AudioFlinger.cpp revision 02fe1bf923bbe5789202dbd5810e2c04794562e6
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 <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_INIT; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_INIT; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 assert(NULL != mPrimaryHardwareDev); 727 assert(NULL != mPrimaryHardwareDev->get_master_volume); 728 729 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 730 mHardwareStatus = AUDIO_HW_IDLE; 731 return ret_val; 732 } 733 734 return mMasterVolume; 735} 736 737status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 738 audio_io_handle_t output) 739{ 740 // check calling permissions 741 if (!settingsAllowed()) { 742 return PERMISSION_DENIED; 743 } 744 745 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 746 ALOGE("setStreamVolume() invalid stream %d", stream); 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 PlaybackThread *thread = NULL; 752 if (output) { 753 thread = checkPlaybackThread_l(output); 754 if (thread == NULL) { 755 return BAD_VALUE; 756 } 757 } 758 759 mStreamTypes[stream].volume = value; 760 761 if (thread == NULL) { 762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 763 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 764 } 765 } else { 766 thread->setStreamVolume(stream, value); 767 } 768 769 return NO_ERROR; 770} 771 772status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 773{ 774 // check calling permissions 775 if (!settingsAllowed()) { 776 return PERMISSION_DENIED; 777 } 778 779 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 780 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 781 ALOGE("setStreamMute() invalid stream %d", stream); 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 mStreamTypes[stream].mute = muted; 787 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 789 790 return NO_ERROR; 791} 792 793float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 794{ 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 return 0.0f; 797 } 798 799 AutoMutex lock(mLock); 800 float volume; 801 if (output) { 802 PlaybackThread *thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return 0.0f; 805 } 806 volume = thread->streamVolume(stream); 807 } else { 808 volume = streamVolume_l(stream); 809 } 810 811 return volume; 812} 813 814bool AudioFlinger::streamMute(audio_stream_type_t stream) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return true; 818 } 819 820 AutoMutex lock(mLock); 821 return streamMute_l(stream); 822} 823 824status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 825{ 826 status_t result; 827 828 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 829 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 830 // check calling permissions 831 if (!settingsAllowed()) { 832 return PERMISSION_DENIED; 833 } 834 835 // ioHandle == 0 means the parameters are global to the audio hardware interface 836 if (ioHandle == 0) { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_SET_PARAMETER; 839 status_t final_result = NO_ERROR; 840 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 841 audio_hw_device_t *dev = mAudioHwDevs[i]; 842 result = dev->set_parameters(dev, keyValuePairs.string()); 843 final_result = result ?: final_result; 844 } 845 mHardwareStatus = AUDIO_HW_IDLE; 846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 847 AudioParameter param = AudioParameter(keyValuePairs); 848 String8 value; 849 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 850 Mutex::Autolock _l(mLock); 851 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 852 if (mBtNrecIsOff != btNrecIsOff) { 853 for (size_t i = 0; i < mRecordThreads.size(); i++) { 854 sp<RecordThread> thread = mRecordThreads.valueAt(i); 855 RecordThread::RecordTrack *track = thread->track(); 856 if (track != NULL) { 857 audio_devices_t device = (audio_devices_t)( 858 thread->device() & AUDIO_DEVICE_IN_ALL); 859 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 860 thread->setEffectSuspended(FX_IID_AEC, 861 suspend, 862 track->sessionId()); 863 thread->setEffectSuspended(FX_IID_NS, 864 suspend, 865 track->sessionId()); 866 } 867 } 868 mBtNrecIsOff = btNrecIsOff; 869 } 870 } 871 return final_result; 872 } 873 874 // hold a strong ref on thread in case closeOutput() or closeInput() is called 875 // and the thread is exited once the lock is released 876 sp<ThreadBase> thread; 877 { 878 Mutex::Autolock _l(mLock); 879 thread = checkPlaybackThread_l(ioHandle); 880 if (thread == NULL) { 881 thread = checkRecordThread_l(ioHandle); 882 } else if (thread == primaryPlaybackThread_l()) { 883 // indicate output device change to all input threads for pre processing 884 AudioParameter param = AudioParameter(keyValuePairs); 885 int value; 886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 887 for (size_t i = 0; i < mRecordThreads.size(); i++) { 888 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 889 } 890 } 891 } 892 } 893 if (thread != 0) { 894 return thread->setParameters(keyValuePairs); 895 } 896 return BAD_VALUE; 897} 898 899String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 900{ 901// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 902// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 903 904 if (ioHandle == 0) { 905 String8 out_s8; 906 907 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 908 audio_hw_device_t *dev = mAudioHwDevs[i]; 909 char *s = dev->get_parameters(dev, keys.string()); 910 out_s8 += String8(s ? s : ""); 911 free(s); 912 } 913 return out_s8; 914 } 915 916 Mutex::Autolock _l(mLock); 917 918 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 919 if (playbackThread != NULL) { 920 return playbackThread->getParameters(keys); 921 } 922 RecordThread *recordThread = checkRecordThread_l(ioHandle); 923 if (recordThread != NULL) { 924 return recordThread->getParameters(keys); 925 } 926 return String8(""); 927} 928 929size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 930{ 931 status_t ret = initCheck(); 932 if (ret != NO_ERROR) { 933 return 0; 934 } 935 936 AutoMutex lock(mHardwareLock); 937 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 938 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 939 mHardwareStatus = AUDIO_HW_IDLE; 940 return size; 941} 942 943unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 944{ 945 if (ioHandle == 0) { 946 return 0; 947 } 948 949 Mutex::Autolock _l(mLock); 950 951 RecordThread *recordThread = checkRecordThread_l(ioHandle); 952 if (recordThread != NULL) { 953 return recordThread->getInputFramesLost(); 954 } 955 return 0; 956} 957 958status_t AudioFlinger::setVoiceVolume(float value) 959{ 960 status_t ret = initCheck(); 961 if (ret != NO_ERROR) { 962 return ret; 963 } 964 965 // check calling permissions 966 if (!settingsAllowed()) { 967 return PERMISSION_DENIED; 968 } 969 970 AutoMutex lock(mHardwareLock); 971 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 972 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 973 mHardwareStatus = AUDIO_HW_IDLE; 974 975 return ret; 976} 977 978status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 979 audio_io_handle_t output) const 980{ 981 status_t status; 982 983 Mutex::Autolock _l(mLock); 984 985 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 986 if (playbackThread != NULL) { 987 return playbackThread->getRenderPosition(halFrames, dspFrames); 988 } 989 990 return BAD_VALUE; 991} 992 993void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 994{ 995 996 Mutex::Autolock _l(mLock); 997 998 pid_t pid = IPCThreadState::self()->getCallingPid(); 999 if (mNotificationClients.indexOfKey(pid) < 0) { 1000 sp<NotificationClient> notificationClient = new NotificationClient(this, 1001 client, 1002 pid); 1003 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1004 1005 mNotificationClients.add(pid, notificationClient); 1006 1007 sp<IBinder> binder = client->asBinder(); 1008 binder->linkToDeath(notificationClient); 1009 1010 // the config change is always sent from playback or record threads to avoid deadlock 1011 // with AudioSystem::gLock 1012 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1013 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1014 } 1015 1016 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1017 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1018 } 1019 } 1020} 1021 1022void AudioFlinger::removeNotificationClient(pid_t pid) 1023{ 1024 Mutex::Autolock _l(mLock); 1025 1026 ssize_t index = mNotificationClients.indexOfKey(pid); 1027 if (index >= 0) { 1028 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 1029 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 1030 mNotificationClients.removeItem(pid); 1031 } 1032 1033 ALOGV("%d died, releasing its sessions", pid); 1034 size_t num = mAudioSessionRefs.size(); 1035 bool removed = false; 1036 for (size_t i = 0; i< num; ) { 1037 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1038 ALOGV(" pid %d @ %d", ref->pid, i); 1039 if (ref->pid == pid) { 1040 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1041 mAudioSessionRefs.removeAt(i); 1042 delete ref; 1043 removed = true; 1044 num--; 1045 } else { 1046 i++; 1047 } 1048 } 1049 if (removed) { 1050 purgeStaleEffects_l(); 1051 } 1052} 1053 1054// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1055void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1056{ 1057 size_t size = mNotificationClients.size(); 1058 for (size_t i = 0; i < size; i++) { 1059 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1060 param2); 1061 } 1062} 1063 1064// removeClient_l() must be called with AudioFlinger::mLock held 1065void AudioFlinger::removeClient_l(pid_t pid) 1066{ 1067 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1068 mClients.removeItem(pid); 1069} 1070 1071 1072// ---------------------------------------------------------------------------- 1073 1074AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1075 uint32_t device, type_t type) 1076 : Thread(false), 1077 mType(type), 1078 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1079 // mChannelMask 1080 mChannelCount(0), 1081 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1082 mParamStatus(NO_ERROR), 1083 mStandby(false), mId(id), 1084 mDevice(device), 1085 mDeathRecipient(new PMDeathRecipient(this)) 1086{ 1087} 1088 1089AudioFlinger::ThreadBase::~ThreadBase() 1090{ 1091 mParamCond.broadcast(); 1092 // do not lock the mutex in destructor 1093 releaseWakeLock_l(); 1094 if (mPowerManager != 0) { 1095 sp<IBinder> binder = mPowerManager->asBinder(); 1096 binder->unlinkToDeath(mDeathRecipient); 1097 } 1098} 1099 1100void AudioFlinger::ThreadBase::exit() 1101{ 1102 ALOGV("ThreadBase::exit"); 1103 { 1104 // This lock prevents the following race in thread (uniprocessor for illustration): 1105 // if (!exitPending()) { 1106 // // context switch from here to exit() 1107 // // exit() calls requestExit(), what exitPending() observes 1108 // // exit() calls signal(), which is dropped since no waiters 1109 // // context switch back from exit() to here 1110 // mWaitWorkCV.wait(...); 1111 // // now thread is hung 1112 // } 1113 AutoMutex lock(mLock); 1114 requestExit(); 1115 mWaitWorkCV.signal(); 1116 } 1117 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1118 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1119 requestExitAndWait(); 1120} 1121 1122status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1123{ 1124 status_t status; 1125 1126 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1127 Mutex::Autolock _l(mLock); 1128 1129 mNewParameters.add(keyValuePairs); 1130 mWaitWorkCV.signal(); 1131 // wait condition with timeout in case the thread loop has exited 1132 // before the request could be processed 1133 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1134 status = mParamStatus; 1135 mWaitWorkCV.signal(); 1136 } else { 1137 status = TIMED_OUT; 1138 } 1139 return status; 1140} 1141 1142void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1143{ 1144 Mutex::Autolock _l(mLock); 1145 sendConfigEvent_l(event, param); 1146} 1147 1148// sendConfigEvent_l() must be called with ThreadBase::mLock held 1149void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1150{ 1151 ConfigEvent configEvent; 1152 configEvent.mEvent = event; 1153 configEvent.mParam = param; 1154 mConfigEvents.add(configEvent); 1155 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1156 mWaitWorkCV.signal(); 1157} 1158 1159void AudioFlinger::ThreadBase::processConfigEvents() 1160{ 1161 mLock.lock(); 1162 while(!mConfigEvents.isEmpty()) { 1163 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1164 ConfigEvent configEvent = mConfigEvents[0]; 1165 mConfigEvents.removeAt(0); 1166 // release mLock before locking AudioFlinger mLock: lock order is always 1167 // AudioFlinger then ThreadBase to avoid cross deadlock 1168 mLock.unlock(); 1169 mAudioFlinger->mLock.lock(); 1170 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1171 mAudioFlinger->mLock.unlock(); 1172 mLock.lock(); 1173 } 1174 mLock.unlock(); 1175} 1176 1177status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1178{ 1179 const size_t SIZE = 256; 1180 char buffer[SIZE]; 1181 String8 result; 1182 1183 bool locked = tryLock(mLock); 1184 if (!locked) { 1185 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1186 write(fd, buffer, strlen(buffer)); 1187 } 1188 1189 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1190 result.append(buffer); 1191 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1192 result.append(buffer); 1193 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1194 result.append(buffer); 1195 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1196 result.append(buffer); 1197 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1202 result.append(buffer); 1203 1204 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1205 result.append(buffer); 1206 result.append(" Index Command"); 1207 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1208 snprintf(buffer, SIZE, "\n %02d ", i); 1209 result.append(buffer); 1210 result.append(mNewParameters[i]); 1211 } 1212 1213 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1214 result.append(buffer); 1215 snprintf(buffer, SIZE, " Index event param\n"); 1216 result.append(buffer); 1217 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1218 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1219 result.append(buffer); 1220 } 1221 result.append("\n"); 1222 1223 write(fd, result.string(), result.size()); 1224 1225 if (locked) { 1226 mLock.unlock(); 1227 } 1228 return NO_ERROR; 1229} 1230 1231status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1232{ 1233 const size_t SIZE = 256; 1234 char buffer[SIZE]; 1235 String8 result; 1236 1237 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1238 write(fd, buffer, strlen(buffer)); 1239 1240 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1241 sp<EffectChain> chain = mEffectChains[i]; 1242 if (chain != 0) { 1243 chain->dump(fd, args); 1244 } 1245 } 1246 return NO_ERROR; 1247} 1248 1249void AudioFlinger::ThreadBase::acquireWakeLock() 1250{ 1251 Mutex::Autolock _l(mLock); 1252 acquireWakeLock_l(); 1253} 1254 1255void AudioFlinger::ThreadBase::acquireWakeLock_l() 1256{ 1257 if (mPowerManager == 0) { 1258 // use checkService() to avoid blocking if power service is not up yet 1259 sp<IBinder> binder = 1260 defaultServiceManager()->checkService(String16("power")); 1261 if (binder == 0) { 1262 ALOGW("Thread %s cannot connect to the power manager service", mName); 1263 } else { 1264 mPowerManager = interface_cast<IPowerManager>(binder); 1265 binder->linkToDeath(mDeathRecipient); 1266 } 1267 } 1268 if (mPowerManager != 0) { 1269 sp<IBinder> binder = new BBinder(); 1270 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1271 binder, 1272 String16(mName)); 1273 if (status == NO_ERROR) { 1274 mWakeLockToken = binder; 1275 } 1276 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1277 } 1278} 1279 1280void AudioFlinger::ThreadBase::releaseWakeLock() 1281{ 1282 Mutex::Autolock _l(mLock); 1283 releaseWakeLock_l(); 1284} 1285 1286void AudioFlinger::ThreadBase::releaseWakeLock_l() 1287{ 1288 if (mWakeLockToken != 0) { 1289 ALOGV("releaseWakeLock_l() %s", mName); 1290 if (mPowerManager != 0) { 1291 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1292 } 1293 mWakeLockToken.clear(); 1294 } 1295} 1296 1297void AudioFlinger::ThreadBase::clearPowerManager() 1298{ 1299 Mutex::Autolock _l(mLock); 1300 releaseWakeLock_l(); 1301 mPowerManager.clear(); 1302} 1303 1304void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1305{ 1306 sp<ThreadBase> thread = mThread.promote(); 1307 if (thread != 0) { 1308 thread->clearPowerManager(); 1309 } 1310 ALOGW("power manager service died !!!"); 1311} 1312 1313void AudioFlinger::ThreadBase::setEffectSuspended( 1314 const effect_uuid_t *type, bool suspend, int sessionId) 1315{ 1316 Mutex::Autolock _l(mLock); 1317 setEffectSuspended_l(type, suspend, sessionId); 1318} 1319 1320void AudioFlinger::ThreadBase::setEffectSuspended_l( 1321 const effect_uuid_t *type, bool suspend, int sessionId) 1322{ 1323 sp<EffectChain> chain = getEffectChain_l(sessionId); 1324 if (chain != 0) { 1325 if (type != NULL) { 1326 chain->setEffectSuspended_l(type, suspend); 1327 } else { 1328 chain->setEffectSuspendedAll_l(suspend); 1329 } 1330 } 1331 1332 updateSuspendedSessions_l(type, suspend, sessionId); 1333} 1334 1335void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1336{ 1337 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1338 if (index < 0) { 1339 return; 1340 } 1341 1342 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1343 mSuspendedSessions.editValueAt(index); 1344 1345 for (size_t i = 0; i < sessionEffects.size(); i++) { 1346 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1347 for (int j = 0; j < desc->mRefCount; j++) { 1348 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1349 chain->setEffectSuspendedAll_l(true); 1350 } else { 1351 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1352 desc->mType.timeLow); 1353 chain->setEffectSuspended_l(&desc->mType, true); 1354 } 1355 } 1356 } 1357} 1358 1359void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1360 bool suspend, 1361 int sessionId) 1362{ 1363 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1364 1365 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1366 1367 if (suspend) { 1368 if (index >= 0) { 1369 sessionEffects = mSuspendedSessions.editValueAt(index); 1370 } else { 1371 mSuspendedSessions.add(sessionId, sessionEffects); 1372 } 1373 } else { 1374 if (index < 0) { 1375 return; 1376 } 1377 sessionEffects = mSuspendedSessions.editValueAt(index); 1378 } 1379 1380 1381 int key = EffectChain::kKeyForSuspendAll; 1382 if (type != NULL) { 1383 key = type->timeLow; 1384 } 1385 index = sessionEffects.indexOfKey(key); 1386 1387 sp <SuspendedSessionDesc> desc; 1388 if (suspend) { 1389 if (index >= 0) { 1390 desc = sessionEffects.valueAt(index); 1391 } else { 1392 desc = new SuspendedSessionDesc(); 1393 if (type != NULL) { 1394 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1395 } 1396 sessionEffects.add(key, desc); 1397 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1398 } 1399 desc->mRefCount++; 1400 } else { 1401 if (index < 0) { 1402 return; 1403 } 1404 desc = sessionEffects.valueAt(index); 1405 if (--desc->mRefCount == 0) { 1406 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1407 sessionEffects.removeItemsAt(index); 1408 if (sessionEffects.isEmpty()) { 1409 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1410 sessionId); 1411 mSuspendedSessions.removeItem(sessionId); 1412 } 1413 } 1414 } 1415 if (!sessionEffects.isEmpty()) { 1416 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1417 } 1418} 1419 1420void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1421 bool enabled, 1422 int sessionId) 1423{ 1424 Mutex::Autolock _l(mLock); 1425 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1426} 1427 1428void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1429 bool enabled, 1430 int sessionId) 1431{ 1432 if (mType != RECORD) { 1433 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1434 // another session. This gives the priority to well behaved effect control panels 1435 // and applications not using global effects. 