast); 7412 7413 if (mHasControl && mEnabled) { 7414 sp<ThreadBase> thread = mEffect->thread().promote(); 7415 if (thread != 0) { 7416 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7417 } 7418 } 7419 7420 // release sp on module => module destructor can be called now 7421 mEffect.clear(); 7422 if (mClient != 0) { 7423 if (mCblk != NULL) { 7424 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7425 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7426 } 7427 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7428 // Client destructor must run with AudioFlinger mutex locked 7429 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7430 mClient.clear(); 7431 } 7432} 7433 7434status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7435 uint32_t cmdSize, 7436 void *pCmdData, 7437 uint32_t *replySize, 7438 void *pReplyData) 7439{ 7440// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7441// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7442 7443 // only get parameter command is permitted for applications not controlling the effect 7444 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7445 return INVALID_OPERATION; 7446 } 7447 if (mEffect == 0) return DEAD_OBJECT; 7448 if (mClient == 0) return INVALID_OPERATION; 7449 7450 // handle commands that are not forwarded transparently to effect engine 7451 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7452 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7453 // no risk to block the whole media server process or mixer threads is we are stuck here 7454 Mutex::Autolock _l(mCblk->lock); 7455 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7456 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7457 mCblk->serverIndex = 0; 7458 mCblk->clientIndex = 0; 7459 return BAD_VALUE; 7460 } 7461 status_t status = NO_ERROR; 7462 while (mCblk->serverIndex < mCblk->clientIndex) { 7463 int reply; 7464 uint32_t rsize = sizeof(int); 7465 int *p = (int *)(mBuffer + mCblk->serverIndex); 7466 int size = *p++; 7467 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7468 ALOGW("command(): invalid parameter block size"); 7469 break; 7470 } 7471 effect_param_t *param = (effect_param_t *)p; 7472 if (param->psize == 0 || param->vsize == 0) { 7473 ALOGW("command(): null parameter or value size"); 7474 mCblk->serverIndex += size; 7475 continue; 7476 } 7477 uint32_t psize = sizeof(effect_param_t) + 7478 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7479 param->vsize; 7480 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7481 psize, 7482 p, 7483 &rsize, 7484 &reply); 7485 // stop at first error encountered 7486 if (ret != NO_ERROR) { 7487 status = ret; 7488 *(int *)pReplyData = reply; 7489 break; 7490 } else if (reply != NO_ERROR) { 7491 *(int *)pReplyData = reply; 7492 break; 7493 } 7494 mCblk->serverIndex += size; 7495 } 7496 mCblk->serverIndex = 0; 7497 mCblk->clientIndex = 0; 7498 return status; 7499 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7500 *(int *)pReplyData = NO_ERROR; 7501 return enable(); 7502 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7503 *(int *)pReplyData = NO_ERROR; 7504 return disable(); 7505 } 7506 7507 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7508} 7509 7510void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7511{ 7512 ALOGV("setControl %p control %d", this, hasControl); 7513 7514 mHasControl = hasControl; 7515 mEnabled = enabled; 7516 7517 if (signal && mEffectClient != 0) { 7518 mEffectClient->controlStatusChanged(hasControl); 7519 } 7520} 7521 7522void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7523 uint32_t cmdSize, 7524 void *pCmdData, 7525 uint32_t replySize, 7526 void *pReplyData) 7527{ 7528 if (mEffectClient != 0) { 7529 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7530 } 7531} 7532 7533 7534 7535void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7536{ 7537 if (mEffectClient != 0) { 7538 mEffectClient->enableStatusChanged(enabled); 7539 } 7540} 7541 7542status_t AudioFlinger::EffectHandle::onTransact( 7543 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7544{ 7545 return BnEffect::onTransact(code, data, reply, flags); 7546} 7547 7548 7549void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7550{ 7551 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7552 7553 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7554 (mClient == 0) ? getpid_cached : mClient->pid(), 7555 