AudioFlinger.cpp revision c5c49398584f2399af905a931e556ed6e0a29cd4
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#undef ADD_BATTERY_DATA 41 42#ifdef ADD_BATTERY_DATA 43#include <media/IMediaPlayerService.h> 44#include <media/IMediaDeathNotifier.h> 45#endif 46 47#include <private/media/AudioTrackShared.h> 48#include <private/media/AudioEffectShared.h> 49 50#include <system/audio.h> 51#include <hardware/audio.h> 52 53#include "AudioMixer.h" 54#include "AudioFlinger.h" 55#include "ServiceUtilities.h" 56 57#include <media/EffectsFactoryApi.h> 58#include <audio_effects/effect_visualizer.h> 59#include <audio_effects/effect_ns.h> 60#include <audio_effects/effect_aec.h> 61 62#include <audio_utils/primitives.h> 63 64#include <powermanager/PowerManager.h> 65 66// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 67#ifdef DEBUG_CPU_USAGE 68#include <cpustats/CentralTendencyStatistics.h> 69#include <cpustats/ThreadCpuUsage.h> 70#endif 71 72#include <common_time/cc_helper.h> 73#include <common_time/local_clock.h> 74 75// ---------------------------------------------------------------------------- 76 77 78namespace android { 79 80static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 81static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 82 83static const float MAX_GAIN = 4096.0f; 84static const uint32_t MAX_GAIN_INT = 0x1000; 85 86// retry counts for buffer fill timeout 87// 50 * ~20msecs = 1 second 88static const int8_t kMaxTrackRetries = 50; 89static const int8_t kMaxTrackStartupRetries = 50; 90// allow less retry attempts on direct output thread. 91// direct outputs can be a scarce resource in audio hardware and should 92// be released as quickly as possible. 93static const int8_t kMaxTrackRetriesDirect = 2; 94 95static const int kDumpLockRetries = 50; 96static const int kDumpLockSleepUs = 20000; 97 98// don't warn about blocked writes or record buffer overflows more often than this 99static const nsecs_t kWarningThrottleNs = seconds(5); 100 101// RecordThread loop sleep time upon application overrun or audio HAL read error 102static const int kRecordThreadSleepUs = 5000; 103 104// maximum time to wait for setParameters to complete 105static const nsecs_t kSetParametersTimeoutNs = seconds(2); 106 107// minimum sleep time for the mixer thread loop when tracks are active but in underrun 108static const uint32_t kMinThreadSleepTimeUs = 5000; 109// maximum divider applied to the active sleep time in the mixer thread loop 110static const uint32_t kMaxThreadSleepTimeShift = 2; 111 112nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 113 114// ---------------------------------------------------------------------------- 115 116#ifdef ADD_BATTERY_DATA 117// To collect the amplifier usage 118static void addBatteryData(uint32_t params) { 119 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 120 if (service == NULL) { 121 // it already logged 122 return; 123 } 124 125 service->addBatteryData(params); 126} 127#endif 128 129static int load_audio_interface(const char *if_name, const hw_module_t **mod, 130 audio_hw_device_t **dev) 131{ 132 int rc; 133 134 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 135 if (rc) 136 goto out; 137 138 rc = audio_hw_device_open(*mod, dev); 139 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 140 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 141 if (rc) 142 goto out; 143 144 return 0; 145 146out: 147 *mod = NULL; 148 *dev = NULL; 149 return rc; 150} 151 152static const char * const audio_interfaces[] = { 153 "primary", 154 "a2dp", 155 "usb", 156}; 157#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 158 159// ---------------------------------------------------------------------------- 160 161AudioFlinger::AudioFlinger() 162 : BnAudioFlinger(), 163 mPrimaryHardwareDev(NULL), 164 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 165 mMasterVolume(1.0f), 166 mMasterVolumeSupportLvl(MVS_NONE), 167 mMasterMute(false), 168 mNextUniqueId(1), 169 mMode(AUDIO_MODE_INVALID), 170 mBtNrecIsOff(false) 171{ 172} 173 174void AudioFlinger::onFirstRef() 175{ 176 int rc = 0; 177 178 Mutex::Autolock _l(mLock); 179 180 /* TODO: move all this work into an Init() function */ 181 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 182 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 183 uint32_t int_val; 184 if (1 == sscanf(val_str, "%u", &int_val)) { 185 mStandbyTimeInNsecs = milliseconds(int_val); 186 ALOGI("Using %u mSec as standby time.", int_val); 187 } else { 188 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 189 ALOGI("Using default %u mSec as standby time.", 190 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 191 } 192 } 193 194 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 195 const hw_module_t *mod; 196 audio_hw_device_t *dev; 197 198 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 199 if (rc) 200 continue; 201 202 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 203 mod->name, mod->id); 204 mAudioHwDevs.push(dev); 205 206 if (mPrimaryHardwareDev == NULL) { 207 mPrimaryHardwareDev = dev; 208 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 209 mod->name, mod->id, audio_interfaces[i]); 210 } 211 } 212 213 if (mPrimaryHardwareDev == NULL) { 214 ALOGE("Primary audio interface not found"); 215 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 216 } 217 218 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 219 // primary HW dev is selected can change so these conditions might not always be equivalent. 220 // When that happens, re-visit all the code that assumes this. 221 222 AutoMutex lock(mHardwareLock); 223 224 // Determine the level of master volume support the primary audio HAL has, 225 // and set the initial master volume at the same time. 226 float initialVolume = 1.0; 227 mMasterVolumeSupportLvl = MVS_NONE; 228 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 229 audio_hw_device_t *dev = mPrimaryHardwareDev; 230 231 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 232 if ((NULL != dev->get_master_volume) && 233 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 234 mMasterVolumeSupportLvl = MVS_FULL; 235 } else { 236 mMasterVolumeSupportLvl = MVS_SETONLY; 237 initialVolume = 1.0; 238 } 239 240 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 241 if ((NULL == dev->set_master_volume) || 242 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 243 mMasterVolumeSupportLvl = MVS_NONE; 244 } 245 mHardwareStatus = AUDIO_HW_IDLE; 246 } 247 248 // Set the mode for each audio HAL, and try to set the initial volume (if 249 // supported) for all of the non-primary audio HALs. 250 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 251 audio_hw_device_t *dev = mAudioHwDevs[i]; 252 253 mHardwareStatus = AUDIO_HW_INIT; 254 rc = dev->init_check(dev); 255 mHardwareStatus = AUDIO_HW_IDLE; 256 if (rc == 0) { 257 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 258 mHardwareStatus = AUDIO_HW_SET_MODE; 259 dev->set_mode(dev, mMode); 260 261 if ((dev != mPrimaryHardwareDev) && 262 (NULL != dev->set_master_volume)) { 263 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 264 dev->set_master_volume(dev, initialVolume); 265 } 266 267 mHardwareStatus = AUDIO_HW_IDLE; 268 } 269 } 270 271 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 272 ? initialVolume 273 : 1.0; 274 mMasterVolume = initialVolume; 275 mHardwareStatus = AUDIO_HW_IDLE; 276} 277 278AudioFlinger::~AudioFlinger() 279{ 280 281 while (!mRecordThreads.isEmpty()) { 282 // closeInput() will remove first entry from mRecordThreads 283 closeInput(mRecordThreads.keyAt(0)); 284 } 285 while (!mPlaybackThreads.isEmpty()) { 286 // closeOutput() will remove first entry from mPlaybackThreads 287 closeOutput(mPlaybackThreads.keyAt(0)); 288 } 289 290 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 291 // no mHardwareLock needed, as there are no other references to this 292 audio_hw_device_close(mAudioHwDevs[i]); 293 } 294} 295 296audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 297{ 298 /* first matching HW device is returned */ 299 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 300 audio_hw_device_t *dev = mAudioHwDevs[i]; 301 if ((dev->get_supported_devices(dev) & devices) == devices) 302 return dev; 303 } 304 return NULL; 305} 306 307status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Global session refs:\n"); 323 result.append(" session pid count\n"); 324 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 325 AudioSessionRef *r = mAudioSessionRefs[i]; 326 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 327 result.append(buffer); 328 } 329 write(fd, result.string(), result.size()); 330 return NO_ERROR; 331} 332 333 334status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 335{ 336 const size_t SIZE = 256; 337 char buffer[SIZE]; 338 String8 result; 339 hardware_call_state hardwareStatus = mHardwareStatus; 340 341 snprintf(buffer, SIZE, "Hardware status: %d\n" 342 "Standby Time mSec: %u\n", 343 hardwareStatus, 344 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 345 result.append(buffer); 346 write(fd, result.string(), result.size()); 347 return NO_ERROR; 348} 349 350status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 351{ 352 const size_t SIZE = 256; 353 char buffer[SIZE]; 354 String8 result; 355 snprintf(buffer, SIZE, "Permission Denial: " 356 "can't dump AudioFlinger from pid=%d, uid=%d\n", 357 IPCThreadState::self()->getCallingPid(), 358 IPCThreadState::self()->getCallingUid()); 359 result.append(buffer); 360 write(fd, result.string(), result.size()); 361 return NO_ERROR; 362} 363 364static bool tryLock(Mutex& mutex) 365{ 366 bool locked = false; 367 for (int i = 0; i < kDumpLockRetries; ++i) { 368 if (mutex.tryLock() == NO_ERROR) { 369 locked = true; 370 break; 371 } 372 usleep(kDumpLockSleepUs); 373 } 374 return locked; 375} 376 377status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 378{ 379 if (!dumpAllowed()) { 380 dumpPermissionDenial(fd, args); 381 } else { 382 // get state of hardware lock 383 bool hardwareLocked = tryLock(mHardwareLock); 384 if (!hardwareLocked) { 385 String8 result(kHardwareLockedString); 386 write(fd, result.string(), result.size()); 387 } else { 388 mHardwareLock.unlock(); 389 } 390 391 bool locked = tryLock(mLock); 392 393 // failed to lock - AudioFlinger is probably deadlocked 394 if (!locked) { 395 String8 result(kDeadlockedString); 396 write(fd, result.string(), result.size()); 397 } 398 399 dumpClients(fd, args); 400 dumpInternals(fd, args); 401 402 // dump playback threads 403 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 404 mPlaybackThreads.valueAt(i)->dump(fd, args); 405 } 406 407 // dump record threads 408 for (size_t i = 0; i < mRecordThreads.size(); i++) { 409 mRecordThreads.valueAt(i)->dump(fd, args); 410 } 411 412 // dump all hardware devs 413 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 414 audio_hw_device_t *dev = mAudioHwDevs[i]; 415 dev->dump(dev, fd); 416 } 417 if (locked) mLock.unlock(); 418 } 419 return NO_ERROR; 420} 421 422sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 423{ 424 // If pid is already in the mClients wp<> map, then use that entry 425 // (for which promote() is always != 0), otherwise create a new entry and Client. 426 sp<Client> client = mClients.valueFor(pid).promote(); 427 if (client == 0) { 428 client = new Client(this, pid); 429 mClients.add(pid, client); 430 } 431 432 return client; 433} 434 435// IAudioFlinger interface 436 437 438sp<IAudioTrack> AudioFlinger::createTrack( 439 pid_t pid, 440 audio_stream_type_t streamType, 441 uint32_t sampleRate, 442 audio_format_t format, 443 uint32_t channelMask, 444 int frameCount, 445 // FIXME dead, remove from IAudioFlinger 446 uint32_t flags, 447 const sp<IMemory>& sharedBuffer, 448 audio_io_handle_t output, 449 bool isTimed, 450 int *sessionId, 451 status_t *status) 452{ 453 sp<PlaybackThread::Track> track; 454 sp<TrackHandle> trackHandle; 455 sp<Client> client; 456 status_t lStatus; 457 int lSessionId; 458 459 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 460 // but if someone uses binder directly they could bypass that and cause us to crash 461 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 462 ALOGE("createTrack() invalid stream type %d", streamType); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 { 468 Mutex::Autolock _l(mLock); 469 PlaybackThread *thread = checkPlaybackThread_l(output); 470 PlaybackThread *effectThread = NULL; 471 if (thread == NULL) { 472 ALOGE("unknown output thread"); 473 lStatus = BAD_VALUE; 474 goto Exit; 475 } 476 477 client = registerPid_l(pid); 478 479 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 480 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 481 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 482 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 483 if (mPlaybackThreads.keyAt(i) != output) { 484 // prevent same audio session on different output threads 485 uint32_t sessions = t->hasAudioSession(*sessionId); 486 if (sessions & PlaybackThread::TRACK_SESSION) { 487 ALOGE("createTrack() session ID %d already in use", *sessionId); 488 lStatus = BAD_VALUE; 489 goto Exit; 490 } 491 // check if an effect with same session ID is waiting for a track to be created 492 if (sessions & PlaybackThread::EFFECT_SESSION) { 493 effectThread = t.get(); 494 } 495 } 496 } 497 lSessionId = *sessionId; 498 } else { 499 // if no audio session id is provided, create one here 500 lSessionId = nextUniqueId(); 501 if (sessionId != NULL) { 502 *sessionId = lSessionId; 503 } 504 } 505 ALOGV("createTrack() lSessionId: %d", lSessionId); 506 507 track = thread->createTrack_l(client, streamType, sampleRate, format, 508 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 509 510 // move effect chain to this output thread if an effect on same session was waiting 511 // for a track to be created 512 if (lStatus == NO_ERROR && effectThread != NULL) { 513 Mutex::Autolock _dl(thread->mLock); 514 Mutex::Autolock _sl(effectThread->mLock); 515 moveEffectChain_l(lSessionId, effectThread, thread, true); 516 } 517 } 518 if (lStatus == NO_ERROR) { 519 trackHandle = new TrackHandle(track); 520 } else { 521 // remove local strong reference to Client before deleting the Track so that the Client 522 // destructor is called by the TrackBase destructor with mLock held 523 client.clear(); 524 track.clear(); 525 } 526 527Exit: 528 if (status != NULL) { 529 *status = lStatus; 530 } 531 return trackHandle; 532} 533 534uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 535{ 536 Mutex::Autolock _l(mLock); 537 PlaybackThread *thread = checkPlaybackThread_l(output); 538 if (thread == NULL) { 539 ALOGW("sampleRate() unknown thread %d", output); 540 return 0; 541 } 542 return thread->sampleRate(); 543} 544 545int AudioFlinger::channelCount(audio_io_handle_t output) const 546{ 547 Mutex::Autolock _l(mLock); 548 PlaybackThread *thread = checkPlaybackThread_l(output); 549 if (thread == NULL) { 550 ALOGW("channelCount() unknown thread %d", output); 551 return 0; 552 } 553 return thread->channelCount(); 554} 555 556audio_format_t AudioFlinger::format(audio_io_handle_t output) const 557{ 558 Mutex::Autolock _l(mLock); 559 PlaybackThread *thread = checkPlaybackThread_l(output); 560 if (thread == NULL) { 561 ALOGW("format() unknown thread %d", output); 562 return AUDIO_FORMAT_INVALID; 563 } 564 return thread->format(); 565} 566 567size_t AudioFlinger::frameCount(audio_io_handle_t output) const 568{ 569 Mutex::Autolock _l(mLock); 570 PlaybackThread *thread = checkPlaybackThread_l(output); 571 if (thread == NULL) { 572 ALOGW("frameCount() unknown thread %d", output); 573 return 0; 574 } 575 return thread->frameCount(); 576} 577 578uint32_t AudioFlinger::latency(audio_io_handle_t output) const 579{ 580 Mutex::Autolock _l(mLock); 581 PlaybackThread *thread = checkPlaybackThread_l(output); 582 if (thread == NULL) { 583 ALOGW("latency() unknown thread %d", output); 584 return 0; 585 } 586 return thread->latency(); 587} 588 589status_t AudioFlinger::setMasterVolume(float value) 590{ 591 status_t ret = initCheck(); 592 if (ret != NO_ERROR) { 593 return ret; 594 } 595 596 // check calling permissions 597 if (!settingsAllowed()) { 598 return PERMISSION_DENIED; 599 } 600 601 float swmv = value; 602 603 // when hw supports master volume, don't scale in sw mixer 604 if (MVS_NONE != mMasterVolumeSupportLvl) { 605 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 606 AutoMutex lock(mHardwareLock); 607 audio_hw_device_t *dev = mAudioHwDevs[i]; 608 609 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 610 if (NULL != dev->set_master_volume) { 611 dev->set_master_volume(dev, value); 612 } 613 mHardwareStatus = AUDIO_HW_IDLE; 614 } 615 616 swmv = 1.0; 617 } 618 619 Mutex::Autolock _l(mLock); 620 mMasterVolume = value; 621 mMasterVolumeSW = swmv; 622 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 623 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 624 625 return NO_ERROR; 626} 627 628status_t AudioFlinger::setMode(audio_mode_t mode) 629{ 630 status_t ret = initCheck(); 631 if (ret != NO_ERROR) { 632 return ret; 633 } 634 635 // check calling permissions 636 if (!settingsAllowed()) { 637 return PERMISSION_DENIED; 638 } 639 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 640 ALOGW("Illegal value: setMode(%d)", mode); 641 return BAD_VALUE; 642 } 643 644 { // scope for the lock 645 AutoMutex lock(mHardwareLock); 646 mHardwareStatus = AUDIO_HW_SET_MODE; 647 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 648 mHardwareStatus = AUDIO_HW_IDLE; 649 } 650 651 if (NO_ERROR == ret) { 652 Mutex::Autolock _l(mLock); 653 mMode = mode; 654 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 655 mPlaybackThreads.valueAt(i)->setMode(mode); 656 } 657 658 return ret; 659} 660 661status_t AudioFlinger::setMicMute(bool state) 662{ 663 status_t ret = initCheck(); 664 if (ret != NO_ERROR) { 665 return ret; 666 } 667 668 // check calling permissions 669 if (!settingsAllowed()) { 670 return PERMISSION_DENIED; 671 } 672 673 AutoMutex lock(mHardwareLock); 674 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 675 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 676 mHardwareStatus = AUDIO_HW_IDLE; 677 return ret; 678} 679 680bool AudioFlinger::getMicMute() const 681{ 682 status_t ret = initCheck(); 683 if (ret != NO_ERROR) { 684 return false; 685 } 686 687 bool state = AUDIO_MODE_INVALID; 688 AutoMutex lock(mHardwareLock); 689 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 690 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 691 mHardwareStatus = AUDIO_HW_IDLE; 692 return state; 693} 694 695status_t AudioFlinger::setMasterMute(bool muted) 696{ 697 // check calling permissions 698 if (!settingsAllowed()) { 699 return PERMISSION_DENIED; 700 } 701 702 Mutex::Autolock _l(mLock); 703 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 704 mMasterMute = muted; 705 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 706 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 707 708 return NO_ERROR; 709} 710 711float AudioFlinger::masterVolume() const 712{ 713 Mutex::Autolock _l(mLock); 714 return masterVolume_l(); 715} 716 717float AudioFlinger::masterVolumeSW() const 718{ 719 Mutex::Autolock _l(mLock); 720 return masterVolumeSW_l(); 721} 722 723bool AudioFlinger::masterMute() const 724{ 725 Mutex::Autolock _l(mLock); 726 return masterMute_l(); 727} 728 729float AudioFlinger::masterVolume_l() const 730{ 731 if (MVS_FULL == mMasterVolumeSupportLvl) { 732 float ret_val; 733 AutoMutex lock(mHardwareLock); 734 735 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 736 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 737 (NULL != mPrimaryHardwareDev->get_master_volume), 738 "can't get master volume"); 739 740 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 741 mHardwareStatus = AUDIO_HW_IDLE; 742 return ret_val; 743 } 744 745 return mMasterVolume; 746} 747 748status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 749 audio_io_handle_t output) 750{ 751 // check calling permissions 752 if (!settingsAllowed()) { 753 return PERMISSION_DENIED; 754 } 755 756 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 757 ALOGE("setStreamVolume() invalid stream %d", stream); 758 return BAD_VALUE; 759 } 760 761 AutoMutex lock(mLock); 762 PlaybackThread *thread = NULL; 763 if (output) { 764 thread = checkPlaybackThread_l(output); 765 if (thread == NULL) { 766 return BAD_VALUE; 767 } 768 } 769 770 mStreamTypes[stream].volume = value; 771 772 if (thread == NULL) { 773 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 774 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 775 } 776 } else { 777 thread->setStreamVolume(stream, value); 778 } 779 780 return NO_ERROR; 781} 782 783status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 784{ 785 // check calling permissions 786 if (!settingsAllowed()) { 787 return PERMISSION_DENIED; 788 } 789 790 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 791 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 792 ALOGE("setStreamMute() invalid stream %d", stream); 793 return BAD_VALUE; 794 } 795 796 AutoMutex lock(mLock); 797 mStreamTypes[stream].mute = muted; 798 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 799 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 800 801 return NO_ERROR; 802} 803 804float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 805{ 806 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 807 return 0.0f; 808 } 809 810 AutoMutex lock(mLock); 811 float volume; 812 if (output) { 813 PlaybackThread *thread = checkPlaybackThread_l(output); 814 if (thread == NULL) { 815 return 0.0f; 816 } 817 volume = thread->streamVolume(stream); 818 } else { 819 volume = streamVolume_l(stream); 820 } 821 822 return volume; 823} 824 825bool AudioFlinger::streamMute(audio_stream_type_t stream) const 826{ 827 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 828 return true; 829 } 830 831 AutoMutex lock(mLock); 832 return streamMute_l(stream); 833} 834 835status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 836{ 837 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 838 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 