1436 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1437 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1438 } 1439 } 1440 1441 sp<EffectChain> chain = getEffectChain_l(sessionId); 1442 if (chain != 0) { 1443 chain->checkSuspendOnEffectEnabled(effect, enabled); 1444 } 1445} 1446 1447// ---------------------------------------------------------------------------- 1448 1449AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1450 AudioStreamOut* output, 1451 audio_io_handle_t id, 1452 uint32_t device, 1453 type_t type) 1454 : ThreadBase(audioFlinger, id, device, type), 1455 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1456 // Assumes constructor is called by AudioFlinger with it's mLock held, 1457 // but it would be safer to explicitly pass initial masterMute as parameter 1458 mMasterMute(audioFlinger->masterMute_l()), 1459 // mStreamTypes[] initialized in constructor body 1460 mOutput(output), 1461 // Assumes constructor is called by AudioFlinger with it's mLock held, 1462 // but it would be safer to explicitly pass initial masterVolume as parameter 1463 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1464 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1465{ 1466 snprintf(mName, kNameLength, "AudioOut_%d", id); 1467 1468 readOutputParameters(); 1469 1470 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1471 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1472 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1473 stream = (audio_stream_type_t) (stream + 1)) { 1474 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1475 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1476 // initialized by stream_type_t default constructor 1477 // mStreamTypes[stream].valid = true; 1478 } 1479 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1480 // because mAudioFlinger doesn't have one to copy from 1481} 1482 1483AudioFlinger::PlaybackThread::~PlaybackThread() 1484{ 1485 delete [] mMixBuffer; 1486} 1487 1488status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1489{ 1490 dumpInternals(fd, args); 1491 dumpTracks(fd, args); 1492 dumpEffectChains(fd, args); 1493 return NO_ERROR; 1494} 1495 1496status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1497{ 1498 const size_t SIZE = 256; 1499 char buffer[SIZE]; 1500 String8 result; 1501 1502 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1503 result.append(buffer); 1504 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1505 for (size_t i = 0; i < mTracks.size(); ++i) { 1506 sp<Track> track = mTracks[i]; 1507 if (track != 0) { 1508 track->dump(buffer, SIZE); 1509 result.append(buffer); 1510 } 1511 } 1512 1513 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1514 result.append(buffer); 1515 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1516 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1517 sp<Track> track = mActiveTracks[i].promote(); 1518 if (track != 0) { 1519 track->dump(buffer, SIZE); 1520 result.append(buffer); 1521 } 1522 } 1523 write(fd, result.string(), result.size()); 1524 return NO_ERROR; 1525} 1526 1527status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1528{ 1529 const size_t SIZE = 256; 1530 char buffer[SIZE]; 1531 String8 result; 1532 1533 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1534 result.append(buffer); 1535 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1536 result.append(buffer); 1537 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1538 result.append(buffer); 1539 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1540 result.append(buffer); 1541 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1542 result.append(buffer); 1543 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1544 result.append(buffer); 1545 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1546 result.append(buffer); 1547 write(fd, result.string(), result.size()); 1548 1549 dumpBase(fd, args); 1550 1551 return NO_ERROR; 1552} 1553 1554// Thread virtuals 1555status_t AudioFlinger::PlaybackThread::readyToRun() 1556{ 1557 status_t status = initCheck(); 1558 if (status == NO_ERROR) { 1559 ALOGI("AudioFlinger's thread %p ready to run", this); 1560 } else { 1561 ALOGE("No working audio driver found."); 1562 } 1563 return status; 1564} 1565 1566void AudioFlinger::PlaybackThread::onFirstRef() 1567{ 1568 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1569} 1570 1571// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1572sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1573 const sp<AudioFlinger::Client>& client, 1574 audio_stream_type_t streamType, 1575 uint32_t sampleRate, 1576 audio_format_t format, 1577 uint32_t channelMask, 1578 int frameCount, 1579 const sp<IMemory>& sharedBuffer, 1580 int sessionId, 1581 bool isTimed, 1582 status_t *status) 1583{ 1584 sp<Track> track; 1585 status_t lStatus; 1586 1587 if (mType == DIRECT) { 1588 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1589 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1590 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1591 "for output %p with format %d", 1592 sampleRate, format, channelMask, mOutput, mFormat); 1593 lStatus = BAD_VALUE; 1594 goto Exit; 1595 } 1596 } 1597 } else { 1598 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1599 if (sampleRate > mSampleRate*2) { 1600 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1601 lStatus = BAD_VALUE; 1602 goto Exit; 1603 } 1604 } 1605 1606 lStatus = initCheck(); 1607 if (lStatus != NO_ERROR) { 1608 ALOGE("Audio driver not initialized."); 1609 goto Exit; 1610 } 1611 1612 { // scope for mLock 1613 Mutex::Autolock _l(mLock); 1614 1615 // all tracks in same audio session must share the same routing strategy otherwise 1616 // conflicts will happen when tracks are moved from one output to another by audio policy 1617 // manager 1618 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1619 for (size_t i = 0; i < mTracks.size(); ++i) { 1620 sp<Track> t = mTracks[i]; 1621 if (t != 0) { 1622 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1623 if (sessionId == t->sessionId() && strategy != actual) { 1624 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1625 strategy, actual); 1626 lStatus = BAD_VALUE; 1627 goto Exit; 1628 } 1629 } 1630 } 1631 1632 if (!isTimed) { 1633 track = new Track(this, client, streamType, sampleRate, format, 1634 channelMask, frameCount, sharedBuffer, sessionId); 1635 } else { 1636 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1637 channelMask, frameCount, sharedBuffer, sessionId); 1638 } 1639 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1640 lStatus = NO_MEMORY; 1641 goto Exit; 1642 } 1643 mTracks.add(track); 1644 1645 sp<EffectChain> chain = getEffectChain_l(sessionId); 1646 if (chain != 0) { 1647 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1648 track->setMainBuffer(chain->inBuffer()); 1649 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1650 chain->incTrackCnt(); 1651 } 1652 1653 // invalidate track immediately if the stream type was moved to another thread since 1654 // createTrack() was called by the client process. 1655 if (!mStreamTypes[streamType].valid) { 1656 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1657 this, streamType); 1658 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1659 } 1660 } 1661 lStatus = NO_ERROR; 1662 1663Exit: 1664 if(status) { 1665 *status = lStatus; 1666 } 1667 return track; 1668} 1669 1670uint32_t AudioFlinger::PlaybackThread::latency() const 1671{ 1672 Mutex::Autolock _l(mLock); 1673 if (initCheck() == NO_ERROR) { 1674 return mOutput->stream->get_latency(mOutput->stream); 1675 } else { 1676 return 0; 1677 } 1678} 1679 1680void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1681{ 1682 Mutex::Autolock _l(mLock); 1683 mMasterVolume = value; 1684} 1685 1686void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1687{ 1688 Mutex::Autolock _l(mLock); 1689 setMasterMute_l(muted); 1690} 1691 1692void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1693{ 1694 Mutex::Autolock _l(mLock); 1695 mStreamTypes[stream].volume = value; 1696} 1697 1698void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1699{ 1700 Mutex::Autolock _l(mLock); 1701 mStreamTypes[stream].mute = muted; 1702} 1703 1704float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1705{ 1706 Mutex::Autolock _l(mLock); 1707 return mStreamTypes[stream].volume; 1708} 1709 1710// addTrack_l() must be called with ThreadBase::mLock held 1711status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1712{ 1713 status_t status = ALREADY_EXISTS; 1714 1715 // set retry count for buffer fill 1716 track->mRetryCount = kMaxTrackStartupRetries; 1717 if (mActiveTracks.indexOf(track) < 0) { 1718 // the track is newly added, make sure it fills up all its 1719 // buffers before playing. This is to ensure the client will 1720 // effectively get the latency it requested. 1721 track->mFillingUpStatus = Track::FS_FILLING; 1722 track->mResetDone = false; 1723 mActiveTracks.add(track); 1724 if (track->mainBuffer() != mMixBuffer) { 1725 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1726 if (chain != 0) { 1727 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1728 chain->incActiveTrackCnt(); 1729 } 1730 } 1731 1732 status = NO_ERROR; 1733 } 1734 1735 ALOGV("mWaitWorkCV.broadcast"); 1736 mWaitWorkCV.broadcast(); 1737 1738 return status; 1739} 1740 1741// destroyTrack_l() must be called with ThreadBase::mLock held 1742void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1743{ 1744 track->mState = TrackBase::TERMINATED; 1745 if (mActiveTracks.indexOf(track) < 0) { 1746 removeTrack_l(track); 1747 } 1748} 1749 1750void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1751{ 1752 mTracks.remove(track); 1753 deleteTrackName_l(track->name()); 1754 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1755 if (chain != 0) { 1756 chain->decTrackCnt(); 1757 } 1758} 1759 1760String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1761{ 1762 String8 out_s8 = String8(""); 1763 char *s; 1764 1765 Mutex::Autolock _l(mLock); 1766 if (initCheck() != NO_ERROR) { 1767 return out_s8; 1768 } 1769 1770 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1771 out_s8 = String8(s); 1772 free(s); 1773 return out_s8; 1774} 1775 1776// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1777void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1778 AudioSystem::OutputDescriptor desc; 1779 void *param2 = NULL; 1780 1781 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1782 1783 switch (event) { 1784 case AudioSystem::OUTPUT_OPENED: 1785 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1786 desc.channels = mChannelMask; 1787 desc.samplingRate = mSampleRate; 1788 desc.format = mFormat; 1789 desc.frameCount = mFrameCount; 1790 desc.latency = latency(); 1791 param2 = &desc; 1792 break; 1793 1794 case AudioSystem::STREAM_CONFIG_CHANGED: 1795 param2 = ¶m; 1796 case AudioSystem::OUTPUT_CLOSED: 1797 default: 1798 break; 1799 } 1800 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1801} 1802 1803void AudioFlinger::PlaybackThread::readOutputParameters() 1804{ 1805 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1806 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1807 mChannelCount = (uint16_t)popcount(mChannelMask); 1808 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1809 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1810 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1811 1812 // FIXME - Current mixer implementation only supports stereo output: Always 1813 // Allocate a stereo buffer even if HW output is mono. 1814 delete[] mMixBuffer; 1815 mMixBuffer = new int16_t[mFrameCount * 2]; 1816 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1817 1818 // force reconfiguration of effect chains and engines to take new buffer size and audio 1819 // parameters into account 1820 // Note that mLock is not held when readOutputParameters() is called from the constructor 1821 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1822 // matter. 1823 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1824 Vector< sp<EffectChain> > effectChains = mEffectChains; 1825 for (size_t i = 0; i < effectChains.size(); i ++) { 1826 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1827 } 1828} 1829 1830status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1831{ 1832 if (halFrames == NULL || dspFrames == NULL) { 1833 return BAD_VALUE; 1834 } 1835 Mutex::Autolock _l(mLock); 1836 if (initCheck() != NO_ERROR) { 1837 return INVALID_OPERATION; 1838 } 1839 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1840 1841 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1842} 1843 1844uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1845{ 1846 Mutex::Autolock _l(mLock); 1847 uint32_t result = 0; 1848 if (getEffectChain_l(sessionId) != 0) { 1849 result = EFFECT_SESSION; 1850 } 1851 1852 for (size_t i = 0; i < mTracks.size(); ++i) { 1853 sp<Track> track = mTracks[i]; 1854 if (sessionId == track->sessionId() && 1855 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1856 result |= TRACK_SESSION; 1857 break; 1858 } 1859 } 1860 1861 return result; 1862} 1863 1864uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1865{ 1866 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1867 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1868 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1869 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1870 } 1871 for (size_t i = 0; i < mTracks.size(); i++) { 1872 sp<Track> track = mTracks[i]; 1873 if (sessionId == track->sessionId() && 1874 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1875 return AudioSystem::getStrategyForStream(track->streamType()); 1876 } 1877 } 1878 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1879} 1880 1881 1882AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1883{ 1884 Mutex::Autolock _l(mLock); 1885 return mOutput; 1886} 1887 1888AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1889{ 1890 Mutex::Autolock _l(mLock); 1891 AudioStreamOut *output = mOutput; 1892 mOutput = NULL; 1893 return output; 1894} 1895 1896// this method must always be called either with ThreadBase mLock held or inside the thread loop 1897audio_stream_t* AudioFlinger::PlaybackThread::stream() 1898{ 1899 if (mOutput == NULL) { 1900 return NULL; 1901 } 1902 return &mOutput->stream->common; 1903} 1904 1905uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1906{ 1907 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1908 // decoding and transfer time. So sleeping for half of the latency would likely cause 1909 // underruns 1910 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1911 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1912 } else { 1913 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1914 } 1915} 1916 1917// ---------------------------------------------------------------------------- 1918 1919AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1920 audio_io_handle_t id, uint32_t device, type_t type) 1921 : PlaybackThread(audioFlinger, output, id, device, type), 1922 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1923 mPrevMixerStatus(MIXER_IDLE) 1924{ 1925 // FIXME - Current mixer implementation only supports stereo output 1926 if (mChannelCount == 1) { 1927 ALOGE("Invalid audio hardware channel count"); 1928 } 1929} 1930 1931AudioFlinger::MixerThread::~MixerThread() 1932{ 1933 delete mAudioMixer; 1934} 1935 1936class CpuStats { 1937public: 1938 void sample(); 1939#ifdef DEBUG_CPU_USAGE 1940private: 1941 ThreadCpuUsage mCpu; 1942#endif 1943}; 1944 1945void CpuStats::sample() { 1946#ifdef DEBUG_CPU_USAGE 1947 const CentralTendencyStatistics& stats = mCpu.statistics(); 1948 mCpu.sampleAndEnable(); 1949 unsigned n = stats.n(); 1950 // mCpu.elapsed() is expensive, so don't call it every loop 1951 if ((n & 127) == 1) { 1952 long long elapsed = mCpu.elapsed(); 1953 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1954 double perLoop = elapsed / (double) n; 1955 double perLoop100 = perLoop * 0.01; 1956 double mean = stats.mean(); 1957 double stddev = stats.stddev(); 1958 double minimum = stats.minimum(); 1959 double maximum = stats.maximum(); 1960 mCpu.resetStatistics(); 1961 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1962 elapsed * .000000001, n, perLoop * .000001, 1963 mean * .001, 1964 stddev * .001, 1965 minimum * .001, 1966 maximum * .001, 1967 mean / perLoop100, 1968 stddev / perLoop100, 1969 minimum / perLoop100, 1970 maximum / perLoop100); 1971 } 1972 } 1973#endif 1974}; 1975 1976void AudioFlinger::PlaybackThread::checkSilentMode_l() 1977{ 1978 if (!mMasterMute) { 1979 char value[PROPERTY_VALUE_MAX]; 1980 if (property_get("ro.audio.silent", value, "0") > 0) { 1981 char *endptr; 1982 unsigned long ul = strtoul(value, &endptr, 0); 1983 if (*endptr == '\0' && ul != 0) { 1984 ALOGD("Silence is golden"); 1985 // The setprop command will not allow a property to be changed after 1986 // the first time it is set, so we don't have to worry about un-muting. 