mPriority, 7556 mHasControl, 7557 !locked, 7558 mCblk ? mCblk->clientIndex : 0, 7559 mCblk ? mCblk->serverIndex : 0 7560 ); 7561 7562 if (locked) { 7563 mCblk->lock.unlock(); 7564 } 7565} 7566 7567#undef LOG_TAG 7568#define LOG_TAG "AudioFlinger::EffectChain" 7569 7570AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7571 int sessionId) 7572 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7573 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7574 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7575{ 7576 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7577 if (thread == NULL) { 7578 return; 7579 } 7580 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7581 thread->frameCount(); 7582} 7583 7584AudioFlinger::EffectChain::~EffectChain() 7585{ 7586 if (mOwnInBuffer) { 7587 delete mInBuffer; 7588 } 7589 7590} 7591 7592// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7593sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7594{ 7595 size_t size = mEffects.size(); 7596 7597 for (size_t i = 0; i < size; i++) { 7598 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7599 return mEffects[i]; 7600 } 7601 } 7602 return 0; 7603} 7604 7605// getEffectFromId_l() must be called with ThreadBase::mLock held 7606sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7607{ 7608 size_t size = mEffects.size(); 7609 7610 for (size_t i = 0; i < size; i++) { 7611 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7612 if (id == 0 || mEffects[i]->id() == id) { 7613 return mEffects[i]; 7614 } 7615 } 7616 return 0; 7617} 7618 7619// getEffectFromType_l() must be called with ThreadBase::mLock held 7620sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7621 const effect_uuid_t *type) 7622{ 7623 size_t size = mEffects.size(); 7624 7625 for (size_t i = 0; i < size; i++) { 7626 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7627 return mEffects[i]; 7628 } 7629 } 7630 return 0; 7631} 7632 7633// Must be called with EffectChain::mLock locked 7634void AudioFlinger::EffectChain::process_l() 7635{ 7636 sp<ThreadBase> thread = mThread.promote(); 7637 if (thread == 0) { 7638 ALOGW("process_l(): cannot promote mixer thread"); 7639 return; 7640 } 7641 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7642 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7643 // always process effects unless no more tracks are on the session and the effect tail 7644 // has been rendered 7645 bool doProcess = true; 7646 if (!isGlobalSession) { 7647 bool tracksOnSession = (trackCnt() != 0); 7648 7649 if (!tracksOnSession && mTailBufferCount == 0) { 7650 doProcess = false; 7651 } 7652 7653 if (activeTrackCnt() == 0) { 7654 // if no track is active and the effect tail has not been rendered, 7655 // the input buffer must be cleared here as the mixer process will not do it 7656 if (tracksOnSession || mTailBufferCount > 0) { 7657 size_t numSamples = thread->frameCount() * thread->channelCount(); 7658 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7659 if (mTailBufferCount > 0) { 7660 mTailBufferCount--; 7661 } 7662 } 7663 } 7664 } 7665 7666 size_t size = mEffects.size(); 7667 if (doProcess) { 7668 for (size_t i = 0; i < size; i++) { 7669 mEffects[i]->process(); 7670 } 7671 } 7672 for (size_t i = 0; i < size; i++) { 7673 mEffects[i]->updateState(); 7674 } 7675} 7676 7677// addEffect_l() must be called with PlaybackThread::mLock held 7678status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7679{ 7680 effect_descriptor_t desc = effect->desc(); 7681 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7682 7683 Mutex::Autolock _l(mLock); 7684 effect->setChain(this); 7685 sp<ThreadBase> thread = mThread.promote(); 7686 if (thread == 0) { 7687 return NO_INIT; 7688 } 7689 effect->setThread(thread); 7690 7691 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7692 // Auxiliary effects are inserted at the beginning of mEffects vector as 7693 // they are processed first and accumulated in chain input buffer 7694 mEffects.insertAt(effect, 0); 7695 7696 // the input buffer for auxiliary effect contains mono samples in 7697 // 32 bit format. This is to avoid saturation in AudoMixer 7698 // accumulation stage. Saturation is done in EffectModule::process() before 7699 // calling the process in effect engine 7700 size_t numSamples = thread->frameCount(); 7701 int32_t *buffer = new int32_t[numSamples]; 7702 memset(buffer, 0, numSamples * sizeof(int32_t)); 7703 effect->setInBuffer((int16_t *)buffer); 7704 // auxiliary effects output samples to chain input buffer for further processing 7705 // by insert effects 7706 effect->setOutBuffer(mInBuffer); 7707 } else { 7708 // Insert effects are inserted at the end of mEffects vector as they are processed 7709 // after track and auxiliary effects. 