839 // check calling permissions 840 if (!settingsAllowed()) { 841 return PERMISSION_DENIED; 842 } 843 844 // ioHandle == 0 means the parameters are global to the audio hardware interface 845 if (ioHandle == 0) { 846 status_t final_result = NO_ERROR; 847 { 848 AutoMutex lock(mHardwareLock); 849 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 850 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 851 audio_hw_device_t *dev = mAudioHwDevs[i]; 852 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 853 final_result = result ?: final_result; 854 } 855 mHardwareStatus = AUDIO_HW_IDLE; 856 } 857 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 858 AudioParameter param = AudioParameter(keyValuePairs); 859 String8 value; 860 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 861 Mutex::Autolock _l(mLock); 862 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 863 if (mBtNrecIsOff != btNrecIsOff) { 864 for (size_t i = 0; i < mRecordThreads.size(); i++) { 865 sp<RecordThread> thread = mRecordThreads.valueAt(i); 866 RecordThread::RecordTrack *track = thread->track(); 867 if (track != NULL) { 868 audio_devices_t device = (audio_devices_t)( 869 thread->device() & AUDIO_DEVICE_IN_ALL); 870 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 871 thread->setEffectSuspended(FX_IID_AEC, 872 suspend, 873 track->sessionId()); 874 thread->setEffectSuspended(FX_IID_NS, 875 suspend, 876 track->sessionId()); 877 } 878 } 879 mBtNrecIsOff = btNrecIsOff; 880 } 881 } 882 return final_result; 883 } 884 885 // hold a strong ref on thread in case closeOutput() or closeInput() is called 886 // and the thread is exited once the lock is released 887 sp<ThreadBase> thread; 888 { 889 Mutex::Autolock _l(mLock); 890 thread = checkPlaybackThread_l(ioHandle); 891 if (thread == NULL) { 892 thread = checkRecordThread_l(ioHandle); 893 } else if (thread == primaryPlaybackThread_l()) { 894 // indicate output device change to all input threads for pre processing 895 AudioParameter param = AudioParameter(keyValuePairs); 896 int value; 897 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 898 (value != 0)) { 899 for (size_t i = 0; i < mRecordThreads.size(); i++) { 900 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 901 } 902 } 903 } 904 } 905 if (thread != 0) { 906 return thread->setParameters(keyValuePairs); 907 } 908 return BAD_VALUE; 909} 910 911String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 912{ 913// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 914// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 915 916 if (ioHandle == 0) { 917 String8 out_s8; 918 919 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 920 char *s; 921 { 922 AutoMutex lock(mHardwareLock); 923 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 924 audio_hw_device_t *dev = mAudioHwDevs[i]; 925 s = dev->get_parameters(dev, keys.string()); 926 mHardwareStatus = AUDIO_HW_IDLE; 927 } 928 out_s8 += String8(s ? s : ""); 929 free(s); 930 } 931 return out_s8; 932 } 933 934 Mutex::Autolock _l(mLock); 935 936 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 937 if (playbackThread != NULL) { 938 return playbackThread->getParameters(keys); 939 } 940 RecordThread *recordThread = checkRecordThread_l(ioHandle); 941 if (recordThread != NULL) { 942 return recordThread->getParameters(keys); 943 } 944 return String8(""); 945} 946 947size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 948{ 949 status_t ret = initCheck(); 950 if (ret != NO_ERROR) { 951 return 0; 952 } 953 954 AutoMutex lock(mHardwareLock); 955 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 956 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 957 mHardwareStatus = AUDIO_HW_IDLE; 958 return size; 959} 960 961unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 962{ 963 if (ioHandle == 0) { 964 return 0; 965 } 966 967 Mutex::Autolock _l(mLock); 968 969 RecordThread *recordThread = checkRecordThread_l(ioHandle); 970 if (recordThread != NULL) { 971 return recordThread->getInputFramesLost(); 972 } 973 return 0; 974} 975 976status_t AudioFlinger::setVoiceVolume(float value) 977{ 978 status_t ret = initCheck(); 979 if (ret != NO_ERROR) { 980 return ret; 981 } 982 983 // check calling permissions 984 if (!settingsAllowed()) { 985 return PERMISSION_DENIED; 986 } 987 988 AutoMutex lock(mHardwareLock); 989 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 990 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 991 mHardwareStatus = AUDIO_HW_IDLE; 992 993 return ret; 994} 995 996status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 997 audio_io_handle_t output) const 998{ 999 status_t status; 1000 1001 Mutex::Autolock _l(mLock); 1002 1003 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1004 if (playbackThread != NULL) { 1005 return playbackThread->getRenderPosition(halFrames, dspFrames); 1006 } 1007 1008 return BAD_VALUE; 1009} 1010 1011void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1012{ 1013 1014 Mutex::Autolock _l(mLock); 1015 1016 pid_t pid = IPCThreadState::self()->getCallingPid(); 1017 if (mNotificationClients.indexOfKey(pid) < 0) { 1018 sp<NotificationClient> notificationClient = new NotificationClient(this, 1019 client, 1020 pid); 1021 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1022 1023 mNotificationClients.add(pid, notificationClient); 1024 1025 sp<IBinder> binder = client->asBinder(); 1026 binder->linkToDeath(notificationClient); 1027 1028 // the config change is always sent from playback or record threads to avoid deadlock 1029 // with AudioSystem::gLock 1030 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1031 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1032 } 1033 1034 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1035 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1036 } 1037 } 1038} 1039 1040void AudioFlinger::removeNotificationClient(pid_t pid) 1041{ 1042 Mutex::Autolock _l(mLock); 1043 1044 mNotificationClients.removeItem(pid); 1045 1046 ALOGV("%d died, releasing its sessions", pid); 1047 size_t num = mAudioSessionRefs.size(); 1048 bool removed = false; 1049 for (size_t i = 0; i< num; ) { 1050 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1051 ALOGV(" pid %d @ %d", ref->mPid, i); 1052 if (ref->mPid == pid) { 1053 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1054 mAudioSessionRefs.removeAt(i); 1055 delete ref; 1056 removed = true; 1057 num--; 1058 } else { 1059 i++; 1060 } 1061 } 1062 if (removed) { 1063 purgeStaleEffects_l(); 1064 } 1065} 1066 1067// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1068void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1069{ 1070 size_t size = mNotificationClients.size(); 1071 for (size_t i = 0; i < size; i++) { 1072 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1073 param2); 1074 } 1075} 1076 1077// removeClient_l() must be called with AudioFlinger::mLock held 1078void AudioFlinger::removeClient_l(pid_t pid) 1079{ 1080 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1081 mClients.removeItem(pid); 1082} 1083 1084 1085// ---------------------------------------------------------------------------- 1086 1087AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1088 uint32_t device, type_t type) 1089 : Thread(false), 1090 mType(type), 1091 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1092 // mChannelMask 1093 mChannelCount(0), 1094 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1095 mParamStatus(NO_ERROR), 1096 mStandby(false), mId(id), 1097 mDevice(device), 1098 mDeathRecipient(new PMDeathRecipient(this)) 1099{ 1100} 1101 1102AudioFlinger::ThreadBase::~ThreadBase() 1103{ 1104 mParamCond.broadcast(); 1105 // do not lock the mutex in destructor 1106 releaseWakeLock_l(); 1107 if (mPowerManager != 0) { 1108 sp<IBinder> binder = mPowerManager->asBinder(); 1109 binder->unlinkToDeath(mDeathRecipient); 1110 } 1111} 1112 1113void AudioFlinger::ThreadBase::exit() 1114{ 1115 ALOGV("ThreadBase::exit"); 1116 { 1117 // This lock prevents the following race in thread (uniprocessor for illustration): 1118 // if (!exitPending()) { 1119 // // context switch from here to exit() 1120 // // exit() calls requestExit(), what exitPending() observes 1121 // // exit() calls signal(), which is dropped since no waiters 1122 // // context switch back from exit() to here 1123 // mWaitWorkCV.wait(...); 1124 // // now thread is hung 1125 // } 1126 AutoMutex lock(mLock); 1127 requestExit(); 1128 mWaitWorkCV.signal(); 1129 } 1130 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1131 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1132 requestExitAndWait(); 1133} 1134 1135status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1136{ 1137 status_t status; 1138 1139 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1140 Mutex::Autolock _l(mLock); 1141 1142 mNewParameters.add(keyValuePairs); 1143 mWaitWorkCV.signal(); 1144 // wait condition with timeout in case the thread loop has exited 1145 // before the request could be processed 1146 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1147 status = mParamStatus; 1148 mWaitWorkCV.signal(); 1149 } else { 1150 status = TIMED_OUT; 1151 } 1152 return status; 1153} 1154 1155void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1156{ 1157 Mutex::Autolock _l(mLock); 1158 sendConfigEvent_l(event, param); 1159} 1160 1161// sendConfigEvent_l() must be called with ThreadBase::mLock held 1162void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1163{ 1164 ConfigEvent configEvent; 1165 configEvent.mEvent = event; 1166 configEvent.mParam = param; 1167 mConfigEvents.add(configEvent); 1168 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1169 mWaitWorkCV.signal(); 1170} 1171 1172void AudioFlinger::ThreadBase::processConfigEvents() 1173{ 1174 mLock.lock(); 1175 while (!mConfigEvents.isEmpty()) { 1176 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1177 ConfigEvent configEvent = mConfigEvents[0]; 1178 mConfigEvents.removeAt(0); 1179 // release mLock before locking AudioFlinger mLock: lock order is always 1180 // AudioFlinger then ThreadBase to avoid cross deadlock 1181 mLock.unlock(); 1182 mAudioFlinger->mLock.lock(); 1183 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1184 mAudioFlinger->mLock.unlock(); 1185 mLock.lock(); 1186 } 1187 mLock.unlock(); 1188} 1189 1190status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1191{ 1192 const size_t SIZE = 256; 1193 char buffer[SIZE]; 1194 String8 result; 1195 1196 bool locked = tryLock(mLock); 1197 if (!locked) { 1198 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1199 write(fd, buffer, strlen(buffer)); 1200 } 1201 1202 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1203 result.append(buffer); 1204 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1205 result.append(buffer); 1206 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1207 result.append(buffer); 1208 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1209 result.append(buffer); 1210 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1211 result.append(buffer); 1212 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1213 result.append(buffer); 1214 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1217 result.append(buffer); 1218 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1219 result.append(buffer); 1220 1221 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1222 result.append(buffer); 1223 result.append(" Index Command"); 1224 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1225 snprintf(buffer, SIZE, "\n %02d ", i); 1226 result.append(buffer); 1227 result.append(mNewParameters[i]); 1228 } 1229 1230 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1231 result.append(buffer); 1232 snprintf(buffer, SIZE, " Index event param\n"); 1233 result.append(buffer); 1234 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1235 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1236 result.append(buffer); 1237 } 1238 result.append("\n"); 1239 1240 write(fd, result.string(), result.size()); 1241 1242 if (locked) { 1243 mLock.unlock(); 1244 } 1245 return NO_ERROR; 1246} 1247 1248status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1249{ 1250 const size_t SIZE = 256; 1251 char buffer[SIZE]; 1252 String8 result; 1253 1254 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1255 write(fd, buffer, strlen(buffer)); 1256 1257 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1258 sp<EffectChain> chain = mEffectChains[i]; 1259 if (chain != 0) { 1260 chain->dump(fd, args); 1261 } 1262 } 1263 return NO_ERROR; 1264} 1265 1266void AudioFlinger::ThreadBase::acquireWakeLock() 1267{ 1268 Mutex::Autolock _l(mLock); 1269 acquireWakeLock_l(); 1270} 1271 1272void AudioFlinger::ThreadBase::acquireWakeLock_l() 1273{ 1274 if (mPowerManager == 0) { 1275 // use checkService() to avoid blocking if power service is not up yet 1276 sp<IBinder> binder = 1277 defaultServiceManager()->checkService(String16("power")); 1278 if (binder == 0) { 1279 ALOGW("Thread %s cannot connect to the power manager service", mName); 1280 } else { 1281 mPowerManager = interface_cast<IPowerManager>(binder); 1282 binder->linkToDeath(mDeathRecipient); 1283 } 1284 } 1285 if (mPowerManager != 0) { 1286 sp<IBinder> binder = new BBinder(); 1287 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1288 binder, 1289 String16(mName)); 1290 if (status == NO_ERROR) { 1291 mWakeLockToken = binder; 1292 } 1293 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1294 } 1295} 1296 1297void AudioFlinger::ThreadBase::releaseWakeLock() 1298{ 1299 Mutex::Autolock _l(mLock); 1300 releaseWakeLock_l(); 1301} 1302 1303void AudioFlinger::ThreadBase::releaseWakeLock_l() 1304{ 1305 if (mWakeLockToken != 0) { 1306 ALOGV("releaseWakeLock_l() %s", mName); 1307 if (mPowerManager != 0) { 1308 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1309 } 1310 mWakeLockToken.clear(); 1311 } 1312} 1313 1314void AudioFlinger::ThreadBase::clearPowerManager() 1315{ 1316 Mutex::Autolock _l(mLock); 1317 releaseWakeLock_l(); 1318 mPowerManager.clear(); 1319} 1320 1321void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1322{ 1323 sp<ThreadBase> thread = mThread.promote(); 1324 if (thread != 0) { 1325 thread->clearPowerManager(); 1326 } 1327 ALOGW("power manager service died !!!"); 1328} 1329 1330void AudioFlinger::ThreadBase::setEffectSuspended( 1331 const effect_uuid_t *type, bool suspend, int sessionId) 1332{ 1333 Mutex::Autolock _l(mLock); 1334 setEffectSuspended_l(type, suspend, sessionId); 1335} 1336 1337void AudioFlinger::ThreadBase::setEffectSuspended_l( 1338 const effect_uuid_t *type, bool suspend, int sessionId) 1339{ 1340 sp<EffectChain> chain = getEffectChain_l(sessionId); 1341 if (chain != 0) { 1342 if (type != NULL) { 1343 chain->setEffectSuspended_l(type, suspend); 1344 } else { 1345 chain->setEffectSuspendedAll_l(suspend); 1346 } 1347 } 1348 1349 updateSuspendedSessions_l(type, suspend, sessionId); 1350} 1351 1352void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1353{ 1354 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1355 if (index < 0) { 1356 return; 1357 } 1358 1359 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1360 mSuspendedSessions.editValueAt(index); 1361 1362 for (size_t i = 0; i < sessionEffects.size(); i++) { 1363 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1364 for (int j = 0; j < desc->mRefCount; j++) { 1365 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1366 chain->setEffectSuspendedAll_l(true); 1367 } else { 1368 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1369 desc->mType.timeLow); 1370 chain->setEffectSuspended_l(&desc->mType, true); 1371 } 1372 } 1373 } 1374} 1375 1376void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1377 bool suspend, 1378 int sessionId) 1379{ 1380 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1381 1382 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1383 1384 if (suspend) { 1385 if (index >= 0) { 1386 sessionEffects = mSuspendedSessions.editValueAt(index); 1387 } else { 1388 mSuspendedSessions.add(sessionId, sessionEffects); 1389 } 1390 } else { 1391 if (index < 0) { 1392 return; 1393 } 1394 sessionEffects = mSuspendedSessions.editValueAt(index); 1395 } 1396 1397 1398 int key = EffectChain::kKeyForSuspendAll; 1399 if (type != NULL) { 1400 key = type->timeLow; 1401 } 1402 index = sessionEffects.indexOfKey(key); 1403 1404 sp<SuspendedSessionDesc> desc; 1405 if (suspend) { 1406 if (index >= 0) { 1407 desc = sessionEffects.valueAt(index); 1408 } else { 1409 desc = new SuspendedSessionDesc(); 1410 if (type != NULL) { 1411 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1412 } 1413 sessionEffects.add(key, desc); 1414 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1415 } 1416 desc->mRefCount++; 1417 } else { 1418 if (index < 0) { 1419 return; 1420 } 1421 desc = sessionEffects.valueAt(index); 1422 if (--desc->mRefCount == 0) { 1423 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1424 sessionEffects.removeItemsAt(index); 1425 if (sessionEffects.isEmpty()) { 1426 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1427 sessionId); 1428 mSuspendedSessions.removeItem(sessionId); 1429 } 1430 } 1431 } 1432 if (!sessionEffects.isEmpty()) { 1433 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1434 } 1435} 1436 1437void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1438 bool enabled, 1439 int sessionId) 1440{ 1441 Mutex::Autolock _l(mLock); 1442 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1443} 1444 1445void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1446 bool enabled, 1447 int sessionId) 1448{ 1449 if (mType != RECORD) { 1450 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1451 // another session. This gives the priority to well behaved effect control panels 1452 // and applications not using global effects. 1453 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1454 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1455 } 1456 } 1457 1458 sp<EffectChain> chain = getEffectChain_l(sessionId); 1459 if (chain != 0) { 1460 chain->checkSuspendOnEffectEnabled(effect, enabled); 1461 } 1462} 1463 1464// ---------------------------------------------------------------------------- 1465 1466AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1467 AudioStreamOut* output, 1468 audio_io_handle_t id, 1469 uint32_t device, 1470 type_t type) 1471 : ThreadBase(audioFlinger, id, device, type), 1472 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1473 // Assumes constructor is called by AudioFlinger with it's mLock held, 1474 // but it would be safer to explicitly pass initial masterMute as parameter 1475 mMasterMute(audioFlinger->masterMute_l()), 1476 // mStreamTypes[] initialized in constructor body 1477 mOutput(output), 1478 // Assumes constructor is called by AudioFlinger with it's mLock held, 1479 // but it would be safer to explicitly pass initial masterVolume as parameter 1480 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1481 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1482 mMixerStatus(MIXER_IDLE), 1483 mPrevMixerStatus(MIXER_IDLE), 1484 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1485{ 1486 snprintf(mName, kNameLength, "AudioOut_%X", id); 1487 1488 readOutputParameters(); 1489 1490 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1491 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1492 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1493 stream = (audio_stream_type_t) (stream + 1)) { 1494 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1495 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1496 // initialized by stream_type_t default constructor 1497 // mStreamTypes[stream].valid = true; 1498 } 1499 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1500 // because mAudioFlinger doesn't have one to copy from 1501} 1502 1503AudioFlinger::PlaybackThread::~PlaybackThread() 1504{ 1505 delete [] mMixBuffer; 1506} 1507 1508status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1509{ 1510 dumpInternals(fd, args); 1511 dumpTracks(fd, args); 1512 dumpEffectChains(fd, args); 1513 return NO_ERROR; 1514} 1515 1516status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1517{ 1518 const size_t SIZE = 256; 1519 char buffer[SIZE]; 1520 String8 result; 1521 1522 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1523 result.append(buffer); 1524 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1525 for (size_t i = 0; i < mTracks.size(); ++i) { 1526 sp<Track> track = mTracks[i]; 1527 if (track != 0) { 1528 track->dump(buffer, SIZE); 1529 result.append(buffer); 1530 } 1531 } 1532 1533 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1534 result.append(buffer); 1535 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1536 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1537 sp<Track> track = mActiveTracks[i].promote(); 1538 if (track != 0) { 1539 track->dump(buffer, SIZE); 1540 result.append(buffer); 1541 } 1542 } 1543 write(fd, result.string(), result.size()); 1544 return NO_ERROR; 1545} 1546 1547status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1548{ 1549 const size_t SIZE = 256; 1550 char buffer[SIZE]; 1551 String8 result; 1552 1553 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1554 result.append(buffer); 1555 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1556 result.append(buffer); 1557 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1558 result.append(buffer); 1559 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1560 result.append(buffer); 1561 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1562 result.append(buffer); 1563 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1564 result.append(buffer); 1565 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1566 result.append(buffer); 1567 write(fd, result.string(), result.size()); 1568 1569 dumpBase(fd, args); 1570 1571 return NO_ERROR; 1572} 1573 1574// Thread virtuals 1575status_t AudioFlinger::PlaybackThread::readyToRun() 1576{ 1577 status_t status = initCheck(); 1578 if (status == NO_ERROR) { 1579 ALOGI("AudioFlinger's thread %p ready to run", this); 1580 } else { 1581 ALOGE("No working audio driver found."); 1582 } 1583 return status; 1584} 1585 1586void AudioFlinger::PlaybackThread::onFirstRef() 1587{ 1588 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1589} 1590 1591// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1592sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1593 const sp<AudioFlinger::Client>& client, 1594 audio_stream_type_t streamType, 1595 uint32_t sampleRate, 1596 audio_format_t format, 1597 uint32_t channelMask, 1598 int frameCount, 1599 const sp<IMemory>& sharedBuffer, 1600 int sessionId, 1601 bool isTimed, 1602 status_t *status) 1603{ 1604 sp<Track> track; 1605 status_t lStatus; 1606 1607 if (mType == DIRECT) { 1608 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1609 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1610 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1611 "for output %p with format %d", 1612 sampleRate, format, channelMask, mOutput, mFormat); 1613 lStatus = BAD_VALUE; 1614 goto Exit; 1615 } 1616 } 1617 } else { 1618 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1619 if (sampleRate > mSampleRate*2) { 1620 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1621 lStatus = BAD_VALUE; 1622 goto Exit; 1623 } 1624 } 1625 1626 lStatus = initCheck(); 1627 if (lStatus != NO_ERROR) { 1628 ALOGE("Audio driver not initialized."); 1629 goto Exit; 1630 } 1631 1632 { // scope for mLock 1633 Mutex::Autolock _l(mLock); 1634 1635 // all tracks in same audio session must share the same routing strategy otherwise 1636 // conflicts will happen when tracks are moved from one output to another by audio policy 1637 // manager 1638 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1639 for (size_t i = 0; i < mTracks.size(); ++i) { 1640 sp<Track> t = mTracks[i]; 1641 if (t != 0 && !t->isOutputTrack()) { 1642 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1643 if (sessionId == t->sessionId() && strategy != actual) { 1644 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1645 strategy, actual); 1646 lStatus = BAD_VALUE; 1647 goto Exit; 1648 } 1649 } 1650 } 1651 1652 if (!isTimed) { 1653 track = new Track(this, client, streamType, sampleRate, format, 1654 channelMask, frameCount, sharedBuffer, sessionId); 1655 } else { 1656 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1657 channelMask, frameCount, sharedBuffer, sessionId); 1658 } 1659 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1660 lStatus = NO_MEMORY; 1661 goto Exit; 1662 } 1663 mTracks.add(track); 1664 1665 sp<EffectChain> chain = getEffectChain_l(sessionId); 1666 if (chain != 0) { 1667 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1668 track->setMainBuffer(chain->inBuffer()); 1669 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1670 chain->incTrackCnt(); 1671 } 1672 1673 // invalidate track immediately if the stream type was moved to another thread since 1674 // createTrack() was called by the client process. 