1987 setMasterMute_l(true); 1988 } 1989 } 1990 } 1991} 1992 1993bool AudioFlinger::MixerThread::threadLoop() 1994{ 1995 Vector< sp<Track> > tracksToRemove; 1996 mixer_state mixerStatus = MIXER_IDLE; 1997 nsecs_t standbyTime = systemTime(); 1998 size_t mixBufferSize = mFrameCount * mFrameSize; 1999 // FIXME: Relaxed timing because of a certain device that can't meet latency 2000 // Should be reduced to 2x after the vendor fixes the driver issue 2001 // increase threshold again due to low power audio mode. The way this warning threshold is 2002 // calculated and its usefulness should be reconsidered anyway. 2003 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2004 nsecs_t lastWarning = 0; 2005 bool longStandbyExit = false; 2006 uint32_t activeSleepTime = activeSleepTimeUs(); 2007 uint32_t idleSleepTime = idleSleepTimeUs(); 2008 uint32_t sleepTime = idleSleepTime; 2009 uint32_t sleepTimeShift = 0; 2010 Vector< sp<EffectChain> > effectChains; 2011 CpuStats cpuStats; 2012 2013 acquireWakeLock(); 2014 2015 while (!exitPending()) 2016 { 2017 cpuStats.sample(); 2018 processConfigEvents(); 2019 2020 mixerStatus = MIXER_IDLE; 2021 { // scope for mLock 2022 2023 Mutex::Autolock _l(mLock); 2024 2025 if (checkForNewParameters_l()) { 2026 mixBufferSize = mFrameCount * mFrameSize; 2027 // FIXME: Relaxed timing because of a certain device that can't meet latency 2028 // Should be reduced to 2x after the vendor fixes the driver issue 2029 // increase threshold again due to low power audio mode. The way this warning 2030 // threshold is calculated and its usefulness should be reconsidered anyway. 2031 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2032 activeSleepTime = activeSleepTimeUs(); 2033 idleSleepTime = idleSleepTimeUs(); 2034 } 2035 2036 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2037 2038 // put audio hardware into standby after short delay 2039 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2040 mSuspended)) { 2041 if (!mStandby) { 2042 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2043 mOutput->stream->common.standby(&mOutput->stream->common); 2044 mStandby = true; 2045 mBytesWritten = 0; 2046 } 2047 2048 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2049 // we're about to wait, flush the binder command buffer 2050 IPCThreadState::self()->flushCommands(); 2051 2052 if (exitPending()) break; 2053 2054 releaseWakeLock_l(); 2055 // wait until we have something to do... 2056 ALOGV("MixerThread %p TID %d going to sleep", this, gettid()); 2057 mWaitWorkCV.wait(mLock); 2058 ALOGV("MixerThread %p TID %d waking up", this, gettid()); 2059 acquireWakeLock_l(); 2060 2061 mPrevMixerStatus = MIXER_IDLE; 2062 checkSilentMode_l(); 2063 2064 standbyTime = systemTime() + mStandbyTimeInNsecs; 2065 sleepTime = idleSleepTime; 2066 sleepTimeShift = 0; 2067 continue; 2068 } 2069 } 2070 2071 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2072 2073 // prevent any changes in effect chain list and in each effect chain 2074 // during mixing and effect process as the audio buffers could be deleted 2075 // or modified if an effect is created or deleted 2076 lockEffectChains_l(effectChains); 2077 } 2078 2079 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2080 // obtain the presentation timestamp of the next output buffer 2081 int64_t pts; 2082 status_t status = INVALID_OPERATION; 2083 2084 if (NULL != mOutput->stream->get_next_write_timestamp) { 2085 status = mOutput->stream->get_next_write_timestamp( 2086 mOutput->stream, &pts); 2087 } 2088 2089 if (status != NO_ERROR) { 2090 pts = AudioBufferProvider::kInvalidPTS; 2091 } 2092 2093 // mix buffers... 2094 mAudioMixer->process(pts); 2095 // increase sleep time progressively when application underrun condition clears. 2096 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2097 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2098 // such that we would underrun the audio HAL. 2099 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2100 sleepTimeShift--; 2101 } 2102 sleepTime = 0; 2103 standbyTime = systemTime() + mStandbyTimeInNsecs; 2104 //TODO: delay standby when effects have a tail 2105 } else { 2106 // If no tracks are ready, sleep once for the duration of an output 2107 // buffer size, then write 0s to the output 2108 if (sleepTime == 0) { 2109 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2110 sleepTime = activeSleepTime >> sleepTimeShift; 2111 if (sleepTime < kMinThreadSleepTimeUs) { 2112 sleepTime = kMinThreadSleepTimeUs; 2113 } 2114 // reduce sleep time in case of consecutive application underruns to avoid 2115 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2116 // duration we would end up writing less data than needed by the audio HAL if 2117 // the condition persists. 2118 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2119 sleepTimeShift++; 2120 } 2121 } else { 2122 sleepTime = idleSleepTime; 2123 } 2124 } else if (mBytesWritten != 0 || 2125 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2126 memset (mMixBuffer, 0, mixBufferSize); 2127 sleepTime = 0; 2128 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2129 } 2130 // TODO add standby time extension fct of effect tail 2131 } 2132 2133 if (mSuspended) { 2134 sleepTime = suspendSleepTimeUs(); 2135 } 2136 // sleepTime == 0 means we must write to audio hardware 2137 if (sleepTime == 0) { 2138 for (size_t i = 0; i < effectChains.size(); i ++) { 2139 effectChains[i]->process_l(); 2140 } 2141 // enable changes in effect chain 2142 unlockEffectChains(effectChains); 2143 mLastWriteTime = systemTime(); 2144 mInWrite = true; 2145 mBytesWritten += mixBufferSize; 2146 2147 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2148 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2149 mNumWrites++; 2150 mInWrite = false; 2151 nsecs_t now = systemTime(); 2152 nsecs_t delta = now - mLastWriteTime; 2153 if (!mStandby && delta > maxPeriod) { 2154 mNumDelayedWrites++; 2155 if ((now - lastWarning) > kWarningThrottleNs) { 2156 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2157 ns2ms(delta), mNumDelayedWrites, this); 2158 lastWarning = now; 2159 } 2160 if (mStandby) { 2161 longStandbyExit = true; 2162 } 2163 } 2164 mStandby = false; 2165 } else { 2166 // enable changes in effect chain 2167 unlockEffectChains(effectChains); 2168 usleep(sleepTime); 2169 } 2170 2171 // finally let go of all our tracks, without the lock held 2172 // since we can't guarantee the destructors won't acquire that 2173 // same lock. 2174 tracksToRemove.clear(); 2175 2176 // Effect chains will be actually deleted here if they were removed from 2177 // mEffectChains list during mixing or effects processing 2178 effectChains.clear(); 2179 } 2180 2181 if (!mStandby) { 2182 mOutput->stream->common.standby(&mOutput->stream->common); 2183 } 2184 2185 releaseWakeLock(); 2186 2187 ALOGV("MixerThread %p exiting", this); 2188 return false; 2189} 2190 2191// prepareTracks_l() must be called with ThreadBase::mLock held 2192AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2193 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2194{ 2195 2196 mixer_state mixerStatus = MIXER_IDLE; 2197 // find out which tracks need to be processed 2198 size_t count = activeTracks.size(); 2199 size_t mixedTracks = 0; 2200 size_t tracksWithEffect = 0; 2201 2202 float masterVolume = mMasterVolume; 2203 bool masterMute = mMasterMute; 2204 2205 if (masterMute) { 2206 masterVolume = 0; 2207 } 2208 // Delegate master volume control to effect in output mix effect chain if needed 2209 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2210 if (chain != 0) { 2211 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2212 chain->setVolume_l(&v, &v); 2213 masterVolume = (float)((v + (1 << 23)) >> 24); 2214 chain.clear(); 2215 } 2216 2217 for (size_t i=0 ; i<count ; i++) { 2218 sp<Track> t = activeTracks[i].promote(); 2219 if (t == 0) continue; 2220 2221 // this const just means the local variable doesn't change 2222 Track* const track = t.get(); 2223 audio_track_cblk_t* cblk = track->cblk(); 2224 2225 // The first time a track is added we wait 2226 // for all its buffers to be filled before processing it 2227 int name = track->name(); 2228 // make sure that we have enough frames to mix one full buffer. 2229 // enforce this condition only once to enable draining the buffer in case the client 2230 // app does not call stop() and relies on underrun to stop: 2231 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2232 // during last round 2233 uint32_t minFrames = 1; 2234 if (!track->isStopped() && !track->isPausing() && 2235 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2236 if (t->sampleRate() == (int)mSampleRate) { 2237 minFrames = mFrameCount; 2238 } else { 2239 // +1 for rounding and +1 for additional sample needed for interpolation 2240 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2241 // add frames already consumed but not yet released by the resampler 2242 // because cblk->framesReady() will include these frames 2243 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2244 // the minimum track buffer size is normally twice the number of frames necessary 2245 // to fill one buffer and the resampler should not leave more than one buffer worth 2246 // of unreleased frames after each pass, but just in case... 2247 ALOG_ASSERT(minFrames <= cblk->frameCount); 2248 } 2249 } 2250 if ((track->framesReady() >= minFrames) && track->isReady() && 2251 !track->isPaused() && !track->isTerminated()) 2252 { 2253 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2254 2255 mixedTracks++; 2256 2257 // track->mainBuffer() != mMixBuffer means there is an effect chain 2258 // connected to the track 2259 chain.clear(); 2260 if (track->mainBuffer() != mMixBuffer) { 2261 chain = getEffectChain_l(track->sessionId()); 2262 // Delegate volume control to effect in track effect chain if needed 2263 if (chain != 0) { 2264 tracksWithEffect++; 2265 } else { 2266 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2267 name, track->sessionId()); 2268 } 2269 } 2270 2271 2272 int param = AudioMixer::VOLUME; 2273 if (track->mFillingUpStatus == Track::FS_FILLED) { 2274 // no ramp for the first volume setting 2275 track->mFillingUpStatus = Track::FS_ACTIVE; 2276 if (track->mState == TrackBase::RESUMING) { 2277 track->mState = TrackBase::ACTIVE; 2278 param = AudioMixer::RAMP_VOLUME; 2279 } 2280 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2281 } else if (cblk->server != 0) { 2282 // If the track is stopped before the first frame was mixed, 2283 // do not apply ramp 2284 param = AudioMixer::RAMP_VOLUME; 2285 } 2286 2287 // compute volume for this track 2288 uint32_t vl, vr, va; 2289 if (track->isMuted() || track->isPausing() || 2290 mStreamTypes[track->streamType()].mute) { 2291 vl = vr = va = 0; 2292 if (track->isPausing()) { 2293 track->setPaused(); 2294 } 2295 } else { 2296 2297 // read original volumes with volume control 2298 float typeVolume = mStreamTypes[track->streamType()].volume; 2299 float v = masterVolume * typeVolume; 2300 uint32_t vlr = cblk->getVolumeLR(); 2301 vl = vlr & 0xFFFF; 2302 vr = vlr >> 16; 2303 // track volumes come from shared memory, so can't be trusted and must be clamped 2304 if (vl > MAX_GAIN_INT) { 2305 ALOGV("Track left volume out of range: %04X", vl); 2306 vl = MAX_GAIN_INT; 2307 } 2308 if (vr > MAX_GAIN_INT) { 2309 ALOGV("Track right volume out of range: %04X", vr); 2310 vr = MAX_GAIN_INT; 2311 } 2312 // now apply the master volume and stream type volume 2313 vl = (uint32_t)(v * vl) << 12; 2314 vr = (uint32_t)(v * vr) << 12; 2315 // assuming master volume and stream type volume each go up to 1.0, 2316 // vl and vr are now in 8.24 format 2317 2318 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2319 // send level comes from shared memory and so may be corrupt 2320 if (sendLevel > MAX_GAIN_INT) { 2321 ALOGV("Track send level out of range: %04X", sendLevel); 2322 sendLevel = MAX_GAIN_INT; 2323 } 2324 va = (uint32_t)(v * sendLevel); 2325 } 2326 // Delegate volume control to effect in track effect chain if needed 2327 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2328 // Do not ramp volume if volume is controlled by effect 2329 param = AudioMixer::VOLUME; 2330 track->mHasVolumeController = true; 2331 } else { 2332 // force no volume ramp when volume controller was just disabled or removed 2333 // from effect chain to avoid volume spike 2334 if (track->mHasVolumeController) { 2335 param = AudioMixer::VOLUME; 2336 } 2337 track->mHasVolumeController = false; 2338 } 2339 2340 // Convert volumes from 8.24 to 4.12 format 2341 // This additional clamping is needed in case chain->setVolume_l() overshot 2342 vl = (vl + (1 << 11)) >> 12; 2343 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2344 vr = (vr + (1 << 11)) >> 12; 2345 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2346 2347 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2348 2349 // XXX: these things DON'T need to be done each time 2350 mAudioMixer->setBufferProvider(name, track); 2351 mAudioMixer->enable(name); 2352 2353 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2354 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2355 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2356 mAudioMixer->setParameter( 2357 name, 2358 AudioMixer::TRACK, 2359 AudioMixer::FORMAT, (void *)track->format()); 2360 mAudioMixer->setParameter( 2361 name, 2362 AudioMixer::TRACK, 2363 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2364 mAudioMixer->setParameter( 2365 name, 2366 AudioMixer::RESAMPLE, 2367 AudioMixer::SAMPLE_RATE, 2368 (void *)(cblk->sampleRate)); 2369 mAudioMixer->setParameter( 2370 name, 2371 AudioMixer::TRACK, 2372 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2373 mAudioMixer->setParameter( 2374 name, 2375 AudioMixer::TRACK, 2376 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2377 2378 // reset retry count 2379 track->mRetryCount = kMaxTrackRetries; 2380 // If one track is ready, set the mixer ready if: 2381 // - the mixer was not ready during previous round OR 2382 // - no other track is not ready 2383 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2384 mixerStatus != MIXER_TRACKS_ENABLED) { 2385 mixerStatus = MIXER_TRACKS_READY; 2386 } 2387 } else { 2388 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2389 if (track->isStopped()) { 2390 track->reset(); 2391 } 2392 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2393 // We have consumed all the buffers of this track. 2394 // Remove it from the list of active tracks. 