7710 // Insert effect order as a function of indicated preference: 7711 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7712 // another effect is present 7713 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7714 // last effect claiming first position 7715 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7716 // first effect claiming last position 7717 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7718 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7719 // already present 7720 7721 size_t size = mEffects.size(); 7722 size_t idx_insert = size; 7723 ssize_t idx_insert_first = -1; 7724 ssize_t idx_insert_last = -1; 7725 7726 for (size_t i = 0; i < size; i++) { 7727 effect_descriptor_t d = mEffects[i]->desc(); 7728 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7729 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7730 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7731 // check invalid effect chaining combinations 7732 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7733 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7734 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7735 return INVALID_OPERATION; 7736 } 7737 // remember position of first insert effect and by default 7738 // select this as insert position for new effect 7739 if (idx_insert == size) { 7740 idx_insert = i; 7741 } 7742 // remember position of last insert effect claiming 7743 // first position 7744 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7745 idx_insert_first = i; 7746 } 7747 // remember position of first insert effect claiming 7748 // last position 7749 if (iPref == EFFECT_FLAG_INSERT_LAST && 7750 idx_insert_last == -1) { 7751 idx_insert_last = i; 7752 } 7753 } 7754 } 7755 7756 // modify idx_insert from first position if needed 7757 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7758 if (idx_insert_last != -1) { 7759 idx_insert = idx_insert_last; 7760 } else { 7761 idx_insert = size; 7762 } 7763 } else { 7764 if (idx_insert_first != -1) { 7765 idx_insert = idx_insert_first + 1; 7766 } 7767 } 7768 7769 // always read samples from chain input buffer 7770 effect->setInBuffer(mInBuffer); 7771 7772 // if last effect in the chain, output samples to chain 7773 // output buffer, otherwise to chain input buffer 7774 if (idx_insert == size) { 7775 if (idx_insert != 0) { 7776 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7777 mEffects[idx_insert-1]->configure(); 7778 } 7779 effect->setOutBuffer(mOutBuffer); 7780 } else { 7781 effect->setOutBuffer(mInBuffer); 7782 } 7783 mEffects.insertAt(effect, idx_insert); 7784 7785 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7786 } 7787 effect->configure(); 7788 return NO_ERROR; 7789} 7790 7791// removeEffect_l() must be called with PlaybackThread::mLock held 7792size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7793{ 7794 Mutex::Autolock _l(mLock); 7795 size_t size = mEffects.size(); 7796 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7797 7798 for (size_t i = 0; i < size; i++) { 7799 if (effect == mEffects[i]) { 7800 // calling stop here will remove pre-processing effect from the audio HAL. 7801 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7802 // the middle of a read from audio HAL 7803 if (mEffects[i]->state() == EffectModule::ACTIVE || 7804 mEffects[i]->state() == EffectModule::STOPPING) { 7805 mEffects[i]->stop(); 7806 } 7807 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7808 delete[] effect->inBuffer(); 7809 } else { 7810 if (i == size - 1 && i != 0) { 7811 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7812 mEffects[i - 1]->configure(); 7813 } 7814 } 7815 mEffects.removeAt(i); 7816 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7817 break; 7818 } 7819 } 7820 7821 return mEffects.size(); 7822} 7823 7824// setDevice_l() must be called with PlaybackThread::mLock held 7825void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7826{ 7827 size_t size = mEffects.size(); 7828 for (size_t i = 0; i < size; i++) { 7829 mEffects[i]->setDevice(device); 7830 } 7831} 7832 7833// setMode_l() must