1675 if (!mStreamTypes[streamType].valid) { 1676 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1677 this, streamType); 1678 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1679 } 1680 } 1681 lStatus = NO_ERROR; 1682 1683Exit: 1684 if (status) { 1685 *status = lStatus; 1686 } 1687 return track; 1688} 1689 1690uint32_t AudioFlinger::PlaybackThread::latency() const 1691{ 1692 Mutex::Autolock _l(mLock); 1693 if (initCheck() == NO_ERROR) { 1694 return mOutput->stream->get_latency(mOutput->stream); 1695 } else { 1696 return 0; 1697 } 1698} 1699 1700void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1701{ 1702 Mutex::Autolock _l(mLock); 1703 mMasterVolume = value; 1704} 1705 1706void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1707{ 1708 Mutex::Autolock _l(mLock); 1709 setMasterMute_l(muted); 1710} 1711 1712void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1713{ 1714 Mutex::Autolock _l(mLock); 1715 mStreamTypes[stream].volume = value; 1716} 1717 1718void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1719{ 1720 Mutex::Autolock _l(mLock); 1721 mStreamTypes[stream].mute = muted; 1722} 1723 1724float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1725{ 1726 Mutex::Autolock _l(mLock); 1727 return mStreamTypes[stream].volume; 1728} 1729 1730// addTrack_l() must be called with ThreadBase::mLock held 1731status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1732{ 1733 status_t status = ALREADY_EXISTS; 1734 1735 // set retry count for buffer fill 1736 track->mRetryCount = kMaxTrackStartupRetries; 1737 if (mActiveTracks.indexOf(track) < 0) { 1738 // the track is newly added, make sure it fills up all its 1739 // buffers before playing. This is to ensure the client will 1740 // effectively get the latency it requested. 1741 track->mFillingUpStatus = Track::FS_FILLING; 1742 track->mResetDone = false; 1743 mActiveTracks.add(track); 1744 if (track->mainBuffer() != mMixBuffer) { 1745 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1746 if (chain != 0) { 1747 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1748 chain->incActiveTrackCnt(); 1749 } 1750 } 1751 1752 status = NO_ERROR; 1753 } 1754 1755 ALOGV("mWaitWorkCV.broadcast"); 1756 mWaitWorkCV.broadcast(); 1757 1758 return status; 1759} 1760 1761// destroyTrack_l() must be called with ThreadBase::mLock held 1762void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1763{ 1764 track->mState = TrackBase::TERMINATED; 1765 if (mActiveTracks.indexOf(track) < 0) { 1766 removeTrack_l(track); 1767 } 1768} 1769 1770void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1771{ 1772 mTracks.remove(track); 1773 deleteTrackName_l(track->name()); 1774 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1775 if (chain != 0) { 1776 chain->decTrackCnt(); 1777 } 1778} 1779 1780String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1781{ 1782 String8 out_s8 = String8(""); 1783 char *s; 1784 1785 Mutex::Autolock _l(mLock); 1786 if (initCheck() != NO_ERROR) { 1787 return out_s8; 1788 } 1789 1790 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1791 out_s8 = String8(s); 1792 free(s); 1793 return out_s8; 1794} 1795 1796// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1797void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1798 AudioSystem::OutputDescriptor desc; 1799 void *param2 = NULL; 1800 1801 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1802 1803 switch (event) { 1804 case AudioSystem::OUTPUT_OPENED: 1805 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1806 desc.channels = mChannelMask; 1807 desc.samplingRate = mSampleRate; 1808 desc.format = mFormat; 1809 desc.frameCount = mFrameCount; 1810 desc.latency = latency(); 1811 param2 = &desc; 1812 break; 1813 1814 case AudioSystem::STREAM_CONFIG_CHANGED: 1815 param2 = ¶m; 1816 case AudioSystem::OUTPUT_CLOSED: 1817 default: 1818 break; 1819 } 1820 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1821} 1822 1823void AudioFlinger::PlaybackThread::readOutputParameters() 1824{ 1825 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1826 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1827 mChannelCount = (uint16_t)popcount(mChannelMask); 1828 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1829 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1830 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1831 1832 // FIXME - Current mixer implementation only supports stereo output: Always 1833 // Allocate a stereo buffer even if HW output is mono. 1834 delete[] mMixBuffer; 1835 mMixBuffer = new int16_t[mFrameCount * 2]; 1836 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1837 1838 // force reconfiguration of effect chains and engines to take new buffer size and audio 1839 // parameters into account 1840 // Note that mLock is not held when readOutputParameters() is called from the constructor 1841 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1842 // matter. 1843 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1844 Vector< sp<EffectChain> > effectChains = mEffectChains; 1845 for (size_t i = 0; i < effectChains.size(); i ++) { 1846 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1847 } 1848} 1849 1850status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1851{ 1852 if (halFrames == NULL || dspFrames == NULL) { 1853 return BAD_VALUE; 1854 } 1855 Mutex::Autolock _l(mLock); 1856 if (initCheck() != NO_ERROR) { 1857 return INVALID_OPERATION; 1858 } 1859 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1860 1861 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1862} 1863 1864uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1865{ 1866 Mutex::Autolock _l(mLock); 1867 uint32_t result = 0; 1868 if (getEffectChain_l(sessionId) != 0) { 1869 result = EFFECT_SESSION; 1870 } 1871 1872 for (size_t i = 0; i < mTracks.size(); ++i) { 1873 sp<Track> track = mTracks[i]; 1874 if (sessionId == track->sessionId() && 1875 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1876 result |= TRACK_SESSION; 1877 break; 1878 } 1879 } 1880 1881 return result; 1882} 1883 1884uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1885{ 1886 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1887 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1888 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1889 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1890 } 1891 for (size_t i = 0; i < mTracks.size(); i++) { 1892 sp<Track> track = mTracks[i]; 1893 if (sessionId == track->sessionId() && 1894 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1895 return AudioSystem::getStrategyForStream(track->streamType()); 1896 } 1897 } 1898 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1899} 1900 1901 1902AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1903{ 1904 Mutex::Autolock _l(mLock); 1905 return mOutput; 1906} 1907 1908AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1909{ 1910 Mutex::Autolock _l(mLock); 1911 AudioStreamOut *output = mOutput; 1912 mOutput = NULL; 1913 return output; 1914} 1915 1916// this method must always be called either with ThreadBase mLock held or inside the thread loop 1917audio_stream_t* AudioFlinger::PlaybackThread::stream() 1918{ 1919 if (mOutput == NULL) { 1920 return NULL; 1921 } 1922 return &mOutput->stream->common; 1923} 1924 1925uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1926{ 1927 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1928 // decoding and transfer time. So sleeping for half of the latency would likely cause 1929 // underruns 1930 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1931 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1932 } else { 1933 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1934 } 1935} 1936 1937// ---------------------------------------------------------------------------- 1938 1939AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1940 audio_io_handle_t id, uint32_t device, type_t type) 1941 : PlaybackThread(audioFlinger, output, id, device, type) 1942{ 1943 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1944 // FIXME - Current mixer implementation only supports stereo output 1945 if (mChannelCount == 1) { 1946 ALOGE("Invalid audio hardware channel count"); 1947 } 1948} 1949 1950AudioFlinger::MixerThread::~MixerThread() 1951{ 1952 delete mAudioMixer; 1953} 1954 1955class CpuStats { 1956public: 1957 CpuStats(); 1958 void sample(const String8 &title); 1959#ifdef DEBUG_CPU_USAGE 1960private: 1961 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 1962 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 1963 1964 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 1965 1966 int mCpuNum; // thread's current CPU number 1967 int mCpukHz; // frequency of thread's current CPU in kHz 1968#endif 1969}; 1970 1971CpuStats::CpuStats() 1972#ifdef DEBUG_CPU_USAGE 1973 : mCpuNum(-1), mCpukHz(-1) 1974#endif 1975{ 1976} 1977 1978void CpuStats::sample(const String8 &title) { 1979#ifdef DEBUG_CPU_USAGE 1980 // get current thread's delta CPU time in wall clock ns 1981 double wcNs; 1982 bool valid = mCpuUsage.sampleAndEnable(wcNs); 1983 1984 // record sample for wall clock statistics 1985 if (valid) { 1986 mWcStats.sample(wcNs); 1987 } 1988 1989 // get the current CPU number 1990 int cpuNum = sched_getcpu(); 1991 1992 // get the current CPU frequency in kHz 1993 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 1994 1995 // check if either CPU number or frequency changed 1996 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 1997 mCpuNum = cpuNum; 1998 mCpukHz = cpukHz; 1999 // ignore sample for purposes of cycles 2000 valid = false; 2001 } 2002 2003 // if no change in CPU number or frequency, then record sample for cycle statistics 2004 if (valid && mCpukHz > 0) { 2005 double cycles = wcNs * cpukHz * 0.000001; 2006 mHzStats.sample(cycles); 2007 } 2008 2009 unsigned n = mWcStats.n(); 2010 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2011 if ((n & 127) == 1) { 2012 long long elapsed = mCpuUsage.elapsed(); 2013 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2014 double perLoop = elapsed / (double) n; 2015 double perLoop100 = perLoop * 0.01; 2016 double perLoop1k = perLoop * 0.001; 2017 double mean = mWcStats.mean(); 2018 double stddev = mWcStats.stddev(); 2019 double minimum = mWcStats.minimum(); 2020 double maximum = mWcStats.maximum(); 2021 double meanCycles = mHzStats.mean(); 2022 double stddevCycles = mHzStats.stddev(); 2023 double minCycles = mHzStats.minimum(); 2024 double maxCycles = mHzStats.maximum(); 2025 mCpuUsage.resetElapsed(); 2026 mWcStats.reset(); 2027 mHzStats.reset(); 2028 ALOGD("CPU usage for %s over past %.1f secs\n" 2029 " (%u mixer loops at %.1f mean ms per loop):\n" 2030 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2031 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2032 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2033 title.string(), 2034 elapsed * .000000001, n, perLoop * .000001, 2035 mean * .001, 2036 stddev * .001, 2037 minimum * .001, 2038 maximum * .001, 2039 mean / perLoop100, 2040 stddev / perLoop100, 2041 minimum / perLoop100, 2042 maximum / perLoop100, 2043 meanCycles / perLoop1k, 2044 stddevCycles / perLoop1k, 2045 minCycles / perLoop1k, 2046 maxCycles / perLoop1k); 2047 2048 } 2049 } 2050#endif 2051}; 2052 2053void AudioFlinger::PlaybackThread::checkSilentMode_l() 2054{ 2055 if (!mMasterMute) { 2056 char value[PROPERTY_VALUE_MAX]; 2057 if (property_get("ro.audio.silent", value, "0") > 0) { 2058 char *endptr; 2059 unsigned long ul = strtoul(value, &endptr, 0); 2060 if (*endptr == '\0' && ul != 0) { 2061 ALOGD("Silence is golden"); 2062 // The setprop command will not allow a property to be changed after 2063 // the first time it is set, so we don't have to worry about un-muting. 2064 setMasterMute_l(true); 2065 } 2066 } 2067 } 2068} 2069 2070bool AudioFlinger::PlaybackThread::threadLoop() 2071{ 2072 Vector< sp<Track> > tracksToRemove; 2073 2074 standbyTime = systemTime(); 2075 2076 // MIXER 2077 nsecs_t lastWarning = 0; 2078if (mType == MIXER) { 2079 longStandbyExit = false; 2080} 2081 2082 // DUPLICATING 2083 // FIXME could this be made local to while loop? 2084 writeFrames = 0; 2085 2086 cacheParameters_l(); 2087 sleepTime = idleSleepTime; 2088 2089if (mType == MIXER) { 2090 sleepTimeShift = 0; 2091} 2092 2093 CpuStats cpuStats; 2094 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2095 2096 acquireWakeLock(); 2097 2098 while (!exitPending()) 2099 { 2100 cpuStats.sample(myName); 2101 2102 Vector< sp<EffectChain> > effectChains; 2103 2104 processConfigEvents(); 2105 2106 { // scope for mLock 2107 2108 Mutex::Autolock _l(mLock); 2109 2110 if (checkForNewParameters_l()) { 2111 cacheParameters_l(); 2112 } 2113 2114 saveOutputTracks(); 2115 2116 // put audio hardware into standby after short delay 2117 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2118 mSuspended > 0)) { 2119 if (!mStandby) { 2120 2121 threadLoop_standby(); 2122 2123 mStandby = true; 2124 mBytesWritten = 0; 2125 } 2126 2127 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2128 // we're about to wait, flush the binder command buffer 2129 IPCThreadState::self()->flushCommands(); 2130 2131 clearOutputTracks(); 2132 2133 if (exitPending()) break; 2134 2135 releaseWakeLock_l(); 2136 // wait until we have something to do... 2137 ALOGV("%s going to sleep", myName.string()); 2138 mWaitWorkCV.wait(mLock); 2139 ALOGV("%s waking up", myName.string()); 2140 acquireWakeLock_l(); 2141 2142 mPrevMixerStatus = MIXER_IDLE; 2143 2144 checkSilentMode_l(); 2145 2146 standbyTime = systemTime() + standbyDelay; 2147 sleepTime = idleSleepTime; 2148 if (mType == MIXER) { 2149 sleepTimeShift = 0; 2150 } 2151 2152 continue; 2153 } 2154 } 2155 2156 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2157 // Shift in the new status; this could be a queue if it's 2158 // useful to filter the mixer status over several cycles. 2159 mPrevMixerStatus = mMixerStatus; 2160 mMixerStatus = newMixerStatus; 2161 2162 // prevent any changes in effect chain list and in each effect chain 2163 // during mixing and effect process as the audio buffers could be deleted 2164 // or modified if an effect is created or deleted 2165 lockEffectChains_l(effectChains); 2166 } 2167 2168 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2169 threadLoop_mix(); 2170 } else { 2171 threadLoop_sleepTime(); 2172 } 2173 2174 if (mSuspended > 0) { 2175 sleepTime = suspendSleepTimeUs(); 2176 } 2177 2178 // only process effects if we're going to write 2179 if (sleepTime == 0) { 2180 for (size_t i = 0; i < effectChains.size(); i ++) { 2181 effectChains[i]->process_l(); 2182 } 2183 } 2184 2185 // enable changes in effect chain 2186 unlockEffectChains(effectChains); 2187 2188 // sleepTime == 0 means we must write to audio hardware 2189 if (sleepTime == 0) { 2190 2191 threadLoop_write(); 2192 2193if (mType == MIXER) { 2194 // write blocked detection 2195 nsecs_t now = systemTime(); 2196 nsecs_t delta = now - mLastWriteTime; 2197 if (!mStandby && delta > maxPeriod) { 2198 mNumDelayedWrites++; 2199 if ((now - lastWarning) > kWarningThrottleNs) { 2200 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2201 ns2ms(delta), mNumDelayedWrites, this); 2202 lastWarning = now; 2203 } 2204 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2205 // a different threshold. Or completely removed for what it is worth anyway... 2206 if (mStandby) { 2207 longStandbyExit = true; 2208 } 2209 } 2210} 2211 2212 mStandby = false; 2213 } else { 2214 usleep(sleepTime); 2215 } 2216 2217 // finally let go of removed track(s), without the lock held 2218 // since we can't guarantee the destructors won't acquire that 2219 // same lock. 2220 tracksToRemove.clear(); 2221 2222 // FIXME I don't understand the need for this here; 2223 // it was in the original code but maybe the 2224 // assignment in saveOutputTracks() makes this unnecessary? 2225 clearOutputTracks(); 2226 2227 // Effect chains will be actually deleted here if they were removed from 2228 // mEffectChains list during mixing or effects processing 2229 effectChains.clear(); 2230 2231 // FIXME Note that the above .clear() is no longer necessary since effectChains 2232 // is now local to this block, but will keep it for now (at least until merge done). 2233 } 2234 2235if (mType == MIXER || mType == DIRECT) { 2236 // put output stream into standby mode 2237 if (!mStandby) { 2238 mOutput->stream->common.standby(&mOutput->stream->common); 2239 } 2240} 2241if (mType == DUPLICATING) { 2242 // for DuplicatingThread, standby mode is handled by the outputTracks 2243} 2244 2245 releaseWakeLock(); 2246 2247 ALOGV("Thread %p type %d exiting", this, mType); 2248 return false; 2249} 2250 2251// shared by MIXER and DIRECT, overridden by DUPLICATING 2252void AudioFlinger::PlaybackThread::threadLoop_write() 2253{ 2254 // FIXME rewrite to reduce number of system calls 2255 mLastWriteTime = systemTime(); 2256 mInWrite = true; 2257 mBytesWritten += mixBufferSize; 2258 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2259 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2260 mNumWrites++; 2261 mInWrite = false; 2262} 2263 2264// shared by MIXER and DIRECT, overridden by DUPLICATING 2265void AudioFlinger::PlaybackThread::threadLoop_standby() 2266{ 2267 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2268 mOutput->stream->common.standby(&mOutput->stream->common); 2269} 2270 2271void AudioFlinger::MixerThread::threadLoop_mix() 2272{ 2273 // obtain the presentation timestamp of the next output buffer 2274 int64_t pts; 2275 status_t status = INVALID_OPERATION; 2276 2277 if (NULL != mOutput->stream->get_next_write_timestamp) { 2278 status = mOutput->stream->get_next_write_timestamp( 2279 mOutput->stream, &pts); 2280 } 2281 2282 if (status != NO_ERROR) { 2283 pts = AudioBufferProvider::kInvalidPTS; 2284 } 2285 2286 // mix buffers... 2287 mAudioMixer->process(pts); 2288 // increase sleep time progressively when application underrun condition clears. 2289 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2290 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2291 // such that we would underrun the audio HAL. 2292 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2293 sleepTimeShift--; 2294 } 2295 sleepTime = 0; 2296 standbyTime = systemTime() + standbyDelay; 2297 //TODO: delay standby when effects have a tail 2298} 2299 2300void AudioFlinger::MixerThread::threadLoop_sleepTime() 2301{ 2302 // If no tracks are ready, sleep once for the duration of an output 2303 // buffer size, then write 0s to the output 2304 if (sleepTime == 0) { 2305 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2306 sleepTime = activeSleepTime >> sleepTimeShift; 2307 if (sleepTime < kMinThreadSleepTimeUs) { 2308 sleepTime = kMinThreadSleepTimeUs; 2309 } 2310 // reduce sleep time in case of consecutive application underruns to avoid 2311 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2312 // duration we would end up writing less data than needed by the audio HAL if 2313 // the condition persists. 