2395 tracksToRemove->add(track); 2396 } else { 2397 // No buffers for this track. Give it a few chances to 2398 // fill a buffer, then remove it from active list. 2399 if (--(track->mRetryCount) <= 0) { 2400 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2401 tracksToRemove->add(track); 2402 // indicate to client process that the track was disabled because of underrun 2403 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2404 // If one track is not ready, mark the mixer also not ready if: 2405 // - the mixer was ready during previous round OR 2406 // - no other track is ready 2407 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2408 mixerStatus != MIXER_TRACKS_READY) { 2409 mixerStatus = MIXER_TRACKS_ENABLED; 2410 } 2411 } 2412 mAudioMixer->disable(name); 2413 } 2414 } 2415 2416 // remove all the tracks that need to be... 2417 count = tracksToRemove->size(); 2418 if (CC_UNLIKELY(count)) { 2419 for (size_t i=0 ; i<count ; i++) { 2420 const sp<Track>& track = tracksToRemove->itemAt(i); 2421 mActiveTracks.remove(track); 2422 if (track->mainBuffer() != mMixBuffer) { 2423 chain = getEffectChain_l(track->sessionId()); 2424 if (chain != 0) { 2425 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2426 chain->decActiveTrackCnt(); 2427 } 2428 } 2429 if (track->isTerminated()) { 2430 removeTrack_l(track); 2431 } 2432 } 2433 } 2434 2435 // mix buffer must be cleared if all tracks are connected to an 2436 // effect chain as in this case the mixer will not write to 2437 // mix buffer and track effects will accumulate into it 2438 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2439 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2440 } 2441 2442 mPrevMixerStatus = mixerStatus; 2443 return mixerStatus; 2444} 2445 2446void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2447{ 2448 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2449 this, streamType, mTracks.size()); 2450 Mutex::Autolock _l(mLock); 2451 2452 size_t size = mTracks.size(); 2453 for (size_t i = 0; i < size; i++) { 2454 sp<Track> t = mTracks[i]; 2455 if (t->streamType() == streamType) { 2456 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2457 t->mCblk->cv.signal(); 2458 } 2459 } 2460} 2461 2462void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2463{ 2464 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2465 this, streamType, valid); 2466 Mutex::Autolock _l(mLock); 2467 2468 mStreamTypes[streamType].valid = valid; 2469} 2470 2471// getTrackName_l() must be called with ThreadBase::mLock held 2472int AudioFlinger::MixerThread::getTrackName_l() 2473{ 2474 return mAudioMixer->getTrackName(); 2475} 2476 2477// deleteTrackName_l() must be called with ThreadBase::mLock held 2478void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2479{ 2480 ALOGV("remove track (%d) and delete from mixer", name); 2481 mAudioMixer->deleteTrackName(name); 2482} 2483 2484// checkForNewParameters_l() must be called with ThreadBase::mLock held 2485bool AudioFlinger::MixerThread::checkForNewParameters_l() 2486{ 2487 bool reconfig = false; 2488 2489 while (!mNewParameters.isEmpty()) { 2490 status_t status = NO_ERROR; 2491 String8 keyValuePair = mNewParameters[0]; 2492 AudioParameter param = AudioParameter(keyValuePair); 2493 int value; 2494 2495 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2496 reconfig = true; 2497 } 2498 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2499 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2500 status = BAD_VALUE; 2501 } else { 2502 reconfig = true; 2503 } 2504 } 2505 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2506 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2507 status = BAD_VALUE; 2508 } else { 2509 reconfig = true; 2510 } 2511 } 2512 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2513 // do not accept frame count changes if tracks are open as the track buffer 2514 // size depends on frame count and correct behavior would not be guaranteed 2515 // if frame count is changed after track creation 2516 if (!mTracks.isEmpty()) { 2517 status = INVALID_OPERATION; 2518 } else { 2519 reconfig = true; 2520 } 2521 } 2522 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2523 // when changing the audio output device, call addBatteryData to notify 2524 // the change 2525 if ((int)mDevice != value) { 2526 uint32_t params = 0; 2527 // check whether speaker is on 2528 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2529 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2530 } 2531 2532 int deviceWithoutSpeaker 2533 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2534 // check if any other device (except speaker) is on 2535 if (value & deviceWithoutSpeaker ) { 2536 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2537 } 2538 2539 if (params != 0) { 2540 addBatteryData(params); 2541 } 2542 } 2543 2544 // forward device change to effects that have requested to be 2545 // aware of attached audio device. 2546 mDevice = (uint32_t)value; 2547 for (size_t i = 0; i < mEffectChains.size(); i++) { 2548 mEffectChains[i]->setDevice_l(mDevice); 2549 } 2550 } 2551 2552 if (status == NO_ERROR) { 2553 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2554 keyValuePair.string()); 2555 if (!mStandby && status == INVALID_OPERATION) { 2556 mOutput->stream->common.standby(&mOutput->stream->common); 2557 mStandby = true; 2558 mBytesWritten = 0; 2559 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2560 keyValuePair.string()); 2561 } 2562 if (status == NO_ERROR && reconfig) { 2563 delete mAudioMixer; 2564 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2565 mAudioMixer = NULL; 2566 readOutputParameters(); 2567 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2568 for (size_t i = 0; i < mTracks.size() ; i++) { 2569 int name = getTrackName_l(); 2570 if (name < 0) break; 2571 mTracks[i]->mName = name; 2572 // limit track sample rate to 2 x new output sample rate 2573 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2574 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2575 } 2576 } 2577 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2578 } 2579 } 2580 2581 mNewParameters.removeAt(0); 2582 2583 mParamStatus = status; 2584 mParamCond.signal(); 2585 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2586 // already timed out waiting for the status and will never signal the condition. 2587 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2588 } 2589 return reconfig; 2590} 2591 2592status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2593{ 2594 const size_t SIZE = 256; 2595 char buffer[SIZE]; 2596 String8 result; 2597 2598 PlaybackThread::dumpInternals(fd, args); 2599 2600 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2601 result.append(buffer); 2602 write(fd, result.string(), result.size()); 2603 return NO_ERROR; 2604} 2605 2606uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2607{ 2608 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2609} 2610 2611uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2612{ 2613 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2614} 2615 2616// ---------------------------------------------------------------------------- 2617AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2618 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2619 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2620 // mLeftVolFloat, mRightVolFloat 2621 // mLeftVolShort, mRightVolShort 2622{ 2623} 2624 2625AudioFlinger::DirectOutputThread::~DirectOutputThread() 2626{ 2627} 2628 2629void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2630{ 2631 // Do not apply volume on compressed audio 2632 if (!audio_is_linear_pcm(mFormat)) { 2633 return; 2634 } 2635 2636 // convert to signed 16 bit before volume calculation 2637 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2638 size_t count = mFrameCount * mChannelCount; 2639 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2640 int16_t *dst = mMixBuffer + count-1; 2641 while(count--) { 2642 *dst-- = (int16_t)(*src--^0x80) << 8; 2643 } 2644 } 2645 2646 size_t frameCount = mFrameCount; 2647 int16_t *out = mMixBuffer; 2648 if (ramp) { 2649 if (mChannelCount == 1) { 2650 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2651 int32_t vlInc = d / (int32_t)frameCount; 2652 int32_t vl = ((int32_t)mLeftVolShort << 16); 2653 do { 2654 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2655 out++; 2656 vl += vlInc; 2657 } while (--frameCount); 2658 2659 } else { 2660 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2661 int32_t vlInc = d / (int32_t)frameCount; 2662 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2663 int32_t vrInc = d / (int32_t)frameCount; 2664 int32_t vl = ((int32_t)mLeftVolShort << 16); 2665 int32_t vr = ((int32_t)mRightVolShort << 16); 2666 do { 2667 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2668 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2669 out += 2; 2670 vl += vlInc; 2671 vr += vrInc; 2672 } while (--frameCount); 2673 } 2674 } else { 2675 if (mChannelCount == 1) { 2676 do { 2677 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2678 out++; 2679 } while (--frameCount); 2680 } else { 2681 do { 2682 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2683 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2684 out += 2; 2685 } while (--frameCount); 2686 } 2687 } 2688 2689 // convert back to unsigned 8 bit after volume calculation 2690 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2691 size_t count = mFrameCount * mChannelCount; 2692 int16_t *src = mMixBuffer; 2693 uint8_t *dst = (uint8_t *)mMixBuffer; 2694 while(count--) { 2695 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2696 } 2697 } 2698 2699 mLeftVolShort = leftVol; 2700 mRightVolShort = rightVol; 2701} 2702 2703bool AudioFlinger::DirectOutputThread::threadLoop() 2704{ 2705 mixer_state mixerStatus = MIXER_IDLE; 2706 sp<Track> trackToRemove; 2707 sp<Track> activeTrack; 2708 nsecs_t standbyTime = systemTime(); 2709 size_t mixBufferSize = mFrameCount*mFrameSize; 2710 uint32_t activeSleepTime = activeSleepTimeUs(); 2711 uint32_t idleSleepTime = idleSleepTimeUs(); 2712 uint32_t sleepTime = idleSleepTime; 2713 // use shorter standby delay as on normal output to release 2714 // hardware resources as soon as possible 2715 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2716 2717 acquireWakeLock(); 2718 2719 while (!exitPending()) 2720 { 2721 bool rampVolume; 2722 uint16_t leftVol; 2723 uint16_t rightVol; 2724 Vector< sp<EffectChain> > effectChains; 2725 2726 processConfigEvents(); 2727 2728 mixerStatus = MIXER_IDLE; 2729 2730 { // scope for the mLock 2731 2732 Mutex::Autolock _l(mLock); 2733 2734 if (checkForNewParameters_l()) { 2735 mixBufferSize = mFrameCount*mFrameSize; 2736 activeSleepTime = activeSleepTimeUs(); 2737 idleSleepTime = idleSleepTimeUs(); 2738 standbyDelay = microseconds(activeSleepTime*2); 2739 } 2740 2741 // put audio hardware into standby after short delay 2742 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2743 mSuspended)) { 2744 // wait until we have something to do... 2745 if (!mStandby) { 2746 ALOGV("Audio hardware entering standby, mixer %p", this); 2747 mOutput->stream->common.standby(&mOutput->stream->common); 2748 mStandby = true; 2749 mBytesWritten = 0; 2750 } 2751 2752 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2753 // we're about to wait, flush the binder command buffer 2754 IPCThreadState::self()->flushCommands(); 2755 2756 if (exitPending()) break; 2757 2758 releaseWakeLock_l(); 2759 ALOGV("DirectOutputThread %p TID %d going to sleep", this, gettid()); 2760 mWaitWorkCV.wait(mLock); 2761 ALOGV("DirectOutputThread %p TID %d waking up in active mode", this, gettid()); 2762 acquireWakeLock_l(); 2763 2764 checkSilentMode_l(); 2765 2766 standbyTime = systemTime() + standbyDelay; 2767 sleepTime = idleSleepTime; 2768 continue; 2769 } 2770 } 2771 2772 effectChains = mEffectChains; 2773 2774 // find out which tracks need to be processed 2775 if (mActiveTracks.size() != 0) { 2776 sp<Track> t = mActiveTracks[0].promote(); 2777 if (t == 0) continue; 2778 2779 Track* const track = t.get(); 2780 audio_track_cblk_t* cblk = track->cblk(); 2781 2782 // The first time a track is added we wait 2783 // for all its buffers to be filled before processing it 2784 if (cblk->framesReady() && track->isReady() && 2785 !track->isPaused() && !track->isTerminated()) 2786 { 2787 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2788 2789 if (track->mFillingUpStatus == Track::FS_FILLED) { 2790 track->mFillingUpStatus = Track::FS_ACTIVE; 2791 mLeftVolFloat = mRightVolFloat = 0; 2792 mLeftVolShort = mRightVolShort = 0; 2793 if (track->mState == TrackBase::RESUMING) { 2794 track->mState = TrackBase::ACTIVE; 2795 rampVolume = true; 2796 } 2797 } else if (cblk->server != 0) { 2798 // If the track is stopped before the first frame was mixed, 2799 // do not apply ramp 2800 rampVolume = true; 2801 } 2802 // compute volume for this track 2803 float left, right; 2804 if (track->isMuted() || mMasterMute || track->isPausing() || 2805 mStreamTypes[track->streamType()].mute) { 2806 left = right = 0; 2807 if (track->isPausing()) { 2808 track->setPaused(); 2809 } 2810 } else { 2811 float typeVolume = mStreamTypes[track->streamType()].volume; 2812 float v = mMasterVolume * typeVolume; 2813 uint32_t vlr = cblk->getVolumeLR(); 2814 float v_clamped = v * (vlr & 0xFFFF); 2815 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2816 left = v_clamped/MAX_GAIN; 2817 v_clamped = v * (vlr >> 16); 2818 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2819 right = v_clamped/MAX_GAIN; 2820 } 2821 2822 if (left != mLeftVolFloat || right != mRightVolFloat) { 2823 mLeftVolFloat = left; 2824 mRightVolFloat = right; 2825 2826 // If audio HAL implements volume control, 2827 // force software volume to nominal value 2828 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2829 left = 1.0f; 2830 right = 1.0f; 2831 } 2832 2833 // Convert volumes from float to 8.24 2834 uint32_t vl = (uint32_t)(left * (1 << 24)); 2835 uint32_t vr = (uint32_t)(right * (1 << 24)); 2836 2837 // Delegate volume control to effect in track effect chain if needed 2838 // only one effect chain can be present on DirectOutputThread, so if 2839 // there is one, the track is connected to it 2840 if (!effectChains.isEmpty()) { 2841 // Do not ramp volume if volume is controlled by effect 2842 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2843 rampVolume = false; 2844 } 2845 } 2846 2847 // Convert volumes from 8.24 to 4.12 format 2848 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2849 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2850 leftVol = (uint16_t)v_clamped; 2851 v_clamped = (vr + (1 << 11)) >> 12; 2852 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2853 rightVol = (uint16_t)v_clamped; 2854 } else { 2855 leftVol = mLeftVolShort; 2856 rightVol = mRightVolShort; 2857 rampVolume = false; 2858 } 2859 2860 // reset retry count 2861 track->mRetryCount = kMaxTrackRetriesDirect; 2862 activeTrack = t; 2863 mixerStatus = MIXER_TRACKS_READY; 2864 } else { 2865 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2866 if (track->isStopped()) { 2867 track->reset(); 2868 } 2869 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2870 // We have consumed all the buffers of this track. 2871 // Remove it from the list of active tracks. 2872 trackToRemove = track; 2873 } else { 2874 // No buffers for this track. Give it a few chances to 2875 // fill a buffer, then remove it from active list. 2876 if (--(track->mRetryCount) <= 0) { 2877 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2878 trackToRemove = track; 2879 } else { 2880 mixerStatus = MIXER_TRACKS_ENABLED; 2881 } 2882 } 2883 } 2884 } 2885 2886 // remove all the tracks that need to be... 2887 if (CC_UNLIKELY(trackToRemove != 0)) { 2888 mActiveTracks.remove(trackToRemove); 2889 if (!effectChains.isEmpty()) { 2890 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2891 trackToRemove->sessionId()); 2892 effectChains[0]->decActiveTrackCnt(); 2893 } 2894 if (trackToRemove->isTerminated()) { 2895 removeTrack_l(trackToRemove); 2896 } 2897 } 2898 2899 lockEffectChains_l(effectChains); 2900 } 2901 2902 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2903 AudioBufferProvider::Buffer buffer; 2904 size_t frameCount = mFrameCount; 2905 int8_t *curBuf = (int8_t *)mMixBuffer; 2906 // output audio to hardware 2907 while (frameCount) { 2908 buffer.frameCount = frameCount; 2909 activeTrack->getNextBuffer(&buffer, 2910 AudioBufferProvider::kInvalidPTS); 2911 if (CC_UNLIKELY(buffer.raw == NULL)) { 2912 memset(curBuf, 0, frameCount * mFrameSize); 2913 break; 2914 } 2915 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2916 frameCount -= buffer.frameCount; 2917 curBuf += buffer.frameCount * mFrameSize; 2918 activeTrack->releaseBuffer(&buffer); 2919 } 2920 sleepTime = 0; 2921 standbyTime = systemTime() + standbyDelay; 2922 } else { 2923 if (sleepTime == 0) { 2924 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2925 sleepTime = activeSleepTime; 2926 } else { 2927 sleepTime = idleSleepTime; 2928 } 2929 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2930 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2931 sleepTime = 0; 2932 } 2933 } 2934 2935 if (mSuspended) { 2936 sleepTime = suspendSleepTimeUs(); 2937 } 2938 // sleepTime == 0 means we must write to audio hardware 2939 if (sleepTime == 0) { 2940 if (mixerStatus == MIXER_TRACKS_READY) { 2941 applyVolume(leftVol, rightVol, rampVolume); 2942 } 2943 for (size_t i = 0; i < effectChains.size(); i ++) { 2944 effectChains[i]->process_l(); 2945 } 2946 unlockEffectChains(effectChains); 2947 2948 mLastWriteTime = systemTime(); 2949 mInWrite = true; 2950 mBytesWritten += mixBufferSize; 2951 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2952 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2953 mNumWrites++; 2954 mInWrite = false; 2955 mStandby = false; 2956 } else { 2957 unlockEffectChains(effectChains); 2958 usleep(sleepTime); 2959 } 2960 2961 // finally let go of removed track, without the lock held 2962 // since we can't guarantee the destructors won't acquire that 2963 // same lock. 2964 trackToRemove.clear(); 2965 activeTrack.clear(); 2966 2967 // Effect chains will be actually deleted here if they were removed from 2968 // mEffectChains list during mixing or effects processing 2969 effectChains.clear(); 2970 } 2971 2972 if (!mStandby) { 2973 mOutput->stream->common.standby(&mOutput->stream->common); 2974 } 2975 2976 releaseWakeLock(); 2977 2978 ALOGV("DirectOutputThread %p exiting", this); 2979 return false; 2980} 2981 2982// getTrackName_l() must be called with ThreadBase::mLock held 2983int AudioFlinger::DirectOutputThread::getTrackName_l() 2984{ 2985 return 0; 2986} 2987 2988// deleteTrackName_l() must be called with ThreadBase::mLock held 2989void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2990{ 2991} 2992 2993// checkForNewParameters_l() must be called with ThreadBase::mLock held 2994bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2995{ 2996 bool reconfig = false; 2997 2998 while (!mNewParameters.isEmpty()) { 2999 status_t status = NO_ERROR; 3000 String8 keyValuePair = mNewParameters[0]; 3001 AudioParameter param = AudioParameter(keyValuePair); 3002 int value; 3003 3004 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3005 // do not accept frame count changes if tracks are open as the track buffer 3006 // size depends on frame count and correct behavior would not be garantied 3007 // if frame count is changed after track creation 3008 if (!mTracks.isEmpty()) { 3009 status = INVALID_OPERATION; 3010 } else { 3011 reconfig = true; 3012 } 3013 } 3014 if (status == NO_ERROR) { 3015 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3016 keyValuePair.string()); 3017 if (!mStandby && status == INVALID_OPERATION) { 3018 mOutput->stream->common.standby(&mOutput->stream->common); 3019 mStandby = true; 3020 mBytesWritten = 0; 3021 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3022 keyValuePair.string()); 3023 } 3024 if (status == NO_ERROR && reconfig) { 3025 readOutputParameters(); 3026 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3027 } 3028 } 3029 3030 mNewParameters.removeAt(0); 3031 3032 mParamStatus = status; 3033 mParamCond.signal(); 3034 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3035 // already timed out waiting for the status and will never signal the condition. 