be called with PlaybackThread::mLock held 7834void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7835{ 7836 size_t size = mEffects.size(); 7837 for (size_t i = 0; i < size; i++) { 7838 mEffects[i]->setMode(mode); 7839 } 7840} 7841 7842// setVolume_l() must be called with PlaybackThread::mLock held 7843bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7844{ 7845 uint32_t newLeft = *left; 7846 uint32_t newRight = *right; 7847 bool hasControl = false; 7848 int ctrlIdx = -1; 7849 size_t size = mEffects.size(); 7850 7851 // first update volume controller 7852 for (size_t i = size; i > 0; i--) { 7853 if (mEffects[i - 1]->isProcessEnabled() && 7854 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7855 ctrlIdx = i - 1; 7856 hasControl = true; 7857 break; 7858 } 7859 } 7860 7861 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7862 if (hasControl) { 7863 *left = mNewLeftVolume; 7864 *right = mNewRightVolume; 7865 } 7866 return hasControl; 7867 } 7868 7869 mVolumeCtrlIdx = ctrlIdx; 7870 mLeftVolume = newLeft; 7871 mRightVolume = newRight; 7872 7873 // second get volume update from volume controller 7874 if (ctrlIdx >= 0) { 7875 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7876 mNewLeftVolume = newLeft; 7877 mNewRightVolume = newRight; 7878 } 7879 // then indicate volume to all other effects in chain. 7880 // Pass altered volume to effects before volume controller 7881 // and requested volume to effects after controller 7882 uint32_t lVol = newLeft; 7883 uint32_t rVol = newRight; 7884 7885 for (size_t i = 0; i < size; i++) { 7886 if ((int)i == ctrlIdx) continue; 7887 // this also works for ctrlIdx == -1 when there is no volume controller 7888 if ((int)i > ctrlIdx) { 7889 lVol = *left; 7890 rVol = *right; 7891 } 7892 mEffects[i]->setVolume(&lVol, &rVol, false); 7893 } 7894 *left = newLeft; 7895 *right = newRight; 7896 7897 return hasControl; 7898} 7899 7900status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7901{ 7902 const size_t SIZE = 256; 7903 char buffer[SIZE]; 7904 String8 result; 7905 7906 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7907 result.append(buffer); 7908 7909 bool locked = tryLock(mLock); 7910 // failed to lock - AudioFlinger is probably deadlocked 7911 if (!locked) { 7912 result.append("\tCould not lock mutex:\n"); 7913 } 7914 7915 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7916 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7917 mEffects.size(), 7918 (uint32_t)mInBuffer, 7919 (uint32_t)mOutBuffer, 7920 mActiveTrackCnt); 7921 result.append(buffer); 7922 write(fd, result.string(), result.size()); 7923 7924 for (size_t i = 0; i < mEffects.size(); ++i) { 7925 sp<EffectModule> effect = mEffects[i]; 7926 if (effect != 0) { 7927 effect->dump(fd, args); 7928 } 7929 } 7930 7931 if (locked) { 7932 mLock.unlock(); 7933 } 7934 7935 return NO_ERROR; 7936} 7937 7938// must be called with ThreadBase::mLock held 7939void AudioFlinger::EffectChain::setEffectSuspended_l( 7940 const effect_uuid_t *type, bool suspend) 7941{ 7942 sp<SuspendedEffectDesc> desc; 7943 // use effect type UUID timelow as key as there is no real risk of identical 7944 // timeLow fields among effect type UUIDs. 7945 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7946 if (suspend) { 7947 if (index >= 0) { 7948 desc = mSuspendedEffects.valueAt(index); 7949 } else { 7950 desc = new SuspendedEffectDesc(); 7951 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7952 mSuspendedEffects.add(type->timeLow, desc); 7953 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7954 } 7955 if (desc->mRefCount++ == 0) { 7956 sp<EffectModule> effect = getEffectIfEnabled(type); 7957 if (effect != 0) { 7958 desc->mEffect = effect; 7959 effect->setSuspended(true); 7960 effect->setEnabled(false); 7961 } 7962 } 7963 } else { 7964 if (index < 0) { 7965 return; 7966 } 7967 desc = mSuspendedEffects.valueAt(index); 7968 if (desc->mRefCount <= 0) { 7969 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7970 desc->mRefCount = 1; 7971 } 7972 if (--desc->mRefCount == 0) { 7973 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7974 if (desc->mEffect != 0) { 7975 sp<EffectModule> effect = desc->mEffect.promote(); 7976 if (effect != 0) { 7977 effect->setSuspended(false); 7978 sp<EffectHandle> handle = effect->controlHandle(); 7979 if (handle != 0) { 7980 effect->setEnabled(handle->enabled()); 7981 } 7982 } 7983 desc->mEffect.clear(); 