2314 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2315 sleepTimeShift++; 2316 } 2317 } else { 2318 sleepTime = idleSleepTime; 2319 } 2320 } else if (mBytesWritten != 0 || 2321 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2322 memset (mMixBuffer, 0, mixBufferSize); 2323 sleepTime = 0; 2324 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2325 } 2326 // TODO add standby time extension fct of effect tail 2327} 2328 2329// prepareTracks_l() must be called with ThreadBase::mLock held 2330AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2331 Vector< sp<Track> > *tracksToRemove) 2332{ 2333 2334 mixer_state mixerStatus = MIXER_IDLE; 2335 // find out which tracks need to be processed 2336 size_t count = mActiveTracks.size(); 2337 size_t mixedTracks = 0; 2338 size_t tracksWithEffect = 0; 2339 2340 float masterVolume = mMasterVolume; 2341 bool masterMute = mMasterMute; 2342 2343 if (masterMute) { 2344 masterVolume = 0; 2345 } 2346 // Delegate master volume control to effect in output mix effect chain if needed 2347 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2348 if (chain != 0) { 2349 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2350 chain->setVolume_l(&v, &v); 2351 masterVolume = (float)((v + (1 << 23)) >> 24); 2352 chain.clear(); 2353 } 2354 2355 for (size_t i=0 ; i<count ; i++) { 2356 sp<Track> t = mActiveTracks[i].promote(); 2357 if (t == 0) continue; 2358 2359 // this const just means the local variable doesn't change 2360 Track* const track = t.get(); 2361 audio_track_cblk_t* cblk = track->cblk(); 2362 2363 // The first time a track is added we wait 2364 // for all its buffers to be filled before processing it 2365 int name = track->name(); 2366 // make sure that we have enough frames to mix one full buffer. 2367 // enforce this condition only once to enable draining the buffer in case the client 2368 // app does not call stop() and relies on underrun to stop: 2369 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2370 // during last round 2371 uint32_t minFrames = 1; 2372 if (!track->isStopped() && !track->isPausing() && 2373 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2374 if (t->sampleRate() == (int)mSampleRate) { 2375 minFrames = mFrameCount; 2376 } else { 2377 // +1 for rounding and +1 for additional sample needed for interpolation 2378 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2379 // add frames already consumed but not yet released by the resampler 2380 // because cblk->framesReady() will include these frames 2381 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2382 // the minimum track buffer size is normally twice the number of frames necessary 2383 // to fill one buffer and the resampler should not leave more than one buffer worth 2384 // of unreleased frames after each pass, but just in case... 2385 ALOG_ASSERT(minFrames <= cblk->frameCount); 2386 } 2387 } 2388 if ((track->framesReady() >= minFrames) && track->isReady() && 2389 !track->isPaused() && !track->isTerminated()) 2390 { 2391 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2392 2393 mixedTracks++; 2394 2395 // track->mainBuffer() != mMixBuffer means there is an effect chain 2396 // connected to the track 2397 chain.clear(); 2398 if (track->mainBuffer() != mMixBuffer) { 2399 chain = getEffectChain_l(track->sessionId()); 2400 // Delegate volume control to effect in track effect chain if needed 2401 if (chain != 0) { 2402 tracksWithEffect++; 2403 } else { 2404 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2405 name, track->sessionId()); 2406 } 2407 } 2408 2409 2410 int param = AudioMixer::VOLUME; 2411 if (track->mFillingUpStatus == Track::FS_FILLED) { 2412 // no ramp for the first volume setting 2413 track->mFillingUpStatus = Track::FS_ACTIVE; 2414 if (track->mState == TrackBase::RESUMING) { 2415 track->mState = TrackBase::ACTIVE; 2416 param = AudioMixer::RAMP_VOLUME; 2417 } 2418 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2419 } else if (cblk->server != 0) { 2420 // If the track is stopped before the first frame was mixed, 2421 // do not apply ramp 2422 param = AudioMixer::RAMP_VOLUME; 2423 } 2424 2425 // compute volume for this track 2426 uint32_t vl, vr, va; 2427 if (track->isMuted() || track->isPausing() || 2428 mStreamTypes[track->streamType()].mute) { 2429 vl = vr = va = 0; 2430 if (track->isPausing()) { 2431 track->setPaused(); 2432 } 2433 } else { 2434 2435 // read original volumes with volume control 2436 float typeVolume = mStreamTypes[track->streamType()].volume; 2437 float v = masterVolume * typeVolume; 2438 uint32_t vlr = cblk->getVolumeLR(); 2439 vl = vlr & 0xFFFF; 2440 vr = vlr >> 16; 2441 // track volumes come from shared memory, so can't be trusted and must be clamped 2442 if (vl > MAX_GAIN_INT) { 2443 ALOGV("Track left volume out of range: %04X", vl); 2444 vl = MAX_GAIN_INT; 2445 } 2446 if (vr > MAX_GAIN_INT) { 2447 ALOGV("Track right volume out of range: %04X", vr); 2448 vr = MAX_GAIN_INT; 2449 } 2450 // now apply the master volume and stream type volume 2451 vl = (uint32_t)(v * vl) << 12; 2452 vr = (uint32_t)(v * vr) << 12; 2453 // assuming master volume and stream type volume each go up to 1.0, 2454 // vl and vr are now in 8.24 format 2455 2456 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2457 // send level comes from shared memory and so may be corrupt 2458 if (sendLevel > MAX_GAIN_INT) { 2459 ALOGV("Track send level out of range: %04X", sendLevel); 2460 sendLevel = MAX_GAIN_INT; 2461 } 2462 va = (uint32_t)(v * sendLevel); 2463 } 2464 // Delegate volume control to effect in track effect chain if needed 2465 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2466 // Do not ramp volume if volume is controlled by effect 2467 param = AudioMixer::VOLUME; 2468 track->mHasVolumeController = true; 2469 } else { 2470 // force no volume ramp when volume controller was just disabled or removed 2471 // from effect chain to avoid volume spike 2472 if (track->mHasVolumeController) { 2473 param = AudioMixer::VOLUME; 2474 } 2475 track->mHasVolumeController = false; 2476 } 2477 2478 // Convert volumes from 8.24 to 4.12 format 2479 // This additional clamping is needed in case chain->setVolume_l() overshot 2480 vl = (vl + (1 << 11)) >> 12; 2481 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2482 vr = (vr + (1 << 11)) >> 12; 2483 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2484 2485 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2486 2487 // XXX: these things DON'T need to be done each time 2488 mAudioMixer->setBufferProvider(name, track); 2489 mAudioMixer->enable(name); 2490 2491 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2492 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2493 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2494 mAudioMixer->setParameter( 2495 name, 2496 AudioMixer::TRACK, 2497 AudioMixer::FORMAT, (void *)track->format()); 2498 mAudioMixer->setParameter( 2499 name, 2500 AudioMixer::TRACK, 2501 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2502 mAudioMixer->setParameter( 2503 name, 2504 AudioMixer::RESAMPLE, 2505 AudioMixer::SAMPLE_RATE, 2506 (void *)(cblk->sampleRate)); 2507 mAudioMixer->setParameter( 2508 name, 2509 AudioMixer::TRACK, 2510 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2511 mAudioMixer->setParameter( 2512 name, 2513 AudioMixer::TRACK, 2514 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2515 2516 // reset retry count 2517 track->mRetryCount = kMaxTrackRetries; 2518 2519 // If one track is ready, set the mixer ready if: 2520 // - the mixer was not ready during previous round OR 2521 // - no other track is not ready 2522 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2523 mixerStatus != MIXER_TRACKS_ENABLED) { 2524 mixerStatus = MIXER_TRACKS_READY; 2525 } 2526 } else { 2527 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2528 if (track->isStopped()) { 2529 track->reset(); 2530 } 2531 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2532 // We have consumed all the buffers of this track. 2533 // Remove it from the list of active tracks. 2534 tracksToRemove->add(track); 2535 } else { 2536 // No buffers for this track. Give it a few chances to 2537 // fill a buffer, then remove it from active list. 2538 if (--(track->mRetryCount) <= 0) { 2539 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2540 tracksToRemove->add(track); 2541 // indicate to client process that the track was disabled because of underrun 2542 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2543 // If one track is not ready, mark the mixer also not ready if: 2544 // - the mixer was ready during previous round OR 2545 // - no other track is ready 2546 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2547 mixerStatus != MIXER_TRACKS_READY) { 2548 mixerStatus = MIXER_TRACKS_ENABLED; 2549 } 2550 } 2551 mAudioMixer->disable(name); 2552 } 2553 } 2554 2555 // remove all the tracks that need to be... 2556 count = tracksToRemove->size(); 2557 if (CC_UNLIKELY(count)) { 2558 for (size_t i=0 ; i<count ; i++) { 2559 const sp<Track>& track = tracksToRemove->itemAt(i); 2560 mActiveTracks.remove(track); 2561 if (track->mainBuffer() != mMixBuffer) { 2562 chain = getEffectChain_l(track->sessionId()); 2563 if (chain != 0) { 2564 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2565 chain->decActiveTrackCnt(); 2566 } 2567 } 2568 if (track->isTerminated()) { 2569 removeTrack_l(track); 2570 } 2571 } 2572 } 2573 2574 // mix buffer must be cleared if all tracks are connected to an 2575 // effect chain as in this case the mixer will not write to 2576 // mix buffer and track effects will accumulate into it 2577 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2578 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2579 } 2580 2581 return mixerStatus; 2582} 2583 2584/* 2585The derived values that are cached: 2586 - mixBufferSize from frame count * frame size 2587 - activeSleepTime from activeSleepTimeUs() 2588 - idleSleepTime from idleSleepTimeUs() 2589 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2590 - maxPeriod from frame count and sample rate (MIXER only) 2591 2592The parameters that affect these derived values are: 2593 - frame count 2594 - frame size 2595 - sample rate 2596 - device type: A2DP or not 2597 - device latency 2598 - format: PCM or not 2599 - active sleep time 2600 - idle sleep time 2601*/ 2602 2603void AudioFlinger::PlaybackThread::cacheParameters_l() 2604{ 2605 mixBufferSize = mFrameCount * mFrameSize; 2606 activeSleepTime = activeSleepTimeUs(); 2607 idleSleepTime = idleSleepTimeUs(); 2608} 2609 2610void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2611{ 2612 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2613 this, streamType, mTracks.size()); 2614 Mutex::Autolock _l(mLock); 2615 2616 size_t size = mTracks.size(); 2617 for (size_t i = 0; i < size; i++) { 2618 sp<Track> t = mTracks[i]; 2619 if (t->streamType() == streamType) { 2620 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2621 t->mCblk->cv.signal(); 2622 } 2623 } 2624} 2625 2626void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2627{ 2628 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2629 this, streamType, valid); 2630 Mutex::Autolock _l(mLock); 2631 2632 mStreamTypes[streamType].valid = valid; 2633} 2634 2635// getTrackName_l() must be called with ThreadBase::mLock held 2636int AudioFlinger::MixerThread::getTrackName_l() 2637{ 2638 return mAudioMixer->getTrackName(); 2639} 2640 2641// deleteTrackName_l() must be called with ThreadBase::mLock held 2642void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2643{ 2644 ALOGV("remove track (%d) and delete from mixer", name); 2645 mAudioMixer->deleteTrackName(name); 2646} 2647 2648// checkForNewParameters_l() must be called with ThreadBase::mLock held 2649bool AudioFlinger::MixerThread::checkForNewParameters_l() 2650{ 2651 bool reconfig = false; 2652 2653 while (!mNewParameters.isEmpty()) { 2654 status_t status = NO_ERROR; 2655 String8 keyValuePair = mNewParameters[0]; 2656 AudioParameter param = AudioParameter(keyValuePair); 2657 int value; 2658 2659 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2660 reconfig = true; 2661 } 2662 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2663 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2664 status = BAD_VALUE; 2665 } else { 2666 reconfig = true; 2667 } 2668 } 2669 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2670 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2671 status = BAD_VALUE; 2672 } else { 2673 reconfig = true; 2674 } 2675 } 2676 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2677 // do not accept frame count changes if tracks are open as the track buffer 2678 // size depends on frame count and correct behavior would not be guaranteed 2679 // if frame count is changed after track creation 2680 if (!mTracks.isEmpty()) { 2681 status = INVALID_OPERATION; 2682 } else { 2683 reconfig = true; 2684 } 2685 } 2686 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2687#ifdef ADD_BATTERY_DATA 2688 // when changing the audio output device, call addBatteryData to notify 2689 // the change 2690 if ((int)mDevice != value) { 2691 uint32_t params = 0; 2692 // check whether speaker is on 2693 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2694 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2695 } 2696 2697 int deviceWithoutSpeaker 2698 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2699 // check if any other device (except speaker) is on 2700 if (value & deviceWithoutSpeaker ) { 2701 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2702 } 2703 2704 if (params != 0) { 2705 addBatteryData(params); 2706 } 2707 } 2708#endif 2709 2710 // forward device change to effects that have requested to be 2711 // aware of attached audio device. 2712 mDevice = (uint32_t)value; 2713 for (size_t i = 0; i < mEffectChains.size(); i++) { 2714 mEffectChains[i]->setDevice_l(mDevice); 2715 } 2716 } 2717 2718 if (status == NO_ERROR) { 2719 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2720 keyValuePair.string()); 2721 if (!mStandby && status == INVALID_OPERATION) { 2722 mOutput->stream->common.standby(&mOutput->stream->common); 2723 mStandby = true; 2724 mBytesWritten = 0; 2725 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2726 keyValuePair.string()); 2727 } 2728 if (status == NO_ERROR && reconfig) { 2729 delete mAudioMixer; 2730 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2731 mAudioMixer = NULL; 2732 readOutputParameters(); 2733 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2734 for (size_t i = 0; i < mTracks.size() ; i++) { 2735 int name = getTrackName_l(); 2736 if (name < 0) break; 2737 mTracks[i]->mName = name; 2738 // limit track sample rate to 2 x new output sample rate 2739 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2740 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2741 } 2742 } 2743 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2744 } 2745 } 2746 2747 mNewParameters.removeAt(0); 2748 2749 mParamStatus = status; 2750 mParamCond.signal(); 2751 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2752 // already timed out waiting for the status and will never signal the condition. 2753 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2754 } 2755 return reconfig; 2756} 2757 2758status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2759{ 2760 const size_t SIZE = 256; 2761 char buffer[SIZE]; 2762 String8 result; 2763 2764 PlaybackThread::dumpInternals(fd, args); 2765 2766 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2767 result.append(buffer); 2768 write(fd, result.string(), result.size()); 2769 return NO_ERROR; 2770} 2771 2772uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2773{ 2774 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2775} 2776 2777uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2778{ 2779 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2780} 2781 2782void AudioFlinger::MixerThread::cacheParameters_l() 2783{ 2784 PlaybackThread::cacheParameters_l(); 2785 2786 // FIXME: Relaxed timing because of a certain device that can't meet latency 2787 // Should be reduced to 2x after the vendor fixes the driver issue 2788 // increase threshold again due to low power audio mode. The way this warning 2789 // threshold is calculated and its usefulness should be reconsidered anyway. 2790 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2791} 2792 2793// ---------------------------------------------------------------------------- 2794AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2795 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2796 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2797 // mLeftVolFloat, mRightVolFloat 2798 // mLeftVolShort, mRightVolShort 2799{ 2800} 2801 2802AudioFlinger::DirectOutputThread::~DirectOutputThread() 2803{ 2804} 2805 2806AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2807 Vector< sp<Track> > *tracksToRemove 2808) 2809{ 2810 sp<Track> trackToRemove; 2811 2812 mixer_state mixerStatus = MIXER_IDLE; 2813 2814 // find out which tracks need to be processed 2815 if (mActiveTracks.size() != 0) { 2816 sp<Track> t = mActiveTracks[0].promote(); 2817 // The track died recently 2818 if (t == 0) return MIXER_IDLE; 2819 2820 Track* const track = t.get(); 2821 audio_track_cblk_t* cblk = track->cblk(); 2822 2823 // The first time a track is added we wait 2824 // for all its buffers to be filled before processing it 2825 if (cblk->framesReady() && track->isReady() && 2826 !track->isPaused() && !track->isTerminated()) 2827 { 2828 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2829 2830 if (track->mFillingUpStatus == Track::FS_FILLED) { 2831 track->mFillingUpStatus = Track::FS_ACTIVE; 2832 mLeftVolFloat = mRightVolFloat = 0; 2833 mLeftVolShort = mRightVolShort = 0; 2834 if (track->mState == TrackBase::RESUMING) { 2835 track->mState = TrackBase::ACTIVE; 2836 rampVolume = true; 2837 } 2838 } else if (cblk->server != 0) { 2839 // If the track is stopped before the first frame was mixed, 2840 // do not apply ramp 2841 rampVolume = true; 2842 } 2843 // compute volume for this track 2844 float left, right; 2845 if (track->isMuted() || mMasterMute || track->isPausing() || 2846 mStreamTypes[track->streamType()].mute) { 2847 left = right = 0; 2848 if (track->isPausing()) { 2849 track->setPaused(); 2850 } 2851 } else { 2852 float typeVolume = mStreamTypes[track->streamType()].volume; 2853 float v = mMasterVolume * typeVolume; 2854 uint32_t vlr = cblk->getVolumeLR(); 2855 float v_clamped = v * (vlr & 0xFFFF); 2856 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2857 left = v_clamped/MAX_GAIN; 2858 v_clamped = v * (vlr >> 16); 2859 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2860 right = v_clamped/MAX_GAIN; 2861 } 2862 2863 if (left != mLeftVolFloat || right != mRightVolFloat) { 2864 mLeftVolFloat = left; 2865 mRightVolFloat = right; 2866 2867 // If audio HAL implements volume control, 2868 // force software volume to nominal value 2869 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2870 left = 1.0f; 2871 right = 1.0f; 2872 } 2873 2874 // Convert volumes from float to 8.24 2875 uint32_t vl = (uint32_t)(left * (1 << 24)); 2876 uint32_t vr = (uint32_t)(right * (1 << 24)); 2877 2878 // Delegate volume control to effect in track effect chain if needed 2879 // only one effect chain can be present on DirectOutputThread, so if 2880 // there is one, the track is connected to it 2881 if (!mEffectChains.isEmpty()) { 2882 // Do not ramp volume if volume is controlled by effect 2883 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2884 rampVolume = false; 2885 } 2886 } 2887 2888 // Convert volumes from 8.24 to 4.12 format 2889 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2890 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2891 leftVol = (uint16_t)v_clamped; 2892 v_clamped = (vr + (1 << 11)) >> 12; 2893 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2894 rightVol = (uint16_t)v_clamped; 2895 } else { 2896 leftVol = mLeftVolShort; 2897 rightVol = mRightVolShort; 2898 rampVolume = false; 2899 } 2900 2901 // reset retry count 2902 track->mRetryCount = kMaxTrackRetriesDirect; 2903 mActiveTrack = t; 2904 mixerStatus = MIXER_TRACKS_READY; 2905 } else { 2906 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2907 if (track->isStopped()) { 2908 track->reset(); 2909 } 2910 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2911 // We have consumed all the buffers of this track. 2912 // Remove it from the list of active tracks. 