3036 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3037 } 3038 return reconfig; 3039} 3040 3041uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3042{ 3043 uint32_t time; 3044 if (audio_is_linear_pcm(mFormat)) { 3045 time = PlaybackThread::activeSleepTimeUs(); 3046 } else { 3047 time = 10000; 3048 } 3049 return time; 3050} 3051 3052uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3053{ 3054 uint32_t time; 3055 if (audio_is_linear_pcm(mFormat)) { 3056 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3057 } else { 3058 time = 10000; 3059 } 3060 return time; 3061} 3062 3063uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3064{ 3065 uint32_t time; 3066 if (audio_is_linear_pcm(mFormat)) { 3067 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3068 } else { 3069 time = 10000; 3070 } 3071 return time; 3072} 3073 3074 3075// ---------------------------------------------------------------------------- 3076 3077AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3078 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3079 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3080 mWaitTimeMs(UINT_MAX) 3081{ 3082 addOutputTrack(mainThread); 3083} 3084 3085AudioFlinger::DuplicatingThread::~DuplicatingThread() 3086{ 3087 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3088 mOutputTracks[i]->destroy(); 3089 } 3090} 3091 3092bool AudioFlinger::DuplicatingThread::threadLoop() 3093{ 3094 Vector< sp<Track> > tracksToRemove; 3095 mixer_state mixerStatus = MIXER_IDLE; 3096 nsecs_t standbyTime = systemTime(); 3097 size_t mixBufferSize = mFrameCount*mFrameSize; 3098 SortedVector< sp<OutputTrack> > outputTracks; 3099 uint32_t writeFrames = 0; 3100 uint32_t activeSleepTime = activeSleepTimeUs(); 3101 uint32_t idleSleepTime = idleSleepTimeUs(); 3102 uint32_t sleepTime = idleSleepTime; 3103 Vector< sp<EffectChain> > effectChains; 3104 3105 acquireWakeLock(); 3106 3107 while (!exitPending()) 3108 { 3109 processConfigEvents(); 3110 3111 mixerStatus = MIXER_IDLE; 3112 { // scope for the mLock 3113 3114 Mutex::Autolock _l(mLock); 3115 3116 if (checkForNewParameters_l()) { 3117 mixBufferSize = mFrameCount*mFrameSize; 3118 updateWaitTime(); 3119 activeSleepTime = activeSleepTimeUs(); 3120 idleSleepTime = idleSleepTimeUs(); 3121 } 3122 3123 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3124 3125 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3126 outputTracks.add(mOutputTracks[i]); 3127 } 3128 3129 // put audio hardware into standby after short delay 3130 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3131 mSuspended)) { 3132 if (!mStandby) { 3133 for (size_t i = 0; i < outputTracks.size(); i++) { 3134 outputTracks[i]->stop(); 3135 } 3136 mStandby = true; 3137 mBytesWritten = 0; 3138 } 3139 3140 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3141 // we're about to wait, flush the binder command buffer 3142 IPCThreadState::self()->flushCommands(); 3143 outputTracks.clear(); 3144 3145 if (exitPending()) break; 3146 3147 releaseWakeLock_l(); 3148 ALOGV("DuplicatingThread %p TID %d going to sleep", this, gettid()); 3149 mWaitWorkCV.wait(mLock); 3150 ALOGV("DuplicatingThread %p TID %d waking up", this, gettid()); 3151 acquireWakeLock_l(); 3152 3153 checkSilentMode_l(); 3154 3155 standbyTime = systemTime() + mStandbyTimeInNsecs; 3156 sleepTime = idleSleepTime; 3157 continue; 3158 } 3159 } 3160 3161 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3162 3163 // prevent any changes in effect chain list and in each effect chain 3164 // during mixing and effect process as the audio buffers could be deleted 3165 // or modified if an effect is created or deleted 3166 lockEffectChains_l(effectChains); 3167 } 3168 3169 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3170 // mix buffers... 3171 if (outputsReady(outputTracks)) { 3172 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3173 } else { 3174 memset(mMixBuffer, 0, mixBufferSize); 3175 } 3176 sleepTime = 0; 3177 writeFrames = mFrameCount; 3178 } else { 3179 if (sleepTime == 0) { 3180 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3181 sleepTime = activeSleepTime; 3182 } else { 3183 sleepTime = idleSleepTime; 3184 } 3185 } else if (mBytesWritten != 0) { 3186 // flush remaining overflow buffers in output tracks 3187 for (size_t i = 0; i < outputTracks.size(); i++) { 3188 if (outputTracks[i]->isActive()) { 3189 sleepTime = 0; 3190 writeFrames = 0; 3191 memset(mMixBuffer, 0, mixBufferSize); 3192 break; 3193 } 3194 } 3195 } 3196 } 3197 3198 if (mSuspended) { 3199 sleepTime = suspendSleepTimeUs(); 3200 } 3201 // sleepTime == 0 means we must write to audio hardware 3202 if (sleepTime == 0) { 3203 for (size_t i = 0; i < effectChains.size(); i ++) { 3204 effectChains[i]->process_l(); 3205 } 3206 // enable changes in effect chain 3207 unlockEffectChains(effectChains); 3208 3209 standbyTime = systemTime() + mStandbyTimeInNsecs; 3210 for (size_t i = 0; i < outputTracks.size(); i++) { 3211 outputTracks[i]->write(mMixBuffer, writeFrames); 3212 } 3213 mStandby = false; 3214 mBytesWritten += mixBufferSize; 3215 } else { 3216 // enable changes in effect chain 3217 unlockEffectChains(effectChains); 3218 usleep(sleepTime); 3219 } 3220 3221 // finally let go of all our tracks, without the lock held 3222 // since we can't guarantee the destructors won't acquire that 3223 // same lock. 3224 tracksToRemove.clear(); 3225 outputTracks.clear(); 3226 3227 // Effect chains will be actually deleted here if they were removed from 3228 // mEffectChains list during mixing or effects processing 3229 effectChains.clear(); 3230 } 3231 3232 releaseWakeLock(); 3233 3234 return false; 3235} 3236 3237void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3238{ 3239 // FIXME explain this formula 3240 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3241 OutputTrack *outputTrack = new OutputTrack(thread, 3242 this, 3243 mSampleRate, 3244 mFormat, 3245 mChannelMask, 3246 frameCount); 3247 if (outputTrack->cblk() != NULL) { 3248 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3249 mOutputTracks.add(outputTrack); 3250 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3251 updateWaitTime(); 3252 } 3253} 3254 3255void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3256{ 3257 Mutex::Autolock _l(mLock); 3258 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3259 if (mOutputTracks[i]->thread() == thread) { 3260 mOutputTracks[i]->destroy(); 3261 mOutputTracks.removeAt(i); 3262 updateWaitTime(); 3263 return; 3264 } 3265 } 3266 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3267} 3268 3269void AudioFlinger::DuplicatingThread::updateWaitTime() 3270{ 3271 mWaitTimeMs = UINT_MAX; 3272 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3273 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3274 if (strong != 0) { 3275 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3276 if (waitTimeMs < mWaitTimeMs) { 3277 mWaitTimeMs = waitTimeMs; 3278 } 3279 } 3280 } 3281} 3282 3283 3284bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3285{ 3286 for (size_t i = 0; i < outputTracks.size(); i++) { 3287 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3288 if (thread == 0) { 3289 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3290 return false; 3291 } 3292 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3293 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3294 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3295 return false; 3296 } 3297 } 3298 return true; 3299} 3300 3301uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3302{ 3303 return (mWaitTimeMs * 1000) / 2; 3304} 3305 3306// ---------------------------------------------------------------------------- 3307 3308// TrackBase constructor must be called with AudioFlinger::mLock held 3309AudioFlinger::ThreadBase::TrackBase::TrackBase( 3310 ThreadBase *thread, 3311 const sp<Client>& client, 3312 uint32_t sampleRate, 3313 audio_format_t format, 3314 uint32_t channelMask, 3315 int frameCount, 3316 const sp<IMemory>& sharedBuffer, 3317 int sessionId) 3318 : RefBase(), 3319 mThread(thread), 3320 mClient(client), 3321 mCblk(NULL), 3322 // mBuffer 3323 // mBufferEnd 3324 mFrameCount(0), 3325 mState(IDLE), 3326 mFormat(format), 3327 mStepServerFailed(false), 3328 mSessionId(sessionId) 3329 // mChannelCount 3330 // mChannelMask 3331{ 3332 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3333 3334 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3335 size_t size = sizeof(audio_track_cblk_t); 3336 uint8_t channelCount = popcount(channelMask); 3337 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3338 if (sharedBuffer == 0) { 3339 size += bufferSize; 3340 } 3341 3342 if (client != NULL) { 3343 mCblkMemory = client->heap()->allocate(size); 3344 if (mCblkMemory != 0) { 3345 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3346 if (mCblk != NULL) { // construct the shared structure in-place. 3347 new(mCblk) audio_track_cblk_t(); 3348 // clear all buffers 3349 mCblk->frameCount = frameCount; 3350 mCblk->sampleRate = sampleRate; 3351 mChannelCount = channelCount; 3352 mChannelMask = channelMask; 3353 if (sharedBuffer == 0) { 3354 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3355 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3356 // Force underrun condition to avoid false underrun callback until first data is 3357 // written to buffer (other flags are cleared) 3358 mCblk->flags = CBLK_UNDERRUN_ON; 3359 } else { 3360 mBuffer = sharedBuffer->pointer(); 3361 } 3362 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3363 } 3364 } else { 3365 ALOGE("not enough memory for AudioTrack size=%u", size); 3366 client->heap()->dump("AudioTrack"); 3367 return; 3368 } 3369 } else { 3370 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3371 // construct the shared structure in-place. 3372 new(mCblk) audio_track_cblk_t(); 3373 // clear all buffers 3374 mCblk->frameCount = frameCount; 3375 mCblk->sampleRate = sampleRate; 3376 mChannelCount = channelCount; 3377 mChannelMask = channelMask; 3378 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3379 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3380 // Force underrun condition to avoid false underrun callback until first data is 3381 // written to buffer (other flags are cleared) 3382 mCblk->flags = CBLK_UNDERRUN_ON; 3383 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3384 } 3385} 3386 3387AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3388{ 3389 if (mCblk != NULL) { 3390 if (mClient == 0) { 3391 delete mCblk; 3392 } else { 3393 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3394 } 3395 } 3396 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3397 if (mClient != 0) { 3398 // Client destructor must run with AudioFlinger mutex locked 3399 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3400 // If the client's reference count drops to zero, the associated destructor 3401 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3402 // relying on the automatic clear() at end of scope. 3403 mClient.clear(); 3404 } 3405} 3406 3407void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3408{ 3409 buffer->raw = NULL; 3410 mFrameCount = buffer->frameCount; 3411 step(); 3412 buffer->frameCount = 0; 3413} 3414 3415bool AudioFlinger::ThreadBase::TrackBase::step() { 3416 bool result; 3417 audio_track_cblk_t* cblk = this->cblk(); 3418 3419 result = cblk->stepServer(mFrameCount); 3420 if (!result) { 3421 ALOGV("stepServer failed acquiring cblk mutex"); 3422 mStepServerFailed = true; 3423 } 3424 return result; 3425} 3426 3427void AudioFlinger::ThreadBase::TrackBase::reset() { 3428 audio_track_cblk_t* cblk = this->cblk(); 3429 3430 cblk->user = 0; 3431 cblk->server = 0; 3432 cblk->userBase = 0; 3433 cblk->serverBase = 0; 3434 mStepServerFailed = false; 3435 ALOGV("TrackBase::reset"); 3436} 3437 3438int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3439 return (int)mCblk->sampleRate; 3440} 3441 3442void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3443 audio_track_cblk_t* cblk = this->cblk(); 3444 size_t frameSize = cblk->frameSize; 3445 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3446 int8_t *bufferEnd = bufferStart + frames * frameSize; 3447 3448 // Check validity of returned pointer in case the track control block would have been corrupted. 3449 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3450 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3451 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3452 server %d, serverBase %d, user %d, userBase %d", 3453 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3454 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3455 return NULL; 3456 } 3457 3458 return bufferStart; 3459} 3460 3461// ---------------------------------------------------------------------------- 3462 3463// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3464AudioFlinger::PlaybackThread::Track::Track( 3465 PlaybackThread *thread, 3466 const sp<Client>& client, 3467 audio_stream_type_t streamType, 3468 uint32_t sampleRate, 3469 audio_format_t format, 3470 uint32_t channelMask, 3471 int frameCount, 3472 const sp<IMemory>& sharedBuffer, 3473 int sessionId) 3474 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3475 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3476 mAuxEffectId(0), mHasVolumeController(false) 3477{ 3478 if (mCblk != NULL) { 3479 if (thread != NULL) { 3480 mName = thread->getTrackName_l(); 3481 mMainBuffer = thread->mixBuffer(); 3482 } 3483 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3484 if (mName < 0) { 3485 ALOGE("no more track names available"); 3486 } 3487 mStreamType = streamType; 3488 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3489 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3490 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3491 } 3492} 3493 3494AudioFlinger::PlaybackThread::Track::~Track() 3495{ 3496 ALOGV("PlaybackThread::Track destructor"); 3497 sp<ThreadBase> thread = mThread.promote(); 3498 if (thread != 0) { 3499 Mutex::Autolock _l(thread->mLock); 3500 mState = TERMINATED; 3501 } 3502} 3503 3504void AudioFlinger::PlaybackThread::Track::destroy() 3505{ 3506 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3507 // by removing it from mTracks vector, so there is a risk that this Tracks's 3508 // destructor is called. As the destructor needs to lock mLock, 3509 // we must acquire a strong reference on this Track before locking mLock 3510 // here so that the destructor is called only when exiting this function. 3511 // On the other hand, as long as Track::destroy() is only called by 3512 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3513 // this Track with its member mTrack. 