7984 } 7985 mSuspendedEffects.removeItemsAt(index); 7986 } 7987 } 7988} 7989 7990// must be called with ThreadBase::mLock held 7991void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7992{ 7993 sp<SuspendedEffectDesc> desc; 7994 7995 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7996 if (suspend) { 7997 if (index >= 0) { 7998 desc = mSuspendedEffects.valueAt(index); 7999 } else { 8000 desc = new SuspendedEffectDesc(); 8001 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8002 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8003 } 8004 if (desc->mRefCount++ == 0) { 8005 Vector< sp<EffectModule> > effects; 8006 getSuspendEligibleEffects(effects); 8007 for (size_t i = 0; i < effects.size(); i++) { 8008 setEffectSuspended_l(&effects[i]->desc().type, true); 8009 } 8010 } 8011 } else { 8012 if (index < 0) { 8013 return; 8014 } 8015 desc = mSuspendedEffects.valueAt(index); 8016 if (desc->mRefCount <= 0) { 8017 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8018 desc->mRefCount = 1; 8019 } 8020 if (--desc->mRefCount == 0) { 8021 Vector<const effect_uuid_t *> types; 8022 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8023 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8024 continue; 8025 } 8026 types.add(&mSuspendedEffects.valueAt(i)->mType); 8027 } 8028 for (size_t i = 0; i < types.size(); i++) { 8029 setEffectSuspended_l(types[i], false); 8030 } 8031 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8032 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8033 } 8034 } 8035} 8036 8037 8038// The volume effect is used for automated tests only 8039#ifndef OPENSL_ES_H_ 8040static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8041 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8042const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8043#endif //OPENSL_ES_H_ 8044 8045bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8046{ 8047 // auxiliary effects and visualizer are never suspended on output mix 8048 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8049 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8050 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8051 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8052 return false; 8053 } 8054 return true; 8055} 8056 8057void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8058{ 8059 effects.clear(); 8060 for (size_t i = 0; i < mEffects.size(); i++) { 8061 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8062 effects.add(mEffects[i]); 8063 } 8064 } 8065} 8066 8067sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8068 const effect_uuid_t *type) 8069{ 8070 sp<EffectModule> effect = getEffectFromType_l(type); 8071 return effect != 0 && effect->isEnabled() ? effect : 0; 8072} 8073 8074void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8075 bool enabled) 8076{ 8077 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8078 if (enabled) { 8079 if (index < 0) { 8080 // if the effect is not suspend check if all effects are suspended 8081 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8082 if (index < 0) { 8083 return; 8084 } 8085 if (!isEffectEligibleForSuspend(effect->desc())) { 8086 return; 8087 } 8088 setEffectSuspended_l(&effect->desc().type, enabled); 8089 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8090 if (index < 0) { 8091 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8092 return; 8093 } 8094 } 8095 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8096 effect->desc().type.timeLow); 8097 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8098 // if effect is requested to suspended but was not yet enabled, supend it now. 8099 if (desc->mEffect == 0) { 8100 desc->mEffect = effect; 8101 effect->setEnabled(false); 8102 effect->setSuspended(true); 8103 } 8104 } else { 8105 if (index < 0) { 8106 return; 8107 } 8108 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8109 effect->desc().type.timeLow); 8110 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8111 desc->mEffect.clear(); 8112 effect->setSuspended(false); 8113 } 8114} 8115 8116#undef LOG_TAG 8117#define LOG_TAG "AudioFlinger" 8118 8119// ---------------------------------------------------------------------------- 8120 8121status_t AudioFlinger::onTransact( 8122 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8123{ 8124 return BnAudioFlinger::onTransact(code, data, reply, flags); 8125} 8126 8127}; // namespace android 8128