2913 trackToRemove = track; 2914 } else { 2915 // No buffers for this track. Give it a few chances to 2916 // fill a buffer, then remove it from active list. 2917 if (--(track->mRetryCount) <= 0) { 2918 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2919 trackToRemove = track; 2920 } else { 2921 mixerStatus = MIXER_TRACKS_ENABLED; 2922 } 2923 } 2924 } 2925 } 2926 2927 // FIXME merge this with similar code for removing multiple tracks 2928 // remove all the tracks that need to be... 2929 if (CC_UNLIKELY(trackToRemove != 0)) { 2930 tracksToRemove->add(trackToRemove); 2931 mActiveTracks.remove(trackToRemove); 2932 if (!mEffectChains.isEmpty()) { 2933 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2934 trackToRemove->sessionId()); 2935 mEffectChains[0]->decActiveTrackCnt(); 2936 } 2937 if (trackToRemove->isTerminated()) { 2938 removeTrack_l(trackToRemove); 2939 } 2940 } 2941 2942 return mixerStatus; 2943} 2944 2945void AudioFlinger::DirectOutputThread::threadLoop_mix() 2946{ 2947 AudioBufferProvider::Buffer buffer; 2948 size_t frameCount = mFrameCount; 2949 int8_t *curBuf = (int8_t *)mMixBuffer; 2950 // output audio to hardware 2951 while (frameCount) { 2952 buffer.frameCount = frameCount; 2953 mActiveTrack->getNextBuffer(&buffer); 2954 if (CC_UNLIKELY(buffer.raw == NULL)) { 2955 memset(curBuf, 0, frameCount * mFrameSize); 2956 break; 2957 } 2958 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2959 frameCount -= buffer.frameCount; 2960 curBuf += buffer.frameCount * mFrameSize; 2961 mActiveTrack->releaseBuffer(&buffer); 2962 } 2963 sleepTime = 0; 2964 standbyTime = systemTime() + standbyDelay; 2965 mActiveTrack.clear(); 2966 2967 // apply volume 2968 2969 // Do not apply volume on compressed audio 2970 if (!audio_is_linear_pcm(mFormat)) { 2971 return; 2972 } 2973 2974 // convert to signed 16 bit before volume calculation 2975 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2976 size_t count = mFrameCount * mChannelCount; 2977 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2978 int16_t *dst = mMixBuffer + count-1; 2979 while (count--) { 2980 *dst-- = (int16_t)(*src--^0x80) << 8; 2981 } 2982 } 2983 2984 frameCount = mFrameCount; 2985 int16_t *out = mMixBuffer; 2986 if (rampVolume) { 2987 if (mChannelCount == 1) { 2988 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2989 int32_t vlInc = d / (int32_t)frameCount; 2990 int32_t vl = ((int32_t)mLeftVolShort << 16); 2991 do { 2992 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2993 out++; 2994 vl += vlInc; 2995 } while (--frameCount); 2996 2997 } else { 2998 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2999 int32_t vlInc = d / (int32_t)frameCount; 3000 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3001 int32_t vrInc = d / (int32_t)frameCount; 3002 int32_t vl = ((int32_t)mLeftVolShort << 16); 3003 int32_t vr = ((int32_t)mRightVolShort << 16); 3004 do { 3005 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3006 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3007 out += 2; 3008 vl += vlInc; 3009 vr += vrInc; 3010 } while (--frameCount); 3011 } 3012 } else { 3013 if (mChannelCount == 1) { 3014 do { 3015 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3016 out++; 3017 } while (--frameCount); 3018 } else { 3019 do { 3020 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3021 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3022 out += 2; 3023 } while (--frameCount); 3024 } 3025 } 3026 3027 // convert back to unsigned 8 bit after volume calculation 3028 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3029 size_t count = mFrameCount * mChannelCount; 3030 int16_t *src = mMixBuffer; 3031 uint8_t *dst = (uint8_t *)mMixBuffer; 3032 while (count--) { 3033 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3034 } 3035 } 3036 3037 mLeftVolShort = leftVol; 3038 mRightVolShort = rightVol; 3039} 3040 3041void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3042{ 3043 if (sleepTime == 0) { 3044 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3045 sleepTime = activeSleepTime; 3046 } else { 3047 sleepTime = idleSleepTime; 3048 } 3049 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3050 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3051 sleepTime = 0; 3052 } 3053} 3054 3055// getTrackName_l() must be called with ThreadBase::mLock held 3056int AudioFlinger::DirectOutputThread::getTrackName_l() 3057{ 3058 return 0; 3059} 3060 3061// deleteTrackName_l() must be called with ThreadBase::mLock held 3062void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3063{ 3064} 3065 3066// checkForNewParameters_l() must be called with ThreadBase::mLock held 3067bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3068{ 3069 bool reconfig = false; 3070 3071 while (!mNewParameters.isEmpty()) { 3072 status_t status = NO_ERROR; 3073 String8 keyValuePair = mNewParameters[0]; 3074 AudioParameter param = AudioParameter(keyValuePair); 3075 int value; 3076 3077 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3078 // do not accept frame count changes if tracks are open as the track buffer 3079 // size depends on frame count and correct behavior would not be garantied 3080 // if frame count is changed after track creation 3081 if (!mTracks.isEmpty()) { 3082 status = INVALID_OPERATION; 3083 } else { 3084 reconfig = true; 3085 } 3086 } 3087 if (status == NO_ERROR) { 3088 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3089 keyValuePair.string()); 3090 if (!mStandby && status == INVALID_OPERATION) { 3091 mOutput->stream->common.standby(&mOutput->stream->common); 3092 mStandby = true; 3093 mBytesWritten = 0; 3094 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3095 keyValuePair.string()); 3096 } 3097 if (status == NO_ERROR && reconfig) { 3098 readOutputParameters(); 3099 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3100 } 3101 } 3102 3103 mNewParameters.removeAt(0); 3104 3105 mParamStatus = status; 3106 mParamCond.signal(); 3107 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3108 // already timed out waiting for the status and will never signal the condition. 3109 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3110 } 3111 return reconfig; 3112} 3113 3114uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3115{ 3116 uint32_t time; 3117 if (audio_is_linear_pcm(mFormat)) { 3118 time = PlaybackThread::activeSleepTimeUs(); 3119 } else { 3120 time = 10000; 3121 } 3122 return time; 3123} 3124 3125uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3126{ 3127 uint32_t time; 3128 if (audio_is_linear_pcm(mFormat)) { 3129 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3130 } else { 3131 time = 10000; 3132 } 3133 return time; 3134} 3135 3136uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3137{ 3138 uint32_t time; 3139 if (audio_is_linear_pcm(mFormat)) { 3140 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3141 } else { 3142 time = 10000; 3143 } 3144 return time; 3145} 3146 3147void AudioFlinger::DirectOutputThread::cacheParameters_l() 3148{ 3149 PlaybackThread::cacheParameters_l(); 3150 3151 // use shorter standby delay as on normal output to release 3152 // hardware resources as soon as possible 3153 standbyDelay = microseconds(activeSleepTime*2); 3154} 3155 3156// ---------------------------------------------------------------------------- 3157 3158AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3159 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3160 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3161 mWaitTimeMs(UINT_MAX) 3162{ 3163 addOutputTrack(mainThread); 3164} 3165 3166AudioFlinger::DuplicatingThread::~DuplicatingThread() 3167{ 3168 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3169 mOutputTracks[i]->destroy(); 3170 } 3171} 3172 3173void AudioFlinger::DuplicatingThread::threadLoop_mix() 3174{ 3175 // mix buffers... 3176 if (outputsReady(outputTracks)) { 3177 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3178 } else { 3179 memset(mMixBuffer, 0, mixBufferSize); 3180 } 3181 sleepTime = 0; 3182 writeFrames = mFrameCount; 3183} 3184 3185void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3186{ 3187 if (sleepTime == 0) { 3188 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3189 sleepTime = activeSleepTime; 3190 } else { 3191 sleepTime = idleSleepTime; 3192 } 3193 } else if (mBytesWritten != 0) { 3194 // flush remaining overflow buffers in output tracks 3195 for (size_t i = 0; i < outputTracks.size(); i++) { 3196 if (outputTracks[i]->isActive()) { 3197 sleepTime = 0; 3198 writeFrames = 0; 3199 memset(mMixBuffer, 0, mixBufferSize); 3200 break; 3201 } 3202 } 3203 } 3204} 3205 3206void AudioFlinger::DuplicatingThread::threadLoop_write() 3207{ 3208 standbyTime = systemTime() + standbyDelay; 3209 for (size_t i = 0; i < outputTracks.size(); i++) { 3210 outputTracks[i]->write(mMixBuffer, writeFrames); 3211 } 3212 mBytesWritten += mixBufferSize; 3213} 3214 3215void AudioFlinger::DuplicatingThread::threadLoop_standby() 3216{ 3217 // DuplicatingThread implements standby by stopping all tracks 3218 for (size_t i = 0; i < outputTracks.size(); i++) { 3219 outputTracks[i]->stop(); 3220 } 3221} 3222 3223void AudioFlinger::DuplicatingThread::saveOutputTracks() 3224{ 3225 outputTracks = mOutputTracks; 3226} 3227 3228void AudioFlinger::DuplicatingThread::clearOutputTracks() 3229{ 3230 outputTracks.clear(); 3231} 3232 3233void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3234{ 3235 Mutex::Autolock _l(mLock); 3236 // FIXME explain this formula 3237 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3238 OutputTrack *outputTrack = new OutputTrack(thread, 3239 this, 3240 mSampleRate, 3241 mFormat, 3242 mChannelMask, 3243 frameCount); 3244 if (outputTrack->cblk() != NULL) { 3245 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3246 mOutputTracks.add(outputTrack); 3247 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3248 updateWaitTime_l(); 3249 } 3250} 3251 3252void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3253{ 3254 Mutex::Autolock _l(mLock); 3255 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3256 if (mOutputTracks[i]->thread() == thread) { 3257 mOutputTracks[i]->destroy(); 3258 mOutputTracks.removeAt(i); 3259 updateWaitTime_l(); 3260 return; 3261 } 3262 } 3263 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3264} 3265 3266// caller must hold mLock 3267void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3268{ 3269 mWaitTimeMs = UINT_MAX; 3270 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3271 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3272 if (strong != 0) { 3273 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3274 if (waitTimeMs < mWaitTimeMs) { 3275 mWaitTimeMs = waitTimeMs; 3276 } 3277 } 3278 } 3279} 3280 3281 3282bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3283{ 3284 for (size_t i = 0; i < outputTracks.size(); i++) { 3285 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3286 if (thread == 0) { 3287 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3288 return false; 3289 } 3290 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3291 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3292 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3293 return false; 3294 } 3295 } 3296 return true; 3297} 3298 3299uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3300{ 3301 return (mWaitTimeMs * 1000) / 2; 3302} 3303 3304void AudioFlinger::DuplicatingThread::cacheParameters_l() 3305{ 3306 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3307 updateWaitTime_l(); 3308 3309 MixerThread::cacheParameters_l(); 3310} 3311 3312// ---------------------------------------------------------------------------- 3313 3314// TrackBase constructor must be called with AudioFlinger::mLock held 3315AudioFlinger::ThreadBase::TrackBase::TrackBase( 3316 ThreadBase *thread, 3317 const sp<Client>& client, 3318 uint32_t sampleRate, 3319 audio_format_t format, 3320 uint32_t channelMask, 3321 int frameCount, 3322 const sp<IMemory>& sharedBuffer, 3323 int sessionId) 3324 : RefBase(), 3325 mThread(thread), 3326 mClient(client), 3327 mCblk(NULL), 3328 // mBuffer 3329 // mBufferEnd 3330 mFrameCount(0), 3331 mState(IDLE), 3332 mFormat(format), 3333 mStepServerFailed(false), 3334 mSessionId(sessionId) 3335 // mChannelCount 3336 // mChannelMask 3337{ 3338 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3339 3340 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3341 size_t size = sizeof(audio_track_cblk_t); 3342 uint8_t channelCount = popcount(channelMask); 3343 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3344 if (sharedBuffer == 0) { 3345 size += bufferSize; 3346 } 3347 3348 if (client != NULL) { 3349 mCblkMemory = client->heap()->allocate(size); 3350 if (mCblkMemory != 0) { 3351 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3352 if (mCblk != NULL) { // construct the shared structure in-place. 3353 new(mCblk) audio_track_cblk_t(); 3354 // clear all buffers 3355 mCblk->frameCount = frameCount; 3356 mCblk->sampleRate = sampleRate; 3357 mChannelCount = channelCount; 3358 mChannelMask = channelMask; 3359 if (sharedBuffer == 0) { 3360 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3361 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3362 // Force underrun condition to avoid false underrun callback until first data is 3363 // written to buffer (other flags are cleared) 3364 mCblk->flags = CBLK_UNDERRUN_ON; 3365 } else { 3366 mBuffer = sharedBuffer->pointer(); 3367 } 3368 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3369 } 3370 } else { 3371 ALOGE("not enough memory for AudioTrack size=%u", size); 3372 client->heap()->dump("AudioTrack"); 3373 return; 3374 } 3375 } else { 3376 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3377 // construct the shared structure in-place. 3378 new(mCblk) audio_track_cblk_t(); 3379 // clear all buffers 3380 mCblk->frameCount = frameCount; 3381 mCblk->sampleRate = sampleRate; 3382 mChannelCount = channelCount; 3383 mChannelMask = channelMask; 3384 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3385 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3386 // Force underrun condition to avoid false underrun callback until first data is 3387 // written to buffer (other flags are cleared) 3388 mCblk->flags = CBLK_UNDERRUN_ON; 3389 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3390 } 3391} 3392 3393AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3394{ 3395 if (mCblk != NULL) { 3396 if (mClient == 0) { 3397 delete mCblk; 3398 } else { 3399 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3400 } 3401 } 3402 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3403 if (mClient != 0) { 3404 // Client destructor must run with AudioFlinger mutex locked 3405 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3406 // If the client's reference count drops to zero, the associated destructor 3407 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3408 // relying on the automatic clear() at end of scope. 3409 mClient.clear(); 3410 } 3411} 3412 3413// AudioBufferProvider interface 3414// getNextBuffer() = 0; 3415// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3416void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3417{ 3418 buffer->raw = NULL; 3419 mFrameCount = buffer->frameCount; 3420 (void) step(); // ignore return value of step() 3421 buffer->frameCount = 0; 3422} 3423 3424bool AudioFlinger::ThreadBase::TrackBase::step() { 3425 bool result; 3426 audio_track_cblk_t* cblk = this->cblk(); 3427 3428 result = cblk->stepServer(mFrameCount); 3429 if (!result) { 3430 ALOGV("stepServer failed acquiring cblk mutex"); 3431 mStepServerFailed = true; 3432 } 3433 return result; 3434} 3435 3436void AudioFlinger::ThreadBase::TrackBase::reset() { 3437 audio_track_cblk_t* cblk = this->cblk(); 3438 3439 cblk->user = 0; 3440 cblk->server = 0; 3441 cblk->userBase = 0; 3442 cblk->serverBase = 0; 3443 mStepServerFailed = false; 3444 ALOGV("TrackBase::reset"); 3445} 3446 3447int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3448 return (int)mCblk->sampleRate; 3449} 3450 3451void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3452 audio_track_cblk_t* cblk = this->cblk(); 3453 size_t frameSize = cblk->frameSize; 3454 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3455 int8_t *bufferEnd = bufferStart + frames * frameSize; 3456 3457 // Check validity of returned pointer in case the track control block would have been corrupted. 3458 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3459 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3460 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3461 server %d, serverBase %d, user %d, userBase %d", 3462 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3463 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3464 return NULL; 3465 } 3466 3467 return bufferStart; 3468} 3469 3470// ---------------------------------------------------------------------------- 3471 3472// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3473AudioFlinger::PlaybackThread::Track::Track( 3474 PlaybackThread *thread, 3475 const sp<Client>& client, 3476 audio_stream_type_t streamType, 3477 uint32_t sampleRate, 3478 audio_format_t format, 3479 uint32_t channelMask, 3480 int frameCount, 3481 const sp<IMemory>& sharedBuffer, 3482 int sessionId) 3483 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3484 mMute(false), 3485 // mFillingUpStatus ? 3486 // mRetryCount initialized later when needed 3487 mSharedBuffer(sharedBuffer), 3488 mStreamType(streamType), 3489 mName(-1), // see note below 3490 mMainBuffer(thread->mixBuffer()), 3491 mAuxBuffer(NULL), 3492 mAuxEffectId(0), mHasVolumeController(false) 3493{ 3494 if (mCblk != NULL) { 3495 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3496 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3497 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3498 // to avoid leaking a track name, do not allocate one unless there is an mCblk 3499 mName = thread->getTrackName_l(); 3500 if (mName < 0) { 3501 ALOGE("no more track names available"); 3502 } 3503 } 3504 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3505} 3506 3507AudioFlinger::PlaybackThread::Track::~Track() 3508{ 3509 ALOGV("PlaybackThread::Track destructor"); 3510 sp<ThreadBase> thread = mThread.promote(); 3511 if (thread != 0) { 3512 Mutex::Autolock _l(thread->mLock); 3513 mState = TERMINATED; 3514 } 3515} 3516 3517void AudioFlinger::PlaybackThread::Track::destroy() 3518{ 3519 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3520 // by removing it from mTracks vector, so there is a risk that this Tracks's 3521 // destructor is called. As the destructor needs to lock mLock, 3522 // we must acquire a strong reference on this Track before locking mLock 3523 // here so that the destructor is called only when exiting this function. 3524 // On the other hand, as long as Track::destroy() is only called by 3525 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3526 // this Track with its member mTrack. 3527 sp<Track> keep(this); 3528 { // scope for mLock 3529 sp<ThreadBase> thread = mThread.promote(); 3530 if (thread != 0) { 3531 if (!isOutputTrack()) { 3532 if (mState == ACTIVE || mState == RESUMING) { 3533 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3534 3535#ifdef ADD_BATTERY_DATA 3536 // to track the speaker usage 3537 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3538#endif 3539 } 3540 AudioSystem::releaseOutput(thread->id()); 3541 } 3542 Mutex::Autolock _l(thread->mLock); 3543 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3544 playbackThread->destroyTrack_l(this); 3545 } 3546 } 3547} 3548 3549void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3550{ 3551 uint32_t vlr = mCblk->getVolumeLR(); 3552 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", 3553 mName - AudioMixer::TRACK0, 3554 (mClient == 0) ? getpid_cached : mClient->pid(), 3555 mStreamType, 3556 mFormat, 3557 mChannelMask, 3558 mSessionId, 3559 mFrameCount, 3560 mState, 3561 mMute, 3562 mFillingUpStatus, 3563 mCblk->sampleRate, 3564 vlr & 0xFFFF, 3565 vlr >> 16, 3566 mCblk->server, 3567 mCblk->user, 3568 (int)mMainBuffer, 3569 (int)mAuxBuffer); 3570} 3571 3572// AudioBufferProvider interface 3573status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3574 AudioBufferProvider::Buffer* buffer, int64_t pts) 3575{ 3576 audio_track_cblk_t* cblk = this->cblk(); 3577 uint32_t framesReady; 3578 uint32_t framesReq = buffer->frameCount; 3579 3580 // Check if last stepServer failed, try to step now 3581 if (mStepServerFailed) { 3582 if (!step()) goto getNextBuffer_exit; 3583 ALOGV("stepServer recovered"); 3584 mStepServerFailed = false; 3585 } 3586 3587 framesReady = cblk->framesReady(); 3588 3589 if (CC_LIKELY(framesReady)) { 3590 uint32_t s = cblk->server; 3591 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3592 3593 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3594 if (framesReq > framesReady) { 3595 framesReq = framesReady; 3596 } 3597 if (s + framesReq > bufferEnd) { 3598 framesReq = bufferEnd - s; 3599 } 3600 3601 buffer->raw = getBuffer(s, framesReq); 3602 if (buffer->raw == NULL) goto getNextBuffer_exit; 3603 3604 buffer->frameCount = framesReq; 3605 return NO_ERROR; 3606 } 3607 3608getNextBuffer_exit: 3609 buffer->raw = NULL; 3610 buffer->frameCount = 0; 3611 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3612 return NOT_ENOUGH_DATA; 3613} 3614 3615uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3616 return mCblk->framesReady(); 3617} 3618 3619bool AudioFlinger::PlaybackThread::Track::isReady() const { 3620 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3621 3622 if (framesReady() >= mCblk->frameCount || 3623 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3624 mFillingUpStatus = FS_FILLED; 3625 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3626 return true; 3627 } 3628 return false; 3629} 3630 3631status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3632{ 3633 status_t status = NO_ERROR; 3634 ALOGV("start(%d), calling pid %d session %d tid %d", 3635 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3636 sp<ThreadBase> thread = mThread.promote(); 3637 if (thread != 0) { 3638 Mutex::Autolock _l(thread->mLock); 3639 track_state state = mState; 3640 // here the track could be either new, or restarted 3641 // in both cases "unstop" the track 3642 if (mState == PAUSED) { 3643 mState = TrackBase::RESUMING; 3644 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3645 } else { 3646 mState = TrackBase::ACTIVE; 3647 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3648 } 3649 3650 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3651 thread->mLock.unlock(); 3652 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3653 thread->mLock.lock(); 3654 3655#ifdef ADD_BATTERY_DATA 3656 // to track the speaker usage 3657 if (status == NO_ERROR) { 3658 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3659 } 3660#endif 3661 } 3662 if (status == NO_ERROR) { 3663 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3664 playbackThread->addTrack_l(this); 3665 } else { 3666 mState = state; 3667 } 3668 } else { 3669 status = BAD_VALUE; 3670 } 3671 return status; 3672} 3673 3674void AudioFlinger::PlaybackThread::Track::stop() 3675{ 3676 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3677 sp<ThreadBase> thread = mThread.promote(); 3678 if (thread != 0) { 3679 Mutex::Autolock _l(thread->mLock); 3680 track_state state = mState; 3681 if (mState > STOPPED) { 3682 mState = STOPPED; 3683 // If the track is not active (PAUSED and buffers full), flush buffers 3684 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3685 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3686 reset(); 3687 } 3688 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3689 } 3690 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3691 thread->mLock.unlock(); 3692 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3693 thread->mLock.lock(); 3694 3695#ifdef ADD_BATTERY_DATA 3696 // to track the speaker usage 3697 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3698#endif 3699 } 3700 } 3701} 3702 3703void AudioFlinger::PlaybackThread::Track::pause() 3704{ 3705 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3706 sp<ThreadBase> thread = mThread.promote(); 3707 if (thread != 0) { 3708 Mutex::Autolock _l(thread->mLock); 3709 if (mState == ACTIVE || mState == RESUMING) { 3710 mState = PAUSING; 3711 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3712 if (!isOutputTrack()) { 3713 thread->mLock.unlock(); 3714 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3715 thread->mLock.lock(); 3716 3717#ifdef ADD_BATTERY_DATA 3718 // to track the speaker usage 3719 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3720#endif 3721 } 3722 } 3723 } 3724} 3725 3726void AudioFlinger::PlaybackThread::Track::flush() 3727{ 3728 ALOGV("flush(%d)", mName); 3729 sp<ThreadBase> thread = mThread.promote(); 3730 if (thread != 0) { 3731 Mutex::Autolock _l(thread->mLock); 3732 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3733 return; 3734 } 3735 // No point remaining in PAUSED state after a flush => go to 3736 // STOPPED state 3737 mState = STOPPED; 3738 3739 // do not reset the track if it is still in the process of being stopped or paused. 