3514 sp<Track> keep(this); 3515 { // scope for mLock 3516 sp<ThreadBase> thread = mThread.promote(); 3517 if (thread != 0) { 3518 if (!isOutputTrack()) { 3519 if (mState == ACTIVE || mState == RESUMING) { 3520 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3521 3522 // to track the speaker usage 3523 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3524 } 3525 AudioSystem::releaseOutput(thread->id()); 3526 } 3527 Mutex::Autolock _l(thread->mLock); 3528 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3529 playbackThread->destroyTrack_l(this); 3530 } 3531 } 3532} 3533 3534void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3535{ 3536 uint32_t vlr = mCblk->getVolumeLR(); 3537 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", 3538 mName - AudioMixer::TRACK0, 3539 (mClient == 0) ? getpid_cached : mClient->pid(), 3540 mStreamType, 3541 mFormat, 3542 mChannelMask, 3543 mSessionId, 3544 mFrameCount, 3545 mState, 3546 mMute, 3547 mFillingUpStatus, 3548 mCblk->sampleRate, 3549 vlr & 0xFFFF, 3550 vlr >> 16, 3551 mCblk->server, 3552 mCblk->user, 3553 (int)mMainBuffer, 3554 (int)mAuxBuffer); 3555} 3556 3557status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3558 AudioBufferProvider::Buffer* buffer, int64_t pts) 3559{ 3560 audio_track_cblk_t* cblk = this->cblk(); 3561 uint32_t framesReady; 3562 uint32_t framesReq = buffer->frameCount; 3563 3564 // Check if last stepServer failed, try to step now 3565 if (mStepServerFailed) { 3566 if (!step()) goto getNextBuffer_exit; 3567 ALOGV("stepServer recovered"); 3568 mStepServerFailed = false; 3569 } 3570 3571 framesReady = cblk->framesReady(); 3572 3573 if (CC_LIKELY(framesReady)) { 3574 uint32_t s = cblk->server; 3575 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3576 3577 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3578 if (framesReq > framesReady) { 3579 framesReq = framesReady; 3580 } 3581 if (s + framesReq > bufferEnd) { 3582 framesReq = bufferEnd - s; 3583 } 3584 3585 buffer->raw = getBuffer(s, framesReq); 3586 if (buffer->raw == NULL) goto getNextBuffer_exit; 3587 3588 buffer->frameCount = framesReq; 3589 return NO_ERROR; 3590 } 3591 3592getNextBuffer_exit: 3593 buffer->raw = NULL; 3594 buffer->frameCount = 0; 3595 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3596 return NOT_ENOUGH_DATA; 3597} 3598 3599uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3600 return mCblk->framesReady(); 3601} 3602 3603bool AudioFlinger::PlaybackThread::Track::isReady() const { 3604 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3605 3606 if (framesReady() >= mCblk->frameCount || 3607 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3608 mFillingUpStatus = FS_FILLED; 3609 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3610 return true; 3611 } 3612 return false; 3613} 3614 3615status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3616{ 3617 status_t status = NO_ERROR; 3618 ALOGV("start(%d), calling pid %d session %d tid %d", 3619 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3620 sp<ThreadBase> thread = mThread.promote(); 3621 if (thread != 0) { 3622 Mutex::Autolock _l(thread->mLock); 3623 track_state state = mState; 3624 // here the track could be either new, or restarted 3625 // in both cases "unstop" the track 3626 if (mState == PAUSED) { 3627 mState = TrackBase::RESUMING; 3628 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3629 } else { 3630 mState = TrackBase::ACTIVE; 3631 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3632 } 3633 3634 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3635 thread->mLock.unlock(); 3636 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3637 thread->mLock.lock(); 3638 3639 // to track the speaker usage 3640 if (status == NO_ERROR) { 3641 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3642 } 3643 } 3644 if (status == NO_ERROR) { 3645 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3646 playbackThread->addTrack_l(this); 3647 } else { 3648 mState = state; 3649 } 3650 } else { 3651 status = BAD_VALUE; 3652 } 3653 return status; 3654} 3655 3656void AudioFlinger::PlaybackThread::Track::stop() 3657{ 3658 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3659 sp<ThreadBase> thread = mThread.promote(); 3660 if (thread != 0) { 3661 Mutex::Autolock _l(thread->mLock); 3662 track_state state = mState; 3663 if (mState > STOPPED) { 3664 mState = STOPPED; 3665 // If the track is not active (PAUSED and buffers full), flush buffers 3666 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3667 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3668 reset(); 3669 } 3670 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3671 } 3672 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3673 thread->mLock.unlock(); 3674 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3675 thread->mLock.lock(); 3676 3677 // to track the speaker usage 3678 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3679 } 3680 } 3681} 3682 3683void AudioFlinger::PlaybackThread::Track::pause() 3684{ 3685 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3686 sp<ThreadBase> thread = mThread.promote(); 3687 if (thread != 0) { 3688 Mutex::Autolock _l(thread->mLock); 3689 if (mState == ACTIVE || mState == RESUMING) { 3690 mState = PAUSING; 3691 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3692 if (!isOutputTrack()) { 3693 thread->mLock.unlock(); 3694 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3695 thread->mLock.lock(); 3696 3697 // to track the speaker usage 3698 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3699 } 3700 } 3701 } 3702} 3703 3704void AudioFlinger::PlaybackThread::Track::flush() 3705{ 3706 ALOGV("flush(%d)", mName); 3707 sp<ThreadBase> thread = mThread.promote(); 3708 if (thread != 0) { 3709 Mutex::Autolock _l(thread->mLock); 3710 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3711 return; 3712 } 3713 // No point remaining in PAUSED state after a flush => go to 3714 // STOPPED state 3715 mState = STOPPED; 3716 3717 // do not reset the track if it is still in the process of being stopped or paused. 3718 // this will be done by prepareTracks_l() when the track is stopped. 3719 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3720 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3721 reset(); 3722 } 3723 } 3724} 3725 3726void AudioFlinger::PlaybackThread::Track::reset() 3727{ 3728 // Do not reset twice to avoid discarding data written just after a flush and before 3729 // the audioflinger thread detects the track is stopped. 3730 if (!mResetDone) { 3731 TrackBase::reset(); 3732 // Force underrun condition to avoid false underrun callback until first data is 3733 // written to buffer 3734 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3735 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3736 mFillingUpStatus = FS_FILLING; 3737 mResetDone = true; 3738 } 3739} 3740 3741void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3742{ 3743 mMute = muted; 3744} 3745 3746status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3747{ 3748 status_t status = DEAD_OBJECT; 3749 sp<ThreadBase> thread = mThread.promote(); 3750 if (thread != 0) { 3751 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3752 status = playbackThread->attachAuxEffect(this, EffectId); 3753 } 3754 return status; 3755} 3756 3757void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3758{ 3759 mAuxEffectId = EffectId; 3760 mAuxBuffer = buffer; 3761} 3762 3763// timed audio tracks 3764 3765sp<AudioFlinger::PlaybackThread::TimedTrack> 3766AudioFlinger::PlaybackThread::TimedTrack::create( 3767 PlaybackThread *thread, 3768 const sp<Client>& client, 3769 audio_stream_type_t streamType, 3770 uint32_t sampleRate, 3771 audio_format_t format, 3772 uint32_t channelMask, 3773 int frameCount, 3774 const sp<IMemory>& sharedBuffer, 3775 int sessionId) { 3776 if (!client->reserveTimedTrack()) 3777 return NULL; 3778 3779 sp<TimedTrack> track = new TimedTrack( 3780 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3781 sharedBuffer, sessionId); 3782 3783 if (track == NULL) { 3784 client->releaseTimedTrack(); 3785 return NULL; 3786 } 3787 3788 return track; 3789} 3790 3791AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3792 PlaybackThread *thread, 3793 const sp<Client>& client, 3794 audio_stream_type_t streamType, 3795 uint32_t sampleRate, 3796 audio_format_t format, 3797 uint32_t channelMask, 3798 int frameCount, 3799 const sp<IMemory>& sharedBuffer, 3800 int sessionId) 3801 : Track(thread, client, streamType, sampleRate, format, channelMask, 3802 frameCount, sharedBuffer, sessionId), 3803 mTimedSilenceBuffer(NULL), 3804 mTimedSilenceBufferSize(0), 3805 mTimedAudioOutputOnTime(false), 3806 mMediaTimeTransformValid(false) 3807{ 3808 LocalClock lc; 3809 mLocalTimeFreq = lc.getLocalFreq(); 3810 3811 mLocalTimeToSampleTransform.a_zero = 0; 3812 mLocalTimeToSampleTransform.b_zero = 0; 3813 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3814 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3815 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3816 &mLocalTimeToSampleTransform.a_to_b_denom); 3817} 3818 3819AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3820 mClient->releaseTimedTrack(); 3821 delete [] mTimedSilenceBuffer; 3822} 3823 3824status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3825 size_t size, sp<IMemory>* buffer) { 3826 3827 Mutex::Autolock _l(mTimedBufferQueueLock); 3828 3829 trimTimedBufferQueue_l(); 3830 3831 // lazily initialize the shared memory heap for timed buffers 3832 if (mTimedMemoryDealer == NULL) { 3833 const int kTimedBufferHeapSize = 512 << 10; 3834 3835 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3836 "AudioFlingerTimed"); 3837 if (mTimedMemoryDealer == NULL) 3838 return NO_MEMORY; 3839 } 3840 3841 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3842 if (newBuffer == NULL) { 3843 newBuffer = mTimedMemoryDealer->allocate(size); 3844 if (newBuffer == NULL) 3845 return NO_MEMORY; 3846 } 3847 3848 *buffer = newBuffer; 3849 return NO_ERROR; 3850} 3851 3852// caller must hold mTimedBufferQueueLock 3853void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3854 int64_t mediaTimeNow; 3855 { 3856 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3857 if (!mMediaTimeTransformValid) 3858 return; 3859 3860 int64_t targetTimeNow; 3861 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3862 ? mCCHelper.getCommonTime(&targetTimeNow) 3863 : mCCHelper.getLocalTime(&targetTimeNow); 3864 3865 if (OK != res) 3866 return; 3867 3868 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3869 &mediaTimeNow)) { 3870 return; 3871 } 3872 } 3873 3874 size_t trimIndex; 3875 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3876 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3877 break; 3878 } 3879 3880 if (trimIndex) { 3881 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3882 } 3883} 3884 3885status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3886 const sp<IMemory>& buffer, int64_t pts) { 3887 3888 { 3889 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3890 if (!mMediaTimeTransformValid) 3891 return INVALID_OPERATION; 3892 } 3893 3894 Mutex::Autolock _l(mTimedBufferQueueLock); 3895 3896 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3897 3898 return NO_ERROR; 3899} 3900 3901status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3902 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3903 3904 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3905 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3906 target); 3907 3908 if (!(target == TimedAudioTrack::LOCAL_TIME || 3909 target == TimedAudioTrack::COMMON_TIME)) { 3910 return BAD_VALUE; 3911 } 3912 3913 Mutex::Autolock lock(mMediaTimeTransformLock); 3914 mMediaTimeTransform = xform; 3915 mMediaTimeTransformTarget = target; 3916 mMediaTimeTransformValid = true; 3917 3918 return NO_ERROR; 3919} 3920 3921#define min(a, b) ((a) < (b) ? (a) : (b)) 3922 3923// implementation of getNextBuffer for tracks whose buffers have timestamps 3924status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3925 AudioBufferProvider::Buffer* buffer, int64_t pts) 3926{ 3927 if (pts == AudioBufferProvider::kInvalidPTS) { 3928 buffer->raw = 0; 3929 buffer->frameCount = 0; 3930 return INVALID_OPERATION; 3931 } 3932 3933 Mutex::Autolock _l(mTimedBufferQueueLock); 3934 3935 while (true) { 3936 3937 // if we have no timed buffers, then fail 3938 if (mTimedBufferQueue.isEmpty()) { 3939 buffer->raw = 0; 3940 buffer->frameCount = 0; 3941 return NOT_ENOUGH_DATA; 3942 } 3943 3944 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3945 3946 // calculate the PTS of the head of the timed buffer queue expressed in 3947 // local time 3948 int64_t headLocalPTS; 3949 { 3950 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3951 3952 assert(mMediaTimeTransformValid); 3953 3954 if (mMediaTimeTransform.a_to_b_denom == 0) { 3955 // the transform represents a pause, so yield silence 3956 timedYieldSilence(buffer->frameCount, buffer); 3957 return NO_ERROR; 3958 } 3959 3960 int64_t transformedPTS; 3961 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3962 &transformedPTS)) { 3963 // the transform failed. this shouldn't happen, but if it does 3964 // then just drop this buffer 3965 ALOGW("timedGetNextBuffer transform failed"); 3966 buffer->raw = 0; 3967 buffer->frameCount = 0; 3968 mTimedBufferQueue.removeAt(0); 3969 return NO_ERROR; 3970 } 3971 3972 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3973 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3974 &headLocalPTS)) { 3975 buffer->raw = 0; 3976 buffer->frameCount = 0; 3977 return INVALID_OPERATION; 3978 } 3979 } else { 3980 headLocalPTS = transformedPTS; 3981 } 3982 } 3983 3984 // adjust the head buffer's PTS to reflect the portion of the head buffer 3985 // that has already been consumed 3986 int64_t effectivePTS = headLocalPTS + 3987 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3988 3989 // Calculate the delta in samples between the head of the input buffer 3990 // queue and the start of the next output buffer that will be written. 3991 // If the transformation fails because of over or underflow, it means 3992 // that the sample's position in the output stream is so far out of 3993 // whack that it should just be dropped. 3994 int64_t sampleDelta; 3995 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3996 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3997 mTimedBufferQueue.removeAt(0); 3998 continue; 3999 } 4000 if (!mLocalTimeToSampleTransform.doForwardTransform( 4001 (effectivePTS - pts) << 32, &sampleDelta)) { 4002 ALOGV("*** too late during sample rate transform: dropped buffer"); 4003 mTimedBufferQueue.removeAt(0); 4004 continue; 4005 } 4006 4007 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4008 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4009 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4010 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4011 4012 // if the delta between the ideal placement for the next input sample and 4013 // the current output position is within this threshold, then we will 4014 // concatenate the next input samples to the previous output 4015 const int64_t kSampleContinuityThreshold = 4016 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4017 4018 // if this is the first buffer of audio that we're emitting from this track 4019 // then it should be almost exactly on time. 4020 const int64_t kSampleStartupThreshold = 1LL << 32; 4021 4022 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4023 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4024 // the next input is close enough to being on time, so concatenate it 4025 // with the last output 4026 timedYieldSamples(buffer); 4027 4028 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4029 return NO_ERROR; 4030 } else if (sampleDelta > 0) { 4031 // the gap between the current output position and the proper start of 4032 // the next input sample is too big, so fill it with silence 4033 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4034 4035 timedYieldSilence(framesUntilNextInput, buffer); 4036 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4037 return NO_ERROR; 4038 } else { 4039 // the next input sample is late 4040 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4041 size_t onTimeSamplePosition = 4042 head.position() + lateFrames * mCblk->frameSize; 4043 4044 if (onTimeSamplePosition > head.buffer()->size()) { 4045 // all the remaining samples in the head are too late, so 4046 // drop it and move on 4047 ALOGV("*** too late: dropped buffer"); 4048 mTimedBufferQueue.removeAt(0); 4049 continue; 4050 } else { 4051 // skip over the late samples 4052 head.setPosition(onTimeSamplePosition); 4053 4054 // yield the available samples 4055 timedYieldSamples(buffer); 4056 4057 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4058 return NO_ERROR; 4059 } 4060 } 4061 } 4062} 4063 4064// Yield samples from the timed buffer queue head up to the given output 4065// buffer's capacity. 4066// 4067// Caller must hold mTimedBufferQueueLock 4068void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4069 AudioBufferProvider::Buffer* buffer) { 4070 4071 const TimedBuffer& head = mTimedBufferQueue[0]; 4072 4073 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4074 head.position()); 4075 4076 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4077 mCblk->frameSize); 4078 size_t framesRequested = buffer->frameCount; 4079 buffer->frameCount = min(framesLeftInHead, framesRequested); 4080 4081 mTimedAudioOutputOnTime = true; 4082} 4083 4084// Yield samples of silence up to the given output buffer's capacity 4085// 4086// Caller must hold mTimedBufferQueueLock 4087void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4088 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4089 4090 // lazily allocate a buffer filled with silence 4091 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4092 delete [] mTimedSilenceBuffer; 4093 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4094 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4095 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4096 } 4097 4098 buffer->raw = mTimedSilenceBuffer; 4099 size_t framesRequested = buffer->frameCount; 4100 buffer->frameCount = min(numFrames, framesRequested); 4101 4102 mTimedAudioOutputOnTime = false; 4103} 4104 4105void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4106 AudioBufferProvider::Buffer* buffer) { 4107 4108 Mutex::Autolock _l(mTimedBufferQueueLock); 4109 4110 // If the buffer which was just released is part of the buffer at the head 4111 // of the queue, be sure to update the amt of the buffer which has been 4112 // consumed. If the buffer being returned is not part of the head of the 4113 // queue, its either because the buffer is part of the silence buffer, or 4114 // because the head of the timed queue was trimmed after the mixer called 4115 // getNextBuffer but before the mixer called releaseBuffer. 