3740 // this will be done by prepareTracks_l() when the track is stopped. 3741 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3742 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3743 reset(); 3744 } 3745 } 3746} 3747 3748void AudioFlinger::PlaybackThread::Track::reset() 3749{ 3750 // Do not reset twice to avoid discarding data written just after a flush and before 3751 // the audioflinger thread detects the track is stopped. 3752 if (!mResetDone) { 3753 TrackBase::reset(); 3754 // Force underrun condition to avoid false underrun callback until first data is 3755 // written to buffer 3756 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3757 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3758 mFillingUpStatus = FS_FILLING; 3759 mResetDone = true; 3760 } 3761} 3762 3763void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3764{ 3765 mMute = muted; 3766} 3767 3768status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3769{ 3770 status_t status = DEAD_OBJECT; 3771 sp<ThreadBase> thread = mThread.promote(); 3772 if (thread != 0) { 3773 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3774 status = playbackThread->attachAuxEffect(this, EffectId); 3775 } 3776 return status; 3777} 3778 3779void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3780{ 3781 mAuxEffectId = EffectId; 3782 mAuxBuffer = buffer; 3783} 3784 3785// timed audio tracks 3786 3787sp<AudioFlinger::PlaybackThread::TimedTrack> 3788AudioFlinger::PlaybackThread::TimedTrack::create( 3789 PlaybackThread *thread, 3790 const sp<Client>& client, 3791 audio_stream_type_t streamType, 3792 uint32_t sampleRate, 3793 audio_format_t format, 3794 uint32_t channelMask, 3795 int frameCount, 3796 const sp<IMemory>& sharedBuffer, 3797 int sessionId) { 3798 if (!client->reserveTimedTrack()) 3799 return NULL; 3800 3801 return new TimedTrack( 3802 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3803 sharedBuffer, sessionId); 3804} 3805 3806AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3807 PlaybackThread *thread, 3808 const sp<Client>& client, 3809 audio_stream_type_t streamType, 3810 uint32_t sampleRate, 3811 audio_format_t format, 3812 uint32_t channelMask, 3813 int frameCount, 3814 const sp<IMemory>& sharedBuffer, 3815 int sessionId) 3816 : Track(thread, client, streamType, sampleRate, format, channelMask, 3817 frameCount, sharedBuffer, sessionId), 3818 mTimedSilenceBuffer(NULL), 3819 mTimedSilenceBufferSize(0), 3820 mTimedAudioOutputOnTime(false), 3821 mMediaTimeTransformValid(false) 3822{ 3823 LocalClock lc; 3824 mLocalTimeFreq = lc.getLocalFreq(); 3825 3826 mLocalTimeToSampleTransform.a_zero = 0; 3827 mLocalTimeToSampleTransform.b_zero = 0; 3828 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3829 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3830 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3831 &mLocalTimeToSampleTransform.a_to_b_denom); 3832} 3833 3834AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3835 mClient->releaseTimedTrack(); 3836 delete [] mTimedSilenceBuffer; 3837} 3838 3839status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3840 size_t size, sp<IMemory>* buffer) { 3841 3842 Mutex::Autolock _l(mTimedBufferQueueLock); 3843 3844 trimTimedBufferQueue_l(); 3845 3846 // lazily initialize the shared memory heap for timed buffers 3847 if (mTimedMemoryDealer == NULL) { 3848 const int kTimedBufferHeapSize = 512 << 10; 3849 3850 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3851 "AudioFlingerTimed"); 3852 if (mTimedMemoryDealer == NULL) 3853 return NO_MEMORY; 3854 } 3855 3856 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3857 if (newBuffer == NULL) { 3858 newBuffer = mTimedMemoryDealer->allocate(size); 3859 if (newBuffer == NULL) 3860 return NO_MEMORY; 3861 } 3862 3863 *buffer = newBuffer; 3864 return NO_ERROR; 3865} 3866 3867// caller must hold mTimedBufferQueueLock 3868void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3869 int64_t mediaTimeNow; 3870 { 3871 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3872 if (!mMediaTimeTransformValid) 3873 return; 3874 3875 int64_t targetTimeNow; 3876 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3877 ? mCCHelper.getCommonTime(&targetTimeNow) 3878 : mCCHelper.getLocalTime(&targetTimeNow); 3879 3880 if (OK != res) 3881 return; 3882 3883 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3884 &mediaTimeNow)) { 3885 return; 3886 } 3887 } 3888 3889 size_t trimIndex; 3890 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3891 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3892 break; 3893 } 3894 3895 if (trimIndex) { 3896 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3897 } 3898} 3899 3900status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3901 const sp<IMemory>& buffer, int64_t pts) { 3902 3903 { 3904 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3905 if (!mMediaTimeTransformValid) 3906 return INVALID_OPERATION; 3907 } 3908 3909 Mutex::Autolock _l(mTimedBufferQueueLock); 3910 3911 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3912 3913 return NO_ERROR; 3914} 3915 3916status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3917 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3918 3919 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3920 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3921 target); 3922 3923 if (!(target == TimedAudioTrack::LOCAL_TIME || 3924 target == TimedAudioTrack::COMMON_TIME)) { 3925 return BAD_VALUE; 3926 } 3927 3928 Mutex::Autolock lock(mMediaTimeTransformLock); 3929 mMediaTimeTransform = xform; 3930 mMediaTimeTransformTarget = target; 3931 mMediaTimeTransformValid = true; 3932 3933 return NO_ERROR; 3934} 3935 3936#define min(a, b) ((a) < (b) ? (a) : (b)) 3937 3938// implementation of getNextBuffer for tracks whose buffers have timestamps 3939status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3940 AudioBufferProvider::Buffer* buffer, int64_t pts) 3941{ 3942 if (pts == AudioBufferProvider::kInvalidPTS) { 3943 buffer->raw = 0; 3944 buffer->frameCount = 0; 3945 return INVALID_OPERATION; 3946 } 3947 3948 Mutex::Autolock _l(mTimedBufferQueueLock); 3949 3950 while (true) { 3951 3952 // if we have no timed buffers, then fail 3953 if (mTimedBufferQueue.isEmpty()) { 3954 buffer->raw = 0; 3955 buffer->frameCount = 0; 3956 return NOT_ENOUGH_DATA; 3957 } 3958 3959 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3960 3961 // calculate the PTS of the head of the timed buffer queue expressed in 3962 // local time 3963 int64_t headLocalPTS; 3964 { 3965 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3966 3967 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 3968 3969 if (mMediaTimeTransform.a_to_b_denom == 0) { 3970 // the transform represents a pause, so yield silence 3971 timedYieldSilence(buffer->frameCount, buffer); 3972 return NO_ERROR; 3973 } 3974 3975 int64_t transformedPTS; 3976 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3977 &transformedPTS)) { 3978 // the transform failed. this shouldn't happen, but if it does 3979 // then just drop this buffer 3980 ALOGW("timedGetNextBuffer transform failed"); 3981 buffer->raw = 0; 3982 buffer->frameCount = 0; 3983 mTimedBufferQueue.removeAt(0); 3984 return NO_ERROR; 3985 } 3986 3987 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3988 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3989 &headLocalPTS)) { 3990 buffer->raw = 0; 3991 buffer->frameCount = 0; 3992 return INVALID_OPERATION; 3993 } 3994 } else { 3995 headLocalPTS = transformedPTS; 3996 } 3997 } 3998 3999 // adjust the head buffer's PTS to reflect the portion of the head buffer 4000 // that has already been consumed 4001 int64_t effectivePTS = headLocalPTS + 4002 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4003 4004 // Calculate the delta in samples between the head of the input buffer 4005 // queue and the start of the next output buffer that will be written. 4006 // If the transformation fails because of over or underflow, it means 4007 // that the sample's position in the output stream is so far out of 4008 // whack that it should just be dropped. 4009 int64_t sampleDelta; 4010 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4011 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4012 mTimedBufferQueue.removeAt(0); 4013 continue; 4014 } 4015 if (!mLocalTimeToSampleTransform.doForwardTransform( 4016 (effectivePTS - pts) << 32, &sampleDelta)) { 4017 ALOGV("*** too late during sample rate transform: dropped buffer"); 4018 mTimedBufferQueue.removeAt(0); 4019 continue; 4020 } 4021 4022 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4023 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4024 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4025 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4026 4027 // if the delta between the ideal placement for the next input sample and 4028 // the current output position is within this threshold, then we will 4029 // concatenate the next input samples to the previous output 4030 const int64_t kSampleContinuityThreshold = 4031 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4032 4033 // if this is the first buffer of audio that we're emitting from this track 4034 // then it should be almost exactly on time. 4035 const int64_t kSampleStartupThreshold = 1LL << 32; 4036 4037 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4038 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4039 // the next input is close enough to being on time, so concatenate it 4040 // with the last output 4041 timedYieldSamples(buffer); 4042 4043 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4044 return NO_ERROR; 4045 } else if (sampleDelta > 0) { 4046 // the gap between the current output position and the proper start of 4047 // the next input sample is too big, so fill it with silence 4048 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4049 4050 timedYieldSilence(framesUntilNextInput, buffer); 4051 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4052 return NO_ERROR; 4053 } else { 4054 // the next input sample is late 4055 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4056 size_t onTimeSamplePosition = 4057 head.position() + lateFrames * mCblk->frameSize; 4058 4059 if (onTimeSamplePosition > head.buffer()->size()) { 4060 // all the remaining samples in the head are too late, so 4061 // drop it and move on 4062 ALOGV("*** too late: dropped buffer"); 4063 mTimedBufferQueue.removeAt(0); 4064 continue; 4065 } else { 4066 // skip over the late samples 4067 head.setPosition(onTimeSamplePosition); 4068 4069 // yield the available samples 4070 timedYieldSamples(buffer); 4071 4072 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4073 return NO_ERROR; 4074 } 4075 } 4076 } 4077} 4078 4079// Yield samples from the timed buffer queue head up to the given output 4080// buffer's capacity. 4081// 4082// Caller must hold mTimedBufferQueueLock 4083void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4084 AudioBufferProvider::Buffer* buffer) { 4085 4086 const TimedBuffer& head = mTimedBufferQueue[0]; 4087 4088 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4089 head.position()); 4090 4091 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4092 mCblk->frameSize); 4093 size_t framesRequested = buffer->frameCount; 4094 buffer->frameCount = min(framesLeftInHead, framesRequested); 4095 4096 mTimedAudioOutputOnTime = true; 4097} 4098 4099// Yield samples of silence up to the given output buffer's capacity 4100// 4101// Caller must hold mTimedBufferQueueLock 4102void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4103 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4104 4105 // lazily allocate a buffer filled with silence 4106 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4107 delete [] mTimedSilenceBuffer; 4108 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4109 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4110 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4111 } 4112 4113 buffer->raw = mTimedSilenceBuffer; 4114 size_t framesRequested = buffer->frameCount; 4115 buffer->frameCount = min(numFrames, framesRequested); 4116 4117 mTimedAudioOutputOnTime = false; 4118} 4119 4120// AudioBufferProvider interface 4121void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4122 AudioBufferProvider::Buffer* buffer) { 4123 4124 Mutex::Autolock _l(mTimedBufferQueueLock); 4125 4126 // If the buffer which was just released is part of the buffer at the head 4127 // of the queue, be sure to update the amt of the buffer which has been 4128 // consumed. If the buffer being returned is not part of the head of the 4129 // queue, its either because the buffer is part of the silence buffer, or 4130 // because the head of the timed queue was trimmed after the mixer called 4131 // getNextBuffer but before the mixer called releaseBuffer. 4132 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4133 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4134 4135 void* start = head.buffer()->pointer(); 4136 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4137 4138 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4139 head.setPosition(head.position() + 4140 (buffer->frameCount * mCblk->frameSize)); 4141 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4142 mTimedBufferQueue.removeAt(0); 4143 } 4144 } 4145 } 4146 4147 buffer->raw = 0; 4148 buffer->frameCount = 0; 4149} 4150 4151uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4152 Mutex::Autolock _l(mTimedBufferQueueLock); 4153 4154 uint32_t frames = 0; 4155 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4156 const TimedBuffer& tb = mTimedBufferQueue[i]; 4157 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4158 } 4159 4160 return frames; 4161} 4162 4163AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4164 : mPTS(0), mPosition(0) {} 4165 4166AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4167 const sp<IMemory>& buffer, int64_t pts) 4168 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4169 4170// ---------------------------------------------------------------------------- 4171 4172// RecordTrack constructor must be called with AudioFlinger::mLock held 4173AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4174 RecordThread *thread, 4175 const sp<Client>& client, 4176 uint32_t sampleRate, 4177 audio_format_t format, 4178 uint32_t channelMask, 4179 int frameCount, 4180 int sessionId) 4181 : TrackBase(thread, client, sampleRate, format, 4182 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4183 mOverflow(false) 4184{ 4185 if (mCblk != NULL) { 4186 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4187 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4188 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4189 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4190 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4191 } else { 4192 mCblk->frameSize = sizeof(int8_t); 4193 } 4194 } 4195} 4196 4197AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4198{ 4199 sp<ThreadBase> thread = mThread.promote(); 4200 if (thread != 0) { 4201 AudioSystem::releaseInput(thread->id()); 4202 } 4203} 4204 4205// AudioBufferProvider interface 4206status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4207{ 4208 audio_track_cblk_t* cblk = this->cblk(); 4209 uint32_t framesAvail; 4210 uint32_t framesReq = buffer->frameCount; 4211 4212 // Check if last stepServer failed, try to step now 4213 if (mStepServerFailed) { 4214 if (!step()) goto getNextBuffer_exit; 4215 ALOGV("stepServer recovered"); 4216 mStepServerFailed = false; 4217 } 4218 4219 framesAvail = cblk->framesAvailable_l(); 4220 4221 if (CC_LIKELY(framesAvail)) { 4222 uint32_t s = cblk->server; 4223 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4224 4225 if (framesReq > framesAvail) { 4226 framesReq = framesAvail; 4227 } 4228 if (s + framesReq > bufferEnd) { 4229 framesReq = bufferEnd - s; 4230 } 4231 4232 buffer->raw = getBuffer(s, framesReq); 4233 if (buffer->raw == NULL) goto getNextBuffer_exit; 4234 4235 buffer->frameCount = framesReq; 4236 return NO_ERROR; 4237 } 4238 4239getNextBuffer_exit: 4240 buffer->raw = NULL; 4241 buffer->frameCount = 0; 4242 return NOT_ENOUGH_DATA; 4243} 4244 4245status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4246{ 4247 sp<ThreadBase> thread = mThread.promote(); 4248 if (thread != 0) { 4249 RecordThread *recordThread = (RecordThread *)thread.get(); 4250 return recordThread->start(this, tid); 4251 } else { 4252 return BAD_VALUE; 4253 } 4254} 4255 4256void AudioFlinger::RecordThread::RecordTrack::stop() 4257{ 4258 sp<ThreadBase> thread = mThread.promote(); 4259 if (thread != 0) { 4260 RecordThread *recordThread = (RecordThread *)thread.get(); 4261 recordThread->stop(this); 4262 TrackBase::reset(); 4263 // Force overrun condition to avoid false overrun callback until first data is 4264 // read from buffer 4265 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4266 } 4267} 4268 4269void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4270{ 4271 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4272 (mClient == 0) ? getpid_cached : mClient->pid(), 4273 mFormat, 4274 mChannelMask, 4275 mSessionId, 4276 mFrameCount, 4277 mState, 4278 mCblk->sampleRate, 4279 mCblk->server, 4280 mCblk->user); 4281} 4282 4283 4284// ---------------------------------------------------------------------------- 4285 4286AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4287 PlaybackThread *playbackThread, 4288 DuplicatingThread *sourceThread, 4289 uint32_t sampleRate, 4290 audio_format_t format, 4291 uint32_t channelMask, 4292 int frameCount) 4293 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4294 mActive(false), mSourceThread(sourceThread) 4295{ 4296 4297 if (mCblk != NULL) { 4298 mCblk->flags |= CBLK_DIRECTION_OUT; 4299 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4300 mOutBuffer.frameCount = 0; 4301 playbackThread->mTracks.add(this); 4302 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4303 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4304 mCblk, mBuffer, mCblk->buffers, 4305 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4306 } else { 4307 ALOGW("Error creating output track on thread %p", playbackThread); 4308 } 4309} 4310 4311AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4312{ 4313 clearBufferQueue(); 4314} 4315 4316status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4317{ 4318 status_t status = Track::start(tid); 4319 if (status != NO_ERROR) { 4320 return status; 4321 } 4322 4323 mActive = true; 4324 mRetryCount = 127; 4325 return status; 4326} 4327 4328void AudioFlinger::PlaybackThread::OutputTrack::stop() 4329{ 4330 Track::stop(); 4331 clearBufferQueue(); 4332 mOutBuffer.frameCount = 0; 4333 mActive = false; 4334} 4335 4336bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4337{ 4338 Buffer *pInBuffer; 4339 Buffer inBuffer; 4340 uint32_t channelCount = mChannelCount; 4341 bool outputBufferFull = false; 4342 inBuffer.frameCount = frames; 4343 inBuffer.i16 = data; 4344 4345 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4346 4347 if (!mActive && frames != 0) { 4348 start(0); 4349 sp<ThreadBase> thread = mThread.promote(); 4350 if (thread != 0) { 4351 MixerThread *mixerThread = (MixerThread *)thread.get(); 4352 if (mCblk->frameCount > frames){ 4353 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4354 uint32_t startFrames = (mCblk->frameCount - frames); 4355 pInBuffer = new Buffer; 4356 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4357 pInBuffer->frameCount = startFrames; 4358 pInBuffer->i16 = pInBuffer->mBuffer; 4359 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4360 mBufferQueue.add(pInBuffer); 4361 } else { 4362 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4363 } 4364 } 4365 } 4366 } 4367 4368 while (waitTimeLeftMs) { 4369 // First write pending buffers, then new data 4370 if (mBufferQueue.size()) { 4371 pInBuffer = mBufferQueue.itemAt(0); 4372 } else { 4373 pInBuffer = &inBuffer; 4374 } 4375 4376 if (pInBuffer->frameCount == 0) { 4377 break; 4378 } 4379 4380 if (mOutBuffer.frameCount == 0) { 4381 mOutBuffer.frameCount = pInBuffer->frameCount; 4382 nsecs_t startTime = systemTime(); 4383 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4384 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4385 outputBufferFull = true; 4386 break; 4387 } 4388 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4389 if (waitTimeLeftMs >= waitTimeMs) { 4390 waitTimeLeftMs -= waitTimeMs; 4391 } else { 4392 waitTimeLeftMs = 0; 4393 } 4394 } 4395 4396 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4397 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4398 mCblk->stepUser(outFrames); 4399 pInBuffer->frameCount -= outFrames; 4400 pInBuffer->i16 += outFrames * channelCount; 4401 mOutBuffer.frameCount -= outFrames; 4402 mOutBuffer.i16 += outFrames * channelCount; 4403 4404 if (pInBuffer->frameCount == 0) { 4405 if (mBufferQueue.size()) { 4406 mBufferQueue.removeAt(0); 4407 delete [] pInBuffer->mBuffer; 4408 delete pInBuffer; 4409 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4410 } else { 4411 break; 4412 } 4413 } 4414 } 4415 4416 // If we could not write all frames, allocate a buffer and queue it for next time. 4417 if (inBuffer.frameCount) { 4418 sp<ThreadBase> thread = mThread.promote(); 4419 if (thread != 0 && !thread->standby()) { 4420 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4421 pInBuffer = new Buffer; 4422 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4423 pInBuffer->frameCount = inBuffer.frameCount; 4424 pInBuffer->i16 = pInBuffer->mBuffer; 4425 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4426 mBufferQueue.add(pInBuffer); 4427 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4428 } else { 4429 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4430 } 4431 } 4432 } 4433 4434 // Calling write() with a 0 length buffer, means that no more data will be written: 4435 // If no more buffers are pending, fill output track buffer to make sure it is started 4436 // by output mixer. 4437 if (frames == 0 && mBufferQueue.size() == 0) { 4438 if (mCblk->user < mCblk->frameCount) { 4439 frames = mCblk->frameCount - mCblk->user; 4440 pInBuffer = new Buffer; 4441 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4442 pInBuffer->frameCount = frames; 4443 pInBuffer->i16 = pInBuffer->mBuffer; 4444 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4445 mBufferQueue.add(pInBuffer); 4446 } else if (mActive) { 4447 stop(); 4448 } 4449 } 4450 4451 return outputBufferFull; 4452} 4453 4454status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4455{ 4456 int active; 4457 status_t result; 4458 audio_track_cblk_t* cblk = mCblk; 4459 uint32_t framesReq = buffer->frameCount; 4460 4461// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4462 buffer->frameCount = 0; 4463 4464 uint32_t framesAvail = cblk->framesAvailable(); 4465 4466 4467 if (framesAvail == 0) { 4468 Mutex::Autolock _l(cblk->lock); 4469 goto start_loop_here; 4470 while (framesAvail == 0) { 4471 active = mActive; 4472 if (CC_UNLIKELY(!active)) { 4473 ALOGV("Not active and NO_MORE_BUFFERS"); 4474 return NO_MORE_BUFFERS; 4475 } 4476 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4477 if (result != NO_ERROR) { 4478 return NO_MORE_BUFFERS; 4479 } 4480 // read the server count again 4481 start_loop_here: 4482 framesAvail = cblk->framesAvailable_l(); 4483 } 4484 } 4485 4486// if (framesAvail < framesReq) { 4487// return NO_MORE_BUFFERS; 4488// } 4489 4490 if (framesReq > framesAvail) { 4491 framesReq = framesAvail; 4492 } 4493 4494 uint32_t u = cblk->user; 4495 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4496 4497 if (u + framesReq > bufferEnd) { 4498 framesReq = bufferEnd - u; 4499 } 4500 4501 buffer->frameCount = framesReq; 4502 buffer->raw = (void *)cblk->buffer(u); 4503 return NO_ERROR; 4504} 4505 4506 4507void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4508{ 4509 size_t size = mBufferQueue.size(); 4510 4511 for (size_t i = 0; i < size; i++) { 4512 Buffer *pBuffer = mBufferQueue.itemAt(i); 4513 delete [] pBuffer->mBuffer; 4514 delete pBuffer; 4515 } 4516 mBufferQueue.clear(); 4517} 4518 4519// ---------------------------------------------------------------------------- 4520 4521AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4522 : RefBase(), 4523 mAudioFlinger(audioFlinger), 4524 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4525 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4526 mPid(pid), 4527 mTimedTrackCount(0) 4528{ 4529 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4530} 4531 4532// Client destructor must be called with AudioFlinger::mLock held 4533AudioFlinger::Client::~Client() 4534{ 4535 mAudioFlinger->removeClient_l(mPid); 4536} 4537 4538sp<MemoryDealer> AudioFlinger::Client::heap() const 4539{ 4540 return mMemoryDealer; 4541} 4542 4543// Reserve one of the limited slots for a timed audio track associated 4544// with this client 4545bool AudioFlinger::Client::reserveTimedTrack() 4546{ 4547 const int kMaxTimedTracksPerClient = 4; 4548 4549 Mutex::Autolock _l(mTimedTrackLock); 4550 4551 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4552 ALOGW("can not create timed track - pid %d has exceeded the limit", 4553 mPid); 4554 return false; 4555 } 4556 4557 mTimedTrackCount++; 4558 return true; 4559} 4560 4561// Release a slot for a timed audio track 4562void AudioFlinger::Client::releaseTimedTrack() 4563{ 4564 Mutex::Autolock _l(mTimedTrackLock); 4565 mTimedTrackCount--; 4566} 4567 4568// ---------------------------------------------------------------------------- 4569 4570AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4571 const sp<IAudioFlingerClient>& client, 4572 pid_t pid) 4573 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4574{ 4575} 4576 4577AudioFlinger::NotificationClient::~NotificationClient() 4578{ 4579} 4580 4581void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4582{ 4583 sp<NotificationClient> keep(this); 4584 mAudioFlinger->removeNotificationClient(mPid); 4585} 4586 4587// ---------------------------------------------------------------------------- 4588 4589AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4590 : BnAudioTrack(), 4591 mTrack(track) 4592{ 4593} 4594 4595AudioFlinger::TrackHandle::~TrackHandle() { 4596 // just stop the track on deletion, associated resources 4597 // will be freed from the main thread once all pending buffers have 4598 // been played. Unless it's not in the active track list, in which 4599 // case we free everything now... 4600 mTrack->destroy(); 4601} 4602 4603sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4604 return mTrack->getCblk(); 4605} 4606 4607status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4608 return mTrack->start(tid); 4609} 4610 4611void AudioFlinger::TrackHandle::stop() { 4612 mTrack->stop(); 4613} 4614 4615void AudioFlinger::TrackHandle::flush() { 4616 mTrack->flush(); 4617} 4618 4619void AudioFlinger::TrackHandle::mute(bool e) { 4620 mTrack->mute(e); 4621} 4622 4623void AudioFlinger::TrackHandle::pause() { 4624 mTrack->pause(); 4625} 4626 4627status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4628{ 4629 return mTrack->attachAuxEffect(EffectId); 4630} 4631 4632status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4633 sp<IMemory>* buffer) { 4634 if (!mTrack->isTimedTrack()) 4635 return INVALID_OPERATION; 4636 4637 PlaybackThread::TimedTrack* tt = 4638 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4639 return tt->allocateTimedBuffer(size, buffer); 4640} 4641 4642status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4643 int64_t pts) { 4644 if (!mTrack->isTimedTrack()) 4645 return INVALID_OPERATION; 4646 4647 PlaybackThread::TimedTrack* tt = 4648 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4649 return tt->queueTimedBuffer(buffer, pts); 4650} 4651 4652status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4653 const LinearTransform& xform, int target) { 4654 4655 if (!mTrack->isTimedTrack()) 4656 return INVALID_OPERATION; 4657 4658 PlaybackThread::TimedTrack* tt = 4659 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4660 return tt->setMediaTimeTransform( 4661 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4662} 4663 4664status_t AudioFlinger::TrackHandle::onTransact( 4665 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4666{ 4667 return BnAudioTrack::onTransact(code, data, reply, flags); 4668} 4669 4670// ---------------------------------------------------------------------------- 4671 4672sp<IAudioRecord> AudioFlinger::openRecord( 4673 pid_t pid, 4674 audio_io_handle_t input, 4675 uint32_t sampleRate, 4676 audio_format_t format, 4677 uint32_t channelMask, 4678 int frameCount, 4679 // FIXME dead, remove from IAudioFlinger 4680 uint32_t flags, 4681 int *sessionId, 4682 status_t *status) 4683{ 4684 sp<RecordThread::RecordTrack> recordTrack; 4685 sp<RecordHandle> recordHandle; 4686 sp<Client> client; 4687 status_t lStatus; 4688 RecordThread *thread; 4689 size_t inFrameCount; 4690 int lSessionId; 4691 4692 // check calling permissions 4693 if (!recordingAllowed()) { 4694 lStatus = PERMISSION_DENIED; 4695 goto Exit; 4696 } 4697 4698 // add client to list 4699 { // scope for mLock 4700 Mutex::Autolock _l(mLock); 4701 thread = checkRecordThread_l(input); 4702 if (thread == NULL) { 4703 lStatus = BAD_VALUE; 4704 goto Exit; 4705 } 4706 4707 client = registerPid_l(pid); 4708 4709 // If no audio session id is provided, create one here 4710 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4711 lSessionId = *sessionId; 4712 } else { 4713 lSessionId = nextUniqueId(); 4714 if (sessionId != NULL) { 4715 *sessionId = lSessionId; 4716 } 4717 } 4718 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4719 recordTrack = thread->createRecordTrack_l(client, 4720 sampleRate, 4721 format, 4722 channelMask, 4723 frameCount, 4724 lSessionId, 4725 &lStatus); 4726 } 4727 if (lStatus != NO_ERROR) { 4728 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4729 // destructor is called by the TrackBase destructor with mLock held 4730 client.clear(); 4731 recordTrack.clear(); 4732 goto Exit; 4733 } 4734 4735 // return to handle to client 4736 recordHandle = new RecordHandle(recordTrack); 4737 lStatus = NO_ERROR; 4738 4739Exit: 4740 if (status) { 4741 *status = lStatus; 4742 } 4743 return recordHandle; 4744} 4745 4746// ---------------------------------------------------------------------------- 4747 4748AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4749 : BnAudioRecord(), 4750 mRecordTrack(recordTrack) 4751{ 4752} 4753 4754AudioFlinger::RecordHandle::~RecordHandle() { 4755 stop(); 4756} 4757 4758sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4759 return mRecordTrack->getCblk(); 4760} 4761 4762status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4763 ALOGV("RecordHandle::start()"); 4764 return mRecordTrack->start(tid); 4765} 4766 4767void AudioFlinger::RecordHandle::stop() { 4768 ALOGV("RecordHandle::stop()"); 4769 mRecordTrack->stop(); 4770} 4771 4772status_t AudioFlinger::RecordHandle::onTransact( 4773 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4774{ 4775 return BnAudioRecord::onTransact(code, data, reply, flags); 4776} 4777 4778// ---------------------------------------------------------------------------- 4779 4780AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4781 AudioStreamIn *input, 4782 uint32_t sampleRate, 4783 uint32_t channels, 4784 audio_io_handle_t id, 4785 uint32_t device) : 4786 ThreadBase(audioFlinger, id, device, RECORD), 4787 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4788 // mRsmpInIndex and mInputBytes set by readInputParameters() 4789 mReqChannelCount(popcount(channels)), 4790 mReqSampleRate(sampleRate) 4791 // mBytesRead is only meaningful while active, and so is cleared in start() 4792 // (but might be better to also clear here for dump?) 4793{ 4794 snprintf(mName, kNameLength, "AudioIn_%X", id); 4795 4796 readInputParameters(); 4797} 4798 4799 4800AudioFlinger::RecordThread::~RecordThread() 4801{ 4802 delete[] mRsmpInBuffer; 4803 delete mResampler; 4804 delete[] mRsmpOutBuffer; 4805} 4806 4807void AudioFlinger::RecordThread::onFirstRef() 4808{ 4809 run(mName, PRIORITY_URGENT_AUDIO); 4810} 4811 4812status_t AudioFlinger::RecordThread::readyToRun() 4813{ 4814 status_t status = initCheck(); 4815 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4816 return status; 4817} 4818 4819bool AudioFlinger::RecordThread::threadLoop() 4820{ 4821 AudioBufferProvider::Buffer buffer; 4822 sp<RecordTrack> activeTrack; 4823 Vector< sp<EffectChain> > effectChains; 4824 4825 nsecs_t lastWarning = 0; 4826 4827 acquireWakeLock(); 4828 4829 // start recording 4830 while (!exitPending()) { 4831 4832 processConfigEvents(); 4833 4834 { // scope for mLock 4835 Mutex::Autolock _l(mLock); 4836 checkForNewParameters_l(); 4837 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4838 if (!mStandby) { 4839 mInput->stream->common.standby(&mInput->stream->common); 4840 mStandby = true; 4841 } 4842 4843 if (exitPending()) break; 4844 4845 releaseWakeLock_l(); 4846 ALOGV("RecordThread: loop stopping"); 4847 // go to sleep 4848 mWaitWorkCV.wait(mLock); 4849 ALOGV("RecordThread: loop starting"); 4850 acquireWakeLock_l(); 4851 continue; 4852 } 4853 if (mActiveTrack != 0) { 4854 if (mActiveTrack->mState == TrackBase::PAUSING) { 4855 if (!mStandby) { 4856 mInput->stream->common.standby(&mInput->stream->common); 4857 mStandby = true; 4858 } 4859 mActiveTrack.clear(); 4860 mStartStopCond.broadcast(); 4861 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4862 if (mReqChannelCount != mActiveTrack->channelCount()) { 4863 mActiveTrack.clear(); 4864 mStartStopCond.broadcast(); 4865 } else if (mBytesRead != 0) { 4866 // record start succeeds only if first read from audio input 4867 // succeeds 4868 if (mBytesRead > 0) { 4869 mActiveTrack->mState = TrackBase::ACTIVE; 4870 } else { 4871 mActiveTrack.clear(); 4872 } 4873 mStartStopCond.broadcast(); 4874 } 4875 mStandby = false; 4876 } 4877 } 4878 lockEffectChains_l(effectChains); 4879 } 4880 4881 if (mActiveTrack != 0) { 4882 if (mActiveTrack->mState != TrackBase::ACTIVE && 4883 mActiveTrack->mState != TrackBase::RESUMING) { 4884 unlockEffectChains(effectChains); 4885 usleep(kRecordThreadSleepUs); 4886 continue; 4887 } 4888 for (size_t i = 0; i < effectChains.size(); i ++) { 4889 effectChains[i]->process_l(); 4890 } 4891 4892 buffer.frameCount = mFrameCount; 4893 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4894 size_t framesOut = buffer.frameCount; 4895 if (mResampler == NULL) { 4896 // no resampling 4897 while (framesOut) { 4898 size_t framesIn = mFrameCount - mRsmpInIndex; 4899 if (framesIn) { 4900 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4901 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4902 if (framesIn > framesOut) 4903 framesIn = framesOut; 4904 mRsmpInIndex += framesIn; 4905 framesOut -= framesIn; 4906 if ((int)mChannelCount == mReqChannelCount || 4907 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4908 memcpy(dst, src, framesIn * mFrameSize); 4909 } else { 4910 int16_t *src16 = (int16_t *)src; 4911 int16_t *dst16 = (int16_t *)dst; 4912 if (mChannelCount == 1) { 4913 while (framesIn--) { 4914 *dst16++ = *src16; 4915 *dst16++ = *src16++; 4916 } 4917 } else { 4918 while (framesIn--) { 4919 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4920 src16 += 2; 4921 } 4922 } 4923 } 4924 } 4925 if (framesOut && mFrameCount == mRsmpInIndex) { 4926 if (framesOut == mFrameCount && 4927 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4928 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4929 framesOut = 0; 4930 } else { 4931 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4932 mRsmpInIndex = 0; 4933 } 4934 if (mBytesRead < 0) { 4935 ALOGE("Error reading audio input"); 4936 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4937 // Force input into standby so that it tries to 4938 // recover at next read attempt 4939 mInput->stream->common.standby(&mInput->stream->common); 4940 usleep(kRecordThreadSleepUs); 4941 } 4942 mRsmpInIndex = mFrameCount; 4943 framesOut = 0; 4944 buffer.frameCount = 0; 4945 } 4946 } 4947 } 4948 } else { 4949 // resampling 4950 4951 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4952 // alter output frame count as if we were expecting stereo samples 4953 if (mChannelCount == 1 && mReqChannelCount == 1) { 4954 framesOut >>= 1; 4955 } 4956 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4957 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4958 // are 32 bit aligned which should be always true. 4959 if (mChannelCount == 2 && mReqChannelCount == 1) { 4960 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4961 // the resampler always outputs stereo samples: do post stereo to mono conversion 4962 int16_t *src = (int16_t *)mRsmpOutBuffer; 4963 int16_t *dst = buffer.i16; 4964 while (framesOut--) { 4965 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4966 src += 2; 4967 } 4968 } else { 4969 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4970 } 4971 4972 } 4973 mActiveTrack->releaseBuffer(&buffer); 4974 mActiveTrack->overflow(); 4975 } 4976 // client isn't retrieving buffers fast enough 4977 else { 4978 if (!mActiveTrack->setOverflow()) { 4979 nsecs_t now = systemTime(); 4980 if ((now - lastWarning) > kWarningThrottleNs) { 4981 ALOGW("RecordThread: buffer overflow"); 4982 lastWarning = now; 4983 } 4984 } 4985 // Release the processor for a while before asking for a new buffer. 4986 // This will give the application more chance to read from the buffer and 4987 // clear the overflow. 4988 usleep(kRecordThreadSleepUs); 4989 } 4990 } 4991 // enable changes in effect chain 4992 unlockEffectChains(effectChains); 4993 effectChains.clear(); 4994 } 4995 4996 if (!mStandby) { 4997 mInput->stream->common.standby(&mInput->stream->common); 4998 } 4999 mActiveTrack.clear(); 5000 5001 mStartStopCond.broadcast(); 5002 5003 releaseWakeLock(); 5004 5005 ALOGV("RecordThread %p exiting", this); 5006 return false; 5007} 5008 5009 5010sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5011 const sp<AudioFlinger::Client>& client, 5012 uint32_t sampleRate, 5013 audio_format_t format, 5014 int channelMask, 5015 int frameCount, 5016 int sessionId, 5017 status_t *status) 5018{ 5019 sp<RecordTrack> track; 5020 status_t lStatus; 5021 5022 lStatus = initCheck(); 5023 if (lStatus != NO_ERROR) { 5024 ALOGE("Audio driver not initialized."); 5025 goto Exit; 5026 } 5027 5028 { // scope for mLock 5029 Mutex::Autolock _l(mLock); 5030 5031 track = new RecordTrack(this, client, sampleRate, 5032 format, channelMask, frameCount, sessionId); 5033 5034 if (track->getCblk() == 0) { 5035 lStatus = NO_MEMORY; 5036 goto Exit; 5037 } 5038 5039 mTrack = track.get(); 5040 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5041 bool suspend = audio_is_bluetooth_sco_device( 5042 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5043 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5044 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5045 } 5046 lStatus = NO_ERROR; 5047 5048Exit: 5049 if (status) { 5050 *status = lStatus; 5051 } 5052 return track; 5053} 5054 5055status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5056{ 5057 ALOGV("RecordThread::start tid=%d", tid); 5058 sp<ThreadBase> strongMe = this; 5059 status_t status = NO_ERROR; 5060 { 5061 AutoMutex lock(mLock); 5062 if (mActiveTrack != 0) { 5063 if (recordTrack != mActiveTrack.get()) { 5064 status = -EBUSY; 5065 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5066 mActiveTrack->mState = TrackBase::ACTIVE; 5067 } 5068 return status; 5069 } 5070 5071 recordTrack->mState = TrackBase::IDLE; 5072 mActiveTrack = recordTrack; 5073 mLock.unlock(); 5074 status_t status = AudioSystem::startInput(mId); 5075 mLock.lock(); 5076 if (status != NO_ERROR) { 5077 mActiveTrack.clear(); 5078 return status; 5079 } 5080 mRsmpInIndex = mFrameCount; 5081 mBytesRead = 0; 5082 if (mResampler != NULL) { 5083 mResampler->reset(); 5084 } 5085 mActiveTrack->mState = TrackBase::RESUMING; 5086 // signal thread to start 5087 ALOGV("Signal record thread"); 5088 mWaitWorkCV.signal(); 5089 // do not wait for mStartStopCond if exiting 5090 if (exitPending()) { 5091 mActiveTrack.clear(); 5092 status = INVALID_OPERATION; 5093 goto startError; 5094 } 5095 mStartStopCond.wait(mLock); 5096 if (mActiveTrack == 0) { 5097 ALOGV("Record failed to start"); 5098 status = BAD_VALUE; 5099 goto startError; 5100 } 5101 ALOGV("Record started OK"); 5102 return status; 5103 } 5104startError: 5105 AudioSystem::stopInput(mId); 5106 return status; 5107} 5108 5109void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5110 ALOGV("RecordThread::stop"); 5111 sp<ThreadBase> strongMe = this; 5112 { 5113 AutoMutex lock(mLock); 5114 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5115 mActiveTrack->mState = TrackBase::PAUSING; 5116 // do not wait for mStartStopCond if exiting 5117 if (exitPending()) { 5118 return; 5119 } 5120 mStartStopCond.wait(mLock); 5121 // if we have been restarted, recordTrack == mActiveTrack.get() here 5122 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5123 mLock.unlock(); 5124 AudioSystem::stopInput(mId); 5125 mLock.lock(); 5126 ALOGV("Record stopped OK"); 5127 } 5128 } 5129 } 5130} 5131 5132status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5133{ 5134 const size_t SIZE = 256; 5135 char buffer[SIZE]; 5136 String8 result; 5137 5138 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5139 result.append(buffer); 5140 5141 if (mActiveTrack != 0) { 5142 result.append("Active Track:\n"); 5143 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5144 mActiveTrack->dump(buffer, SIZE); 5145 result.append(buffer); 5146 5147 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5148 result.append(buffer); 5149 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5150 result.append(buffer); 5151 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5152 result.append(buffer); 5153 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5154 result.append(buffer); 5155 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5156 result.append(buffer); 5157 5158 5159 } else { 5160 result.append("No record client\n"); 5161 } 5162 write(fd, result.string(), result.size()); 5163 5164 dumpBase(fd, args); 5165 dumpEffectChains(fd, args); 5166 5167 return NO_ERROR; 5168} 5169 5170// AudioBufferProvider interface 5171status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5172{ 5173 size_t framesReq = buffer->frameCount; 5174 size_t framesReady = mFrameCount - mRsmpInIndex; 5175 int channelCount; 5176 5177 if (framesReady == 0) { 5178 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5179 if (mBytesRead < 0) { 5180 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5181 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5182 // Force input into standby so that it tries to 5183 // recover at next read attempt 5184 mInput->stream->common.standby(&mInput->stream->common); 5185 usleep(kRecordThreadSleepUs); 5186 } 5187 buffer->raw = NULL; 5188 buffer->frameCount = 0; 5189 return NOT_ENOUGH_DATA; 5190 } 5191 mRsmpInIndex = 0; 5192 framesReady = mFrameCount; 5193 } 5194 5195 if (framesReq > framesReady) { 5196 framesReq = framesReady; 5197 } 5198 5199 if (mChannelCount == 1 && mReqChannelCount == 2) { 5200 channelCount = 1; 5201 } else { 5202 channelCount = 2; 5203 } 5204 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5205 buffer->frameCount = framesReq; 5206 return NO_ERROR; 5207} 5208 5209// AudioBufferProvider interface 5210void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5211{ 5212 mRsmpInIndex += buffer->frameCount; 5213 buffer->frameCount = 0; 5214} 5215 5216bool AudioFlinger::RecordThread::checkForNewParameters_l() 5217{ 5218 bool reconfig = false; 5219 5220 while (!mNewParameters.isEmpty()) { 5221 status_t status = NO_ERROR; 5222 String8 keyValuePair = mNewParameters[0]; 5223 AudioParameter param = AudioParameter(keyValuePair); 5224 int value; 5225 audio_format_t reqFormat = mFormat; 5226 int reqSamplingRate = mReqSampleRate; 5227 int reqChannelCount = mReqChannelCount; 5228 5229 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5230 reqSamplingRate = value; 5231 reconfig = true; 5232 } 5233 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5234 reqFormat = (audio_format_t) value; 5235 reconfig = true; 5236 } 5237 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5238 reqChannelCount = popcount(value); 5239 reconfig = true; 5240 } 5241 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5242 // do not accept frame count changes if tracks are open as the track buffer 5243 // size depends on frame count and correct behavior would not be guaranteed 5244 // if frame count is changed after track creation 5245 if (mActiveTrack != 0) { 5246 status = INVALID_OPERATION; 5247 } else { 5248 reconfig = true; 5249 } 5250 } 5251 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5252 // forward device change to effects that have requested to be 5253 // aware of attached audio device. 5254 for (size_t i = 0; i < mEffectChains.size(); i++) { 5255 mEffectChains[i]->setDevice_l(value); 5256 } 5257 // store input device and output device but do not forward output device to audio HAL. 5258 // Note that status is ignored by the caller for output device 5259 // (see AudioFlinger::setParameters() 5260 if (value & AUDIO_DEVICE_OUT_ALL) { 5261 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5262 status = BAD_VALUE; 5263 } else { 5264 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5265 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5266 if (mTrack != NULL) { 5267 bool suspend = audio_is_bluetooth_sco_device( 5268 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5269 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5270 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5271 } 5272 } 5273 mDevice |= (uint32_t)value; 5274 } 5275 if (status == NO_ERROR) { 5276 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5277 if (status == INVALID_OPERATION) { 5278 mInput->stream->common.standby(&mInput->stream->common); 5279 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5280 keyValuePair.string()); 5281 } 5282 if (reconfig) { 5283 if (status == BAD_VALUE && 5284 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5285 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5286 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5287 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5288 (reqChannelCount <= FCC_2)) { 5289 status = NO_ERROR; 5290 } 5291 if (status == NO_ERROR) { 5292 readInputParameters(); 5293 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5294 } 5295 } 5296 } 5297 5298 mNewParameters.removeAt(0); 5299 5300 mParamStatus = status; 5301 mParamCond.signal(); 5302 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5303 // already timed out waiting for the status and will never signal the condition. 5304 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5305 } 5306 return reconfig; 5307} 5308 5309String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5310{ 5311 char *s; 5312 String8 out_s8 = String8(); 5313 5314 Mutex::Autolock _l(mLock); 5315 if (initCheck() != NO_ERROR) { 5316 return out_s8; 5317 } 5318 5319 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5320 out_s8 = String8(s); 5321 free(s); 5322 return out_s8; 5323} 5324 5325void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5326 AudioSystem::OutputDescriptor desc; 5327 void *param2 = NULL; 5328 5329 switch (event) { 5330 case AudioSystem::INPUT_OPENED: 5331 case AudioSystem::INPUT_CONFIG_CHANGED: 5332 desc.channels = mChannelMask; 5333 desc.samplingRate = mSampleRate; 5334 desc.format = mFormat; 5335 desc.frameCount = mFrameCount; 5336 desc.latency = 0; 5337 param2 = &desc; 5338 break; 5339 5340 case AudioSystem::INPUT_CLOSED: 5341 default: 5342 break; 5343 } 5344 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5345} 5346 5347void AudioFlinger::RecordThread::readInputParameters() 5348{ 5349 delete mRsmpInBuffer; 5350 // mRsmpInBuffer is always assigned a new[] below 5351 delete mRsmpOutBuffer; 5352 mRsmpOutBuffer = NULL; 5353 delete mResampler; 5354 mResampler = NULL; 5355 5356 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5357 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5358 mChannelCount = (uint16_t)popcount(mChannelMask); 5359 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5360 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5361 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5362 mFrameCount = mInputBytes / mFrameSize; 5363 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5364 5365 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5366 { 5367 int channelCount; 5368 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5369 // stereo to mono post process as the resampler always outputs stereo. 5370 if (mChannelCount == 1 && mReqChannelCount == 2) { 5371 channelCount = 1; 5372 } else { 5373 channelCount = 2; 5374 } 5375 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5376 mResampler->setSampleRate(mSampleRate); 5377 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5378 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5379 5380 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5381 if (mChannelCount == 1 && mReqChannelCount == 1) { 5382 mFrameCount >>= 1; 5383 } 5384 5385 } 5386 mRsmpInIndex = mFrameCount; 5387} 5388 5389unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5390{ 5391 Mutex::Autolock _l(mLock); 5392 if (initCheck() != NO_ERROR) { 5393 return 0; 5394 } 5395 5396 return mInput->stream->get_input_frames_lost(mInput->stream); 5397} 5398 5399uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5400{ 5401 Mutex::Autolock _l(mLock); 5402 uint32_t result = 0; 5403 if (getEffectChain_l(sessionId) != 0) { 5404 result = EFFECT_SESSION; 5405 } 5406 5407 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5408 result |= TRACK_SESSION; 5409 } 5410 5411 return result; 5412} 5413 5414AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5415{ 5416 Mutex::Autolock _l(mLock); 5417 return mTrack; 5418} 5419 5420AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5421{ 5422 Mutex::Autolock _l(mLock); 5423 return mInput; 5424} 5425 5426AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5427{ 5428 Mutex::Autolock _l(mLock); 5429 AudioStreamIn *input = mInput; 5430 mInput = NULL; 5431 return input; 5432} 5433 5434// this method must always be called either with ThreadBase mLock held or inside the thread loop 5435audio_stream_t* AudioFlinger::RecordThread::stream() 5436{ 5437 if (mInput == NULL) { 5438 return NULL; 5439 } 5440 return &mInput->stream->common; 5441} 5442 5443 5444// ---------------------------------------------------------------------------- 5445 5446audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5447 uint32_t *pSamplingRate, 5448 audio_format_t *pFormat, 5449 uint32_t *pChannels, 5450 uint32_t *pLatencyMs, 5451 audio_policy_output_flags_t flags) 5452{ 5453 status_t status; 5454 PlaybackThread *thread = NULL; 5455 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5456 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5457 uint32_t channels = pChannels ? *pChannels : 0; 5458 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5459 audio_stream_out_t *outStream; 5460 audio_hw_device_t *outHwDev; 5461 5462 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5463 pDevices ? *pDevices : 0, 5464 samplingRate, 5465 format, 5466 channels, 5467 flags); 5468 5469 if (pDevices == NULL || *pDevices == 0) { 5470 return 0; 5471 } 5472 5473 Mutex::Autolock _l(mLock); 5474 5475 outHwDev = findSuitableHwDev_l(*pDevices); 5476 if (outHwDev == NULL) 5477 return 0; 5478 5479 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5480 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5481 &channels, &samplingRate, &outStream); 5482 mHardwareStatus = AUDIO_HW_IDLE; 5483 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5484 outStream, 5485 samplingRate, 5486 format, 5487 channels, 5488 status); 5489 5490 if (outStream != NULL) { 5491 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5492 audio_io_handle_t id = nextUniqueId(); 5493 5494 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5495 (format != AUDIO_FORMAT_PCM_16_BIT) || 5496 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5497 thread = new DirectOutputThread(this, output, id, *pDevices); 5498 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5499 } else { 5500 thread = new MixerThread(this, output, id, *pDevices); 5501 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5502 } 5503 mPlaybackThreads.add(id, thread); 5504 5505 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5506 if (pFormat != NULL) *pFormat = format; 5507 if (pChannels != NULL) *pChannels = channels; 5508 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5509 5510 // notify client processes of the new output creation 5511 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5512 return id; 5513 } 5514 5515 return 0; 5516} 5517 5518audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5519 audio_io_handle_t output2) 5520{ 5521 Mutex::Autolock _l(mLock); 5522 MixerThread *thread1 = checkMixerThread_l(output1); 5523 MixerThread *thread2 = checkMixerThread_l(output2); 5524 5525 if (thread1 == NULL || thread2 == NULL) { 5526 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5527 return 0; 5528 } 5529 5530 audio_io_handle_t id = nextUniqueId(); 5531 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5532 thread->addOutputTrack(thread2); 5533 mPlaybackThreads.add(id, thread); 5534 // notify client processes of the new output creation 5535 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5536 return id; 5537} 5538 5539status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5540{ 5541 // keep strong reference on the playback thread so that 5542 // it is not destroyed while exit() is executed 5543 sp<PlaybackThread> thread; 5544 { 5545 Mutex::Autolock _l(mLock); 5546 thread = checkPlaybackThread_l(output); 5547 if (thread == NULL) { 5548 return BAD_VALUE; 5549 } 5550 5551 ALOGV("closeOutput() %d", output); 5552 5553 if (thread->type() == ThreadBase::MIXER) { 5554 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5555 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5556 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5557 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5558 } 5559 } 5560 } 5561 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5562 mPlaybackThreads.removeItem(output); 5563 } 5564 thread->exit(); 5565 // The thread entity (active unit of execution) is no longer running here, 5566 // but the ThreadBase container still exists. 5567 5568 if (thread->type() != ThreadBase::DUPLICATING) { 5569 AudioStreamOut *out = thread->clearOutput(); 5570 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5571 // from now on thread->mOutput is NULL 5572 out->hwDev->close_output_stream(out->hwDev, out->stream); 5573 delete out; 5574 } 5575 return NO_ERROR; 5576} 5577 5578status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5579{ 5580 Mutex::Autolock _l(mLock); 5581 PlaybackThread *thread = checkPlaybackThread_l(output); 5582 5583 if (thread == NULL) { 5584 return BAD_VALUE; 5585 } 5586 5587 ALOGV("suspendOutput() %d", output); 5588 thread->suspend(); 5589 5590 return NO_ERROR; 5591} 5592 5593status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5594{ 5595 Mutex::Autolock _l(mLock); 5596 PlaybackThread *thread = checkPlaybackThread_l(output); 5597 5598 if (thread == NULL) { 5599 return BAD_VALUE; 5600 } 5601 5602 ALOGV("restoreOutput() %d", output); 5603 5604 thread->restore(); 5605 5606 return NO_ERROR; 5607} 5608 5609audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5610 uint32_t *pSamplingRate, 5611 audio_format_t *pFormat, 5612 uint32_t *pChannels, 5613 audio_in_acoustics_t acoustics) 5614{ 5615 status_t status; 5616 RecordThread *thread = NULL; 5617 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5618 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5619 uint32_t channels = pChannels ? *pChannels : 0; 5620 uint32_t reqSamplingRate = samplingRate; 5621 audio_format_t reqFormat = format; 5622 uint32_t reqChannels = channels; 5623 audio_stream_in_t *inStream; 5624 audio_hw_device_t *inHwDev; 5625 5626 if (pDevices == NULL || *pDevices == 0) { 5627 return 0; 5628 } 5629 5630 Mutex::Autolock _l(mLock); 5631 5632 inHwDev = findSuitableHwDev_l(*pDevices); 5633 if (inHwDev == NULL) 5634 return 0; 5635 5636 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5637 &channels, &samplingRate, 5638 acoustics, 5639 &inStream); 5640 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5641 inStream, 5642 samplingRate, 5643 format, 5644 channels, 5645 acoustics, 5646 status); 5647 5648 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5649 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5650 // or stereo to mono conversions on 16 bit PCM inputs. 5651 if (inStream == NULL && status == BAD_VALUE && 5652 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5653 (samplingRate <= 2 * reqSamplingRate) && 5654 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5655 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5656 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5657 &channels, &samplingRate, 5658 acoustics, 5659 &inStream); 5660 } 5661 5662 if (inStream != NULL) { 5663 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5664 5665 audio_io_handle_t id = nextUniqueId(); 5666 // Start record thread 5667 // RecorThread require both input and output device indication to forward to audio 5668 // pre processing modules 5669 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5670 thread = new RecordThread(this, 5671 input, 5672 reqSamplingRate, 5673 reqChannels, 5674 id, 5675 device); 5676 mRecordThreads.add(id, thread); 5677 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5678 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5679 if (pFormat != NULL) *pFormat = format; 5680 if (pChannels != NULL) *pChannels = reqChannels; 5681 5682 input->stream->common.standby(&input->stream->common); 5683 5684 // notify client processes of the new input creation 5685 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5686 return id; 5687 } 5688 5689 return 0; 5690} 5691 5692status_t AudioFlinger::closeInput(audio_io_handle_t input) 5693{ 5694 // keep strong reference on the record thread so that 5695 // it is not destroyed while exit() is executed 5696 sp<RecordThread> thread; 5697 { 5698 Mutex::Autolock _l(mLock); 5699 thread = checkRecordThread_l(input); 5700 if (thread == NULL) { 5701 return BAD_VALUE; 5702 } 5703 5704 ALOGV("closeInput() %d", input); 5705 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5706 mRecordThreads.removeItem(input); 5707 } 5708 thread->exit(); 5709 // The thread entity (active unit of execution) is no longer running here, 5710 // but the ThreadBase container still exists. 5711 5712 AudioStreamIn *in = thread->clearInput(); 5713 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5714 // from now on thread->mInput is NULL 5715 in->hwDev->close_input_stream(in->hwDev, in->stream); 5716 delete in; 5717 5718 return NO_ERROR; 5719} 5720 5721status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5722{ 5723 Mutex::Autolock _l(mLock); 5724 MixerThread *dstThread = checkMixerThread_l(output); 5725 if (dstThread == NULL) { 5726 ALOGW("setStreamOutput() bad output id %d", output); 5727 return BAD_VALUE; 5728 } 5729 5730 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5731 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5732 5733 dstThread->setStreamValid(stream, true); 5734 5735 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5736 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5737 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5738 MixerThread *srcThread = (MixerThread *)thread; 5739 srcThread->setStreamValid(stream, false); 5740 srcThread->invalidateTracks(stream); 5741 } 5742 } 5743 5744 return NO_ERROR; 5745} 5746 5747 5748int AudioFlinger::newAudioSessionId() 5749{ 5750 return nextUniqueId(); 5751} 5752 5753void AudioFlinger::acquireAudioSessionId(int audioSession) 5754{ 5755 Mutex::Autolock _l(mLock); 5756 pid_t caller = IPCThreadState::self()->getCallingPid(); 5757 ALOGV("acquiring %d from %d", audioSession, caller); 5758 size_t num = mAudioSessionRefs.size(); 5759 for (size_t i = 0; i< num; i++) { 5760 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5761 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5762 ref->mCnt++; 5763 ALOGV(" incremented refcount to %d", ref->mCnt); 5764 return; 5765 } 5766 } 5767 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5768 ALOGV(" added new entry for %d", audioSession); 5769} 5770 5771void AudioFlinger::releaseAudioSessionId(int audioSession) 5772{ 5773 Mutex::Autolock _l(mLock); 5774 pid_t caller = IPCThreadState::self()->getCallingPid(); 5775 ALOGV("releasing %d from %d", audioSession, caller); 5776 size_t num = mAudioSessionRefs.size(); 5777 for (size_t i = 0; i< num; i++) { 5778 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5779 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5780 ref->mCnt--; 5781 ALOGV(" decremented refcount to %d", ref->mCnt); 5782 if (ref->mCnt == 0) { 5783 mAudioSessionRefs.removeAt(i); 5784 delete ref; 5785 purgeStaleEffects_l(); 5786 } 5787 return; 5788 } 5789 } 5790 ALOGW("session id %d not found for pid %d", audioSession, caller); 5791} 5792 5793void AudioFlinger::purgeStaleEffects_l() { 5794 5795 ALOGV("purging stale effects"); 5796 5797 Vector< sp<EffectChain> > chains; 5798 5799 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5800 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5801 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5802 sp<EffectChain> ec = t->mEffectChains[j]; 5803 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5804 chains.push(ec); 5805 } 5806 } 5807 } 5808 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5809 sp<RecordThread> t = mRecordThreads.valueAt(i); 5810 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5811 sp<EffectChain> ec = t->mEffectChains[j]; 5812 chains.push(ec); 5813 } 5814 } 5815 5816 for (size_t i = 0; i < chains.size(); i++) { 5817 sp<EffectChain> ec = chains[i]; 5818 int sessionid = ec->sessionId(); 5819 sp<ThreadBase> t = ec->mThread.promote(); 5820 if (t == 0) { 5821 continue; 5822 } 5823 size_t numsessionrefs = mAudioSessionRefs.size(); 5824 bool found = false; 5825 for (size_t k = 0; k < numsessionrefs; k++) { 5826 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5827 if (ref->mSessionid == sessionid) { 5828 ALOGV(" session %d still exists for %d with %d refs", 5829 sessionid, ref->mPid, ref->mCnt); 5830 found = true; 5831 break; 5832 } 5833 } 5834 if (!found) { 5835 // remove all effects from the chain 5836 while (ec->mEffects.size()) { 5837 sp<EffectModule> effect = ec->mEffects[0]; 5838 effect->unPin(); 5839 Mutex::Autolock _l (t->mLock); 5840 t->removeEffect_l(effect); 5841 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5842 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5843 if (handle != 0) { 5844 handle->mEffect.clear(); 5845 if (handle->mHasControl && handle->mEnabled) { 5846 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5847 } 5848 } 5849 } 5850 AudioSystem::unregisterEffect(effect->id()); 5851 } 5852 } 5853 } 5854 return; 5855} 5856 5857// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5858AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5859{ 5860 return mPlaybackThreads.valueFor(output).get(); 5861} 5862 5863// checkMixerThread_l() must be called with AudioFlinger::mLock held 5864AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5865{ 5866 PlaybackThread *thread = checkPlaybackThread_l(output); 5867 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5868} 5869 5870// checkRecordThread_l() must be called with AudioFlinger::mLock held 5871AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5872{ 5873 return mRecordThreads.valueFor(input).get(); 5874} 5875 5876uint32_t AudioFlinger::nextUniqueId() 5877{ 5878 return android_atomic_inc(&mNextUniqueId); 5879} 5880 5881AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5882{ 5883 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5884 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5885 AudioStreamOut *output = thread->getOutput(); 5886 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5887 return thread; 5888 } 5889 } 5890 return NULL; 5891} 5892 5893uint32_t AudioFlinger::primaryOutputDevice_l() const 5894{ 5895 PlaybackThread *thread = primaryPlaybackThread_l(); 5896 5897 if (thread == NULL) { 5898 return 0; 5899 } 5900 5901 return thread->device(); 5902} 5903 5904 5905// ---------------------------------------------------------------------------- 5906// Effect management 5907// ---------------------------------------------------------------------------- 5908 5909 5910status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5911{ 5912 Mutex::Autolock _l(mLock); 5913 return EffectQueryNumberEffects(numEffects); 5914} 5915 5916status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5917{ 5918 Mutex::Autolock _l(mLock); 5919 return EffectQueryEffect(index