4116 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4117 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4118 4119 void* start = head.buffer()->pointer(); 4120 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4121 4122 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4123 head.setPosition(head.position() + 4124 (buffer->frameCount * mCblk->frameSize)); 4125 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4126 mTimedBufferQueue.removeAt(0); 4127 } 4128 } 4129 } 4130 4131 buffer->raw = 0; 4132 buffer->frameCount = 0; 4133} 4134 4135uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4136 Mutex::Autolock _l(mTimedBufferQueueLock); 4137 4138 uint32_t frames = 0; 4139 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4140 const TimedBuffer& tb = mTimedBufferQueue[i]; 4141 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4142 } 4143 4144 return frames; 4145} 4146 4147AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4148 : mPTS(0), mPosition(0) {} 4149 4150AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4151 const sp<IMemory>& buffer, int64_t pts) 4152 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4153 4154// ---------------------------------------------------------------------------- 4155 4156// RecordTrack constructor must be called with AudioFlinger::mLock held 4157AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4158 RecordThread *thread, 4159 const sp<Client>& client, 4160 uint32_t sampleRate, 4161 audio_format_t format, 4162 uint32_t channelMask, 4163 int frameCount, 4164 int sessionId) 4165 : TrackBase(thread, client, sampleRate, format, 4166 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4167 mOverflow(false) 4168{ 4169 if (mCblk != NULL) { 4170 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4171 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4172 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4173 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4174 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4175 } else { 4176 mCblk->frameSize = sizeof(int8_t); 4177 } 4178 } 4179} 4180 4181AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4182{ 4183 sp<ThreadBase> thread = mThread.promote(); 4184 if (thread != 0) { 4185 AudioSystem::releaseInput(thread->id()); 4186 } 4187} 4188 4189status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4190{ 4191 audio_track_cblk_t* cblk = this->cblk(); 4192 uint32_t framesAvail; 4193 uint32_t framesReq = buffer->frameCount; 4194 4195 // Check if last stepServer failed, try to step now 4196 if (mStepServerFailed) { 4197 if (!step()) goto getNextBuffer_exit; 4198 ALOGV("stepServer recovered"); 4199 mStepServerFailed = false; 4200 } 4201 4202 framesAvail = cblk->framesAvailable_l(); 4203 4204 if (CC_LIKELY(framesAvail)) { 4205 uint32_t s = cblk->server; 4206 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4207 4208 if (framesReq > framesAvail) { 4209 framesReq = framesAvail; 4210 } 4211 if (s + framesReq > bufferEnd) { 4212 framesReq = bufferEnd - s; 4213 } 4214 4215 buffer->raw = getBuffer(s, framesReq); 4216 if (buffer->raw == NULL) goto getNextBuffer_exit; 4217 4218 buffer->frameCount = framesReq; 4219 return NO_ERROR; 4220 } 4221 4222getNextBuffer_exit: 4223 buffer->raw = NULL; 4224 buffer->frameCount = 0; 4225 return NOT_ENOUGH_DATA; 4226} 4227 4228status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4229{ 4230 sp<ThreadBase> thread = mThread.promote(); 4231 if (thread != 0) { 4232 RecordThread *recordThread = (RecordThread *)thread.get(); 4233 return recordThread->start(this, tid); 4234 } else { 4235 return BAD_VALUE; 4236 } 4237} 4238 4239void AudioFlinger::RecordThread::RecordTrack::stop() 4240{ 4241 sp<ThreadBase> thread = mThread.promote(); 4242 if (thread != 0) { 4243 RecordThread *recordThread = (RecordThread *)thread.get(); 4244 recordThread->stop(this); 4245 TrackBase::reset(); 4246 // Force overerrun condition to avoid false overrun callback until first data is 4247 // read from buffer 4248 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4249 } 4250} 4251 4252void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4253{ 4254 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4255 (mClient == 0) ? getpid_cached : mClient->pid(), 4256 mFormat, 4257 mChannelMask, 4258 mSessionId, 4259 mFrameCount, 4260 mState, 4261 mCblk->sampleRate, 4262 mCblk->server, 4263 mCblk->user); 4264} 4265 4266 4267// ---------------------------------------------------------------------------- 4268 4269AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4270 PlaybackThread *playbackThread, 4271 DuplicatingThread *sourceThread, 4272 uint32_t sampleRate, 4273 audio_format_t format, 4274 uint32_t channelMask, 4275 int frameCount) 4276 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4277 mActive(false), mSourceThread(sourceThread) 4278{ 4279 4280 if (mCblk != NULL) { 4281 mCblk->flags |= CBLK_DIRECTION_OUT; 4282 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4283 mOutBuffer.frameCount = 0; 4284 playbackThread->mTracks.add(this); 4285 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4286 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4287 mCblk, mBuffer, mCblk->buffers, 4288 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4289 } else { 4290 ALOGW("Error creating output track on thread %p", playbackThread); 4291 } 4292} 4293 4294AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4295{ 4296 clearBufferQueue(); 4297} 4298 4299status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4300{ 4301 status_t status = Track::start(tid); 4302 if (status != NO_ERROR) { 4303 return status; 4304 } 4305 4306 mActive = true; 4307 mRetryCount = 127; 4308 return status; 4309} 4310 4311void AudioFlinger::PlaybackThread::OutputTrack::stop() 4312{ 4313 Track::stop(); 4314 clearBufferQueue(); 4315 mOutBuffer.frameCount = 0; 4316 mActive = false; 4317} 4318 4319bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4320{ 4321 Buffer *pInBuffer; 4322 Buffer inBuffer; 4323 uint32_t channelCount = mChannelCount; 4324 bool outputBufferFull = false; 4325 inBuffer.frameCount = frames; 4326 inBuffer.i16 = data; 4327 4328 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4329 4330 if (!mActive && frames != 0) { 4331 start(0); 4332 sp<ThreadBase> thread = mThread.promote(); 4333 if (thread != 0) { 4334 MixerThread *mixerThread = (MixerThread *)thread.get(); 4335 if (mCblk->frameCount > frames){ 4336 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4337 uint32_t startFrames = (mCblk->frameCount - frames); 4338 pInBuffer = new Buffer; 4339 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4340 pInBuffer->frameCount = startFrames; 4341 pInBuffer->i16 = pInBuffer->mBuffer; 4342 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4343 mBufferQueue.add(pInBuffer); 4344 } else { 4345 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4346 } 4347 } 4348 } 4349 } 4350 4351 while (waitTimeLeftMs) { 4352 // First write pending buffers, then new data 4353 if (mBufferQueue.size()) { 4354 pInBuffer = mBufferQueue.itemAt(0); 4355 } else { 4356 pInBuffer = &inBuffer; 4357 } 4358 4359 if (pInBuffer->frameCount == 0) { 4360 break; 4361 } 4362 4363 if (mOutBuffer.frameCount == 0) { 4364 mOutBuffer.frameCount = pInBuffer->frameCount; 4365 nsecs_t startTime = systemTime(); 4366 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4367 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4368 outputBufferFull = true; 4369 break; 4370 } 4371 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4372 if (waitTimeLeftMs >= waitTimeMs) { 4373 waitTimeLeftMs -= waitTimeMs; 4374 } else { 4375 waitTimeLeftMs = 0; 4376 } 4377 } 4378 4379 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4380 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4381 mCblk->stepUser(outFrames); 4382 pInBuffer->frameCount -= outFrames; 4383 pInBuffer->i16 += outFrames * channelCount; 4384 mOutBuffer.frameCount -= outFrames; 4385 mOutBuffer.i16 += outFrames * channelCount; 4386 4387 if (pInBuffer->frameCount == 0) { 4388 if (mBufferQueue.size()) { 4389 mBufferQueue.removeAt(0); 4390 delete [] pInBuffer->mBuffer; 4391 delete pInBuffer; 4392 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4393 } else { 4394 break; 4395 } 4396 } 4397 } 4398 4399 // If we could not write all frames, allocate a buffer and queue it for next time. 4400 if (inBuffer.frameCount) { 4401 sp<ThreadBase> thread = mThread.promote(); 4402 if (thread != 0 && !thread->standby()) { 4403 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4404 pInBuffer = new Buffer; 4405 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4406 pInBuffer->frameCount = inBuffer.frameCount; 4407 pInBuffer->i16 = pInBuffer->mBuffer; 4408 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4409 mBufferQueue.add(pInBuffer); 4410 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4411 } else { 4412 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4413 } 4414 } 4415 } 4416 4417 // Calling write() with a 0 length buffer, means that no more data will be written: 4418 // If no more buffers are pending, fill output track buffer to make sure it is started 4419 // by output mixer. 4420 if (frames == 0 && mBufferQueue.size() == 0) { 4421 if (mCblk->user < mCblk->frameCount) { 4422 frames = mCblk->frameCount - mCblk->user; 4423 pInBuffer = new Buffer; 4424 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4425 pInBuffer->frameCount = frames; 4426 pInBuffer->i16 = pInBuffer->mBuffer; 4427 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4428 mBufferQueue.add(pInBuffer); 4429 } else if (mActive) { 4430 stop(); 4431 } 4432 } 4433 4434 return outputBufferFull; 4435} 4436 4437status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4438{ 4439 int active; 4440 status_t result; 4441 audio_track_cblk_t* cblk = mCblk; 4442 uint32_t framesReq = buffer->frameCount; 4443 4444// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4445 buffer->frameCount = 0; 4446 4447 uint32_t framesAvail = cblk->framesAvailable(); 4448 4449 4450 if (framesAvail == 0) { 4451 Mutex::Autolock _l(cblk->lock); 4452 goto start_loop_here; 4453 while (framesAvail == 0) { 4454 active = mActive; 4455 if (CC_UNLIKELY(!active)) { 4456 ALOGV("Not active and NO_MORE_BUFFERS"); 4457 return NO_MORE_BUFFERS; 4458 } 4459 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4460 if (result != NO_ERROR) { 4461 return NO_MORE_BUFFERS; 4462 } 4463 // read the server count again 4464 start_loop_here: 4465 framesAvail = cblk->framesAvailable_l(); 4466 } 4467 } 4468 4469// if (framesAvail < framesReq) { 4470// return NO_MORE_BUFFERS; 4471// } 4472 4473 if (framesReq > framesAvail) { 4474 framesReq = framesAvail; 4475 } 4476 4477 uint32_t u = cblk->user; 4478 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4479 4480 if (u + framesReq > bufferEnd) { 4481 framesReq = bufferEnd - u; 4482 } 4483 4484 buffer->frameCount = framesReq; 4485 buffer->raw = (void *)cblk->buffer(u); 4486 return NO_ERROR; 4487} 4488 4489 4490void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4491{ 4492 size_t size = mBufferQueue.size(); 4493 4494 for (size_t i = 0; i < size; i++) { 4495 Buffer *pBuffer = mBufferQueue.itemAt(i); 4496 delete [] pBuffer->mBuffer; 4497 delete pBuffer; 4498 } 4499 mBufferQueue.clear(); 4500} 4501 4502// ---------------------------------------------------------------------------- 4503 4504AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4505 : RefBase(), 4506 mAudioFlinger(audioFlinger), 4507 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4508 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4509 mPid(pid), 4510 mTimedTrackCount(0) 4511{ 4512 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4513} 4514 4515// Client destructor must be called with AudioFlinger::mLock held 4516AudioFlinger::Client::~Client() 4517{ 4518 mAudioFlinger->removeClient_l(mPid); 4519} 4520 4521sp<MemoryDealer> AudioFlinger::Client::heap() const 4522{ 4523 return mMemoryDealer; 4524} 4525 4526// Reserve one of the limited slots for a timed audio track associated 4527// with this client 4528bool AudioFlinger::Client::reserveTimedTrack() 4529{ 4530 const int kMaxTimedTracksPerClient = 4; 4531 4532 Mutex::Autolock _l(mTimedTrackLock); 4533 4534 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4535 ALOGW("can not create timed track - pid %d has exceeded the limit", 4536 mPid); 4537 return false; 4538 } 4539 4540 mTimedTrackCount++; 4541 return true; 4542} 4543 4544// Release a slot for a timed audio track 4545void AudioFlinger::Client::releaseTimedTrack() 4546{ 4547 Mutex::Autolock _l(mTimedTrackLock); 4548 mTimedTrackCount--; 4549} 4550 4551// ---------------------------------------------------------------------------- 4552 4553AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4554 const sp<IAudioFlingerClient>& client, 4555 pid_t pid) 4556 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4557{ 4558} 4559 4560AudioFlinger::NotificationClient::~NotificationClient() 4561{ 4562} 4563 4564void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4565{ 4566 sp<NotificationClient> keep(this); 4567 mAudioFlinger->removeNotificationClient(mPid); 4568} 4569 4570// ---------------------------------------------------------------------------- 4571 4572AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4573 : BnAudioTrack(), 4574 mTrack(track) 4575{ 4576} 4577 4578AudioFlinger::TrackHandle::~TrackHandle() { 4579 // just stop the track on deletion, associated resources 4580 // will be freed from the main thread once all pending buffers have 4581 // been played. Unless it's not in the active track list, in which 4582 // case we free everything now... 4583 mTrack->destroy(); 4584} 4585 4586sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4587 return mTrack->getCblk(); 4588} 4589 4590status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4591 return mTrack->start(tid); 4592} 4593 4594void AudioFlinger::TrackHandle::stop() { 4595 mTrack->stop(); 4596} 4597 4598void AudioFlinger::TrackHandle::flush() { 4599 mTrack->flush(); 4600} 4601 4602void AudioFlinger::TrackHandle::mute(bool e) { 4603 mTrack->mute(e); 4604} 4605 4606void AudioFlinger::TrackHandle::pause() { 4607 mTrack->pause(); 4608} 4609 4610status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4611{ 4612 return mTrack->attachAuxEffect(EffectId); 4613} 4614 4615status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4616 sp<IMemory>* buffer) { 4617 if (!mTrack->isTimedTrack()) 4618 return INVALID_OPERATION; 4619 4620 PlaybackThread::TimedTrack* tt = 4621 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4622 return tt->allocateTimedBuffer(size, buffer); 4623} 4624 4625status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4626 int64_t pts) { 4627 if (!mTrack->isTimedTrack()) 4628 return INVALID_OPERATION; 4629 4630 PlaybackThread::TimedTrack* tt = 4631 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4632 return tt->queueTimedBuffer(buffer, pts); 4633} 4634 4635status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4636 const LinearTransform& xform, int target) { 4637 4638 if (!mTrack->isTimedTrack()) 4639 return INVALID_OPERATION; 4640 4641 PlaybackThread::TimedTrack* tt = 4642 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4643 return tt->setMediaTimeTransform( 4644 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4645} 4646 4647status_t AudioFlinger::TrackHandle::onTransact( 4648 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4649{ 4650 return BnAudioTrack::onTransact(code, data, reply, flags); 4651} 4652 4653// ---------------------------------------------------------------------------- 4654 4655sp<IAudioRecord> AudioFlinger::openRecord( 4656 pid_t pid, 4657 audio_io_handle_t input, 4658 uint32_t sampleRate, 4659 audio_format_t format, 4660 uint32_t channelMask, 4661 int frameCount, 4662 // FIXME dead, remove from IAudioFlinger 4663 uint32_t flags, 4664 int *sessionId, 4665 status_t *status) 4666{ 4667 sp<RecordThread::RecordTrack> recordTrack; 4668 sp<RecordHandle> recordHandle; 4669 sp<Client> client; 4670 status_t lStatus; 4671 RecordThread *thread; 4672 size_t inFrameCount; 4673 int lSessionId; 4674 4675 // check calling permissions 4676 if (!recordingAllowed()) { 4677 lStatus = PERMISSION_DENIED; 4678 goto Exit; 4679 } 4680 4681 // add client to list 4682 { // scope for mLock 4683 Mutex::Autolock _l(mLock); 4684 thread = checkRecordThread_l(input); 4685 if (thread == NULL) { 4686 lStatus = BAD_VALUE; 4687 goto Exit; 4688 } 4689 4690 client = registerPid_l(pid); 4691 4692 // If no audio session id is provided, create one here 4693 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4694 lSessionId = *sessionId; 4695 } else { 4696 lSessionId = nextUniqueId(); 4697 if (sessionId != NULL) { 4698 *sessionId = lSessionId; 4699 } 4700 } 4701 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4702 recordTrack = thread->createRecordTrack_l(client, 4703 sampleRate, 4704 format, 4705 channelMask, 4706 frameCount, 4707 lSessionId, 4708 &lStatus); 4709 } 4710 if (lStatus != NO_ERROR) { 4711 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4712 // destructor is called by the TrackBase destructor with mLock held 4713 client.clear(); 4714 recordTrack.clear(); 4715 goto Exit; 4716 } 4717 4718 // return to handle to client 4719 recordHandle = new RecordHandle(recordTrack); 4720 lStatus = NO_ERROR; 4721 4722Exit: 4723 if (status) { 4724 *status = lStatus; 4725 } 4726 return recordHandle; 4727} 4728 4729// ---------------------------------------------------------------------------- 4730 4731AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4732 : BnAudioRecord(), 4733 mRecordTrack(recordTrack) 4734{ 4735} 4736 4737AudioFlinger::RecordHandle::~RecordHandle() { 4738 stop(); 4739} 4740 4741sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4742 return mRecordTrack->getCblk(); 4743} 4744 4745status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4746 ALOGV("RecordHandle::start()"); 4747 return mRecordTrack->start(tid); 4748} 4749 4750void AudioFlinger::RecordHandle::stop() { 4751 ALOGV("RecordHandle::stop()"); 4752 mRecordTrack->stop(); 4753} 4754 4755status_t AudioFlinger::RecordHandle::onTransact( 4756 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4757{ 4758 return BnAudioRecord::onTransact(code, data, reply, flags); 4759} 4760 4761// ---------------------------------------------------------------------------- 4762 4763AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4764 AudioStreamIn *input, 4765 uint32_t sampleRate, 4766 uint32_t channels, 4767 audio_io_handle_t id, 4768 uint32_t device) : 4769 ThreadBase(audioFlinger, id, device, RECORD), 4770 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4771 // mRsmpInIndex and mInputBytes set by readInputParameters() 4772 mReqChannelCount(popcount(channels)), 4773 mReqSampleRate(sampleRate) 4774 // mBytesRead is only meaningful while active, and so is cleared in start() 4775 // (but might be better to also clear here for dump?) 4776{ 4777 snprintf(mName, kNameLength, "AudioIn_%d", id); 4778 4779 readInputParameters(); 4780} 4781 4782 4783AudioFlinger::RecordThread::~RecordThread() 4784{ 4785 delete[] mRsmpInBuffer; 4786 delete mResampler; 4787 delete[] mRsmpOutBuffer; 4788} 4789 4790void AudioFlinger::RecordThread::onFirstRef() 4791{ 4792 run(mName, PRIORITY_URGENT_AUDIO); 4793} 4794 4795status_t AudioFlinger::RecordThread::readyToRun() 4796{ 4797 status_t status = initCheck(); 4798 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4799 return status; 4800} 4801 4802bool AudioFlinger::RecordThread::threadLoop() 4803{ 4804 AudioBufferProvider::Buffer buffer; 4805 sp<RecordTrack> activeTrack; 4806 Vector< sp<EffectChain> > effectChains; 4807 4808 nsecs_t lastWarning = 0; 4809 4810 acquireWakeLock(); 4811 4812 // start recording 4813 while (!exitPending()) { 4814 4815 processConfigEvents(); 4816 4817 { // scope for mLock 4818 Mutex::Autolock _l(mLock); 4819 checkForNewParameters_l(); 4820 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4821 if (!mStandby) { 4822 mInput->stream->common.standby(&mInput->stream->common); 4823 mStandby = true; 4824 } 4825 4826 if (exitPending()) break; 4827 4828 releaseWakeLock_l(); 4829 ALOGV("RecordThread: loop stopping"); 4830 // go to sleep 4831 mWaitWorkCV.wait(mLock); 4832 ALOGV("RecordThread: loop starting"); 4833 acquireWakeLock_l(); 4834 continue; 4835 } 4836 if (mActiveTrack != 0) { 4837 if (mActiveTrack->mState == TrackBase::PAUSING) { 4838 if (!mStandby) { 4839 mInput->stream->common.standby(&mInput->stream->common); 4840 mStandby = true; 4841 } 4842 mActiveTrack.clear(); 4843 mStartStopCond.broadcast(); 4844 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4845 if (mReqChannelCount != mActiveTrack->channelCount()) { 4846 mActiveTrack.clear(); 4847 mStartStopCond.broadcast(); 4848 } else if (mBytesRead != 0) { 4849 // record start succeeds only if first read from audio input 4850 // succeeds 4851 if (mBytesRead > 0) { 4852 mActiveTrack->mState = TrackBase::ACTIVE; 4853 } else { 4854 mActiveTrack.clear(); 4855 } 4856 mStartStopCond.broadcast(); 4857 } 4858 mStandby = false; 4859 } 4860 } 4861 lockEffectChains_l(effectChains); 4862 } 4863 4864 if (mActiveTrack != 0) { 4865 if (mActiveTrack->mState != TrackBase::ACTIVE && 4866 mActiveTrack->mState != TrackBase::RESUMING) { 4867 unlockEffectChains(effectChains); 4868 usleep(kRecordThreadSleepUs); 4869 continue; 4870 } 4871 for (size_t i = 0; i < effectChains.size(); i ++) { 4872 effectChains[i]->process_l(); 4873 } 4874 4875 buffer.frameCount = mFrameCount; 4876 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4877 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4878 size_t framesOut = buffer.frameCount; 4879 if (mResampler == NULL) { 4880 // no resampling 4881 while (framesOut) { 4882 size_t framesIn = mFrameCount - mRsmpInIndex; 4883 if (framesIn) { 4884 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4885 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4886 if (framesIn > framesOut) 4887 framesIn = framesOut; 4888 mRsmpInIndex += framesIn; 4889 framesOut -= framesIn; 4890 if ((int)mChannelCount == mReqChannelCount || 4891 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4892 memcpy(dst, src, framesIn * mFrameSize); 4893 } else { 4894 int16_t *src16 = (int16_t *)src; 4895 int16_t *dst16 = (int16_t *)dst; 4896 if (mChannelCount == 1) { 4897 while (framesIn--) { 4898 *dst16++ = *src16; 4899 *dst16++ = *src16++; 4900 } 4901 } else { 4902 while (framesIn--) { 4903 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4904 src16 += 2; 4905 } 4906 } 4907 } 4908 } 4909 if (framesOut && mFrameCount == mRsmpInIndex) { 4910 if (framesOut == mFrameCount && 4911 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4912 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4913 framesOut = 0; 4914 } else { 4915 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4916 mRsmpInIndex = 0; 4917 } 4918 if (mBytesRead < 0) { 4919 ALOGE("Error reading audio input"); 4920 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4921 // Force input into standby so that it tries to 4922 // recover at next read attempt 4923 mInput->stream->common.standby(&mInput->stream->common); 4924 usleep(kRecordThreadSleepUs); 4925 } 4926 mRsmpInIndex = mFrameCount; 4927 framesOut = 0; 4928 buffer.frameCount = 0; 4929 } 4930 } 4931 } 4932 } else { 4933 // resampling 4934 4935 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4936 // alter output frame count as if we were expecting stereo samples 4937 if (mChannelCount == 1 && mReqChannelCount == 1) { 4938 framesOut >>= 1; 4939 } 4940 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4941 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4942 // are 32 bit aligned which should be always true. 4943 if (mChannelCount == 2 && mReqChannelCount == 1) { 4944 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4945 // the resampler always outputs stereo samples: do post stereo to mono conversion 4946 int16_t *src = (int16_t *)mRsmpOutBuffer; 4947 int16_t *dst = buffer.i16; 4948 while (framesOut--) { 4949 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4950 src += 2; 4951 } 4952 } else { 4953 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4954 } 4955 4956 } 4957 mActiveTrack->releaseBuffer(&buffer); 4958 mActiveTrack->overflow(); 4959 } 4960 // client isn't retrieving buffers fast enough 4961 else { 4962 if (!mActiveTrack->setOverflow()) { 4963 nsecs_t now = systemTime(); 4964 if ((now - lastWarning) > kWarningThrottleNs) { 4965 ALOGW("RecordThread: buffer overflow"); 4966 lastWarning = now; 4967 } 4968 } 4969 // Release the processor for a while before asking for a new buffer. 4970 // This will give the application more chance to read from the buffer and 4971 // clear the overflow. 4972 usleep(kRecordThreadSleepUs); 4973 } 4974 } 4975 // enable changes in effect chain 4976 unlockEffectChains(effectChains); 4977 effectChains.clear(); 4978 } 4979 4980 if (!mStandby) { 4981 mInput->stream->common.standby(&mInput->stream->common); 4982 } 4983 mActiveTrack.clear(); 4984 4985 mStartStopCond.broadcast(); 4986 4987 releaseWakeLock(); 4988 4989 ALOGV("RecordThread %p exiting", this); 4990 return false; 4991} 4992 4993 4994sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4995 const sp<AudioFlinger::Client>& client, 4996 uint32_t sampleRate, 4997 audio_format_t format, 4998 int channelMask, 4999 int frameCount, 5000 int sessionId, 5001 status_t *status) 5002{ 5003 sp<RecordTrack> track; 5004 status_t lStatus; 5005 5006 lStatus = initCheck(); 5007 if (lStatus != NO_ERROR) { 5008 ALOGE("Audio driver not initialized."); 5009 goto Exit; 5010 } 5011 5012 { // scope for mLock 5013 Mutex::Autolock _l(mLock); 5014 5015 track = new RecordTrack(this, client, sampleRate, 5016 format, channelMask, frameCount, sessionId); 5017 5018 if (track->getCblk() == 0) { 5019 lStatus = NO_MEMORY; 5020 goto Exit; 5021 } 5022 5023 mTrack = track.get(); 5024 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5025 bool suspend = audio_is_bluetooth_sco_device( 5026 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5027 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5028 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5029 } 5030 lStatus = NO_ERROR; 5031 5032Exit: 5033 if (status) { 5034 *status = lStatus; 5035 } 5036 return track; 5037} 5038 5039status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5040{ 5041 ALOGV("RecordThread::start tid=%d", tid); 5042 sp <ThreadBase> strongMe = this; 5043 status_t status = NO_ERROR; 5044 { 5045 AutoMutex lock(mLock); 5046 if (mActiveTrack != 0) { 5047 if (recordTrack != mActiveTrack.get()) { 5048 status = -EBUSY; 5049 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5050 mActiveTrack->mState = TrackBase::ACTIVE; 5051 } 5052 return status; 5053 } 5054 5055 recordTrack->mState = TrackBase::IDLE; 5056 mActiveTrack = recordTrack; 5057 mLock.unlock(); 5058 status_t status = AudioSystem::startInput(mId); 5059 mLock.lock(); 5060 if (status != NO_ERROR) { 5061 mActiveTrack.clear(); 5062 return status; 5063 } 5064 mRsmpInIndex = mFrameCount; 5065 mBytesRead = 0; 5066 if (mResampler != NULL) { 5067 mResampler->reset(); 5068 } 5069 mActiveTrack->mState = TrackBase::RESUMING; 5070 // signal thread to start 5071 ALOGV("Signal record thread"); 5072 mWaitWorkCV.signal(); 5073 // do not wait for mStartStopCond if exiting 5074 if (exitPending()) { 5075 mActiveTrack.clear(); 5076 status = INVALID_OPERATION; 5077 goto startError; 5078 } 5079 mStartStopCond.wait(mLock); 5080 if (mActiveTrack == 0) { 5081 ALOGV("Record failed to start"); 5082 status = BAD_VALUE; 5083 goto startError; 5084 } 5085 ALOGV("Record started OK"); 5086 return status; 5087 } 5088startError: 5089 AudioSystem::stopInput(mId); 5090 return status; 5091} 5092 5093void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5094 ALOGV("RecordThread::stop"); 5095 sp <ThreadBase> strongMe = this; 5096 { 5097 AutoMutex lock(mLock); 5098 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5099 mActiveTrack->mState = TrackBase::PAUSING; 5100 // do not wait for mStartStopCond if exiting 5101 if (exitPending()) { 5102 return; 5103 } 5104 mStartStopCond.wait(mLock); 5105 // if we have been restarted, recordTrack == mActiveTrack.get() here 5106 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5107 mLock.unlock(); 5108 AudioSystem::stopInput(mId); 5109 mLock.lock(); 5110 ALOGV("Record stopped OK"); 5111 } 5112 } 5113 } 5114} 5115 5116status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5117{ 5118 const size_t SIZE = 256; 5119 char buffer[SIZE]; 5120 String8 result; 5121 5122 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5123 result.append(buffer); 5124 5125 if (mActiveTrack != 0) { 5126 result.append("Active Track:\n"); 5127 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5128 mActiveTrack->dump(buffer, SIZE); 5129 result.append(buffer); 5130 5131 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5132 result.append(buffer); 5133 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5134 result.append(buffer); 5135 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5136 result.append(buffer); 5137 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5138 result.append(buffer); 5139 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5140 result.append(buffer); 5141 5142 5143 } else { 5144 result.append("No record client\n"); 5145 } 5146 write(fd, result.string(), result.size()); 5147 5148 dumpBase(fd, args); 5149 dumpEffectChains(fd, args); 5150 5151 return NO_ERROR; 5152} 5153 5154status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5155{ 5156 size_t framesReq = buffer->frameCount; 5157 size_t framesReady = mFrameCount - mRsmpInIndex; 5158 int channelCount; 5159 5160 if (framesReady == 0) { 5161 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5162 if (mBytesRead < 0) { 5163 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5164 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5165 // Force input into standby so that it tries to 5166 // recover at next read attempt 5167 mInput->stream->common.standby(&mInput->stream->common); 5168 usleep(kRecordThreadSleepUs); 5169 } 5170 buffer->raw = NULL; 5171 buffer->frameCount = 0; 5172 return NOT_ENOUGH_DATA; 5173 } 5174 mRsmpInIndex = 0; 5175 framesReady = mFrameCount; 5176 } 5177 5178 if (framesReq > framesReady) { 5179 framesReq = framesReady; 5180 } 5181 5182 if (mChannelCount == 1 && mReqChannelCount == 2) { 5183 channelCount = 1; 5184 } else { 5185 channelCount = 2; 5186 } 5187 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5188 buffer->frameCount = framesReq; 5189 return NO_ERROR; 5190} 5191 5192void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5193{ 5194 mRsmpInIndex += buffer->frameCount; 5195 buffer->frameCount = 0; 5196} 5197 5198bool AudioFlinger::RecordThread::checkForNewParameters_l() 5199{ 5200 bool reconfig = false; 5201 5202 while (!mNewParameters.isEmpty()) { 5203 status_t status = NO_ERROR; 5204 String8 keyValuePair = mNewParameters[0]; 5205 AudioParameter param = AudioParameter(keyValuePair); 5206 int value; 5207 audio_format_t reqFormat = mFormat; 5208 int reqSamplingRate = mReqSampleRate; 5209 int reqChannelCount = mReqChannelCount; 5210 5211 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5212 reqSamplingRate = value; 5213 reconfig = true; 5214 } 5215 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5216 reqFormat = (audio_format_t) value; 5217 reconfig = true; 5218 } 5219 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5220 reqChannelCount = popcount(value); 5221 reconfig = true; 5222 } 5223 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5224 // do not accept frame count changes if tracks are open as the track buffer 5225 // size depends on frame count and correct behavior would not be guaranteed 5226 // if frame count is changed after track creation 5227 if (mActiveTrack != 0) { 5228 status = INVALID_OPERATION; 5229 } else { 5230 reconfig = true; 5231 } 5232 } 5233 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5234 // forward device change to effects that have requested to be 5235 // aware of attached audio device. 5236 for (size_t i = 0; i < mEffectChains.size(); i++) { 5237 mEffectChains[i]->setDevice_l(value); 5238 } 5239 // store input device and output device but do not forward output device to audio HAL. 5240 // Note that status is ignored by the caller for output device 5241 // (see AudioFlinger::setParameters() 5242 if (value & AUDIO_DEVICE_OUT_ALL) { 5243 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5244 status = BAD_VALUE; 5245 } else { 5246 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5247 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5248 if (mTrack != NULL) { 5249 bool suspend = audio_is_bluetooth_sco_device( 5250 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5251 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5252 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5253 } 5254 } 5255 mDevice |= (uint32_t)value; 5256 } 5257 if (status == NO_ERROR) { 5258 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5259 if (status == INVALID_OPERATION) { 5260 mInput->stream->common.standby(&mInput->stream->common); 5261 status = mInput->stream->common.set_parameters(&mInput->stream->common, 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)) < 3) && 5269 (reqChannelCount < 3)) { 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 < 3 && mReqChannelCount < 3) 5347 { 5348 int channelCount; 5349 // optmization: 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 uint32_t flags) 5433{ 5434 status_t status; 5435 PlaybackThread *thread = NULL; 5436 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5437 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5438 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5439 uint32_t channels = pChannels ? *pChannels : 0; 5440 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5441 audio_stream_out_t *outStream; 5442 audio_hw_device_t *outHwDev; 5443 5444 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5445 pDevices ? *pDevices : 0, 5446 samplingRate, 5447 format, 5448 channels, 5449 flags); 5450 5451 if (pDevices == NULL || *pDevices == 0) { 5452 return 0; 5453 } 5454 5455 Mutex::Autolock _l(mLock); 5456 5457 outHwDev = findSuitableHwDev_l(*pDevices); 5458 if (outHwDev == NULL) 5459 return 0; 5460 5461 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5462 &channels, &samplingRate, &outStream); 5463 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5464 outStream, 5465 samplingRate, 5466 format, 5467 channels, 5468 status); 5469 5470 mHardwareStatus = AUDIO_HW_IDLE; 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 assert(out != 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) < 3) && (popcount(reqChannels) < 3)) { 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 assert(in != 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->sessionid == audioSession && ref->pid == caller) { 5743 ref->cnt++; 5744 ALOGV(" incremented refcount to %d", ref->cnt); 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->sessionid == audioSession && ref->pid == caller) { 5761 ref->cnt--; 5762 ALOGV(" decremented refcount to %d", ref->cnt); 5763 if (ref->cnt == 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->sessionid == sessionid) { 5809 ALOGV(" session %d still exists for %d with %d refs", 5810 sessionid, ref->pid, ref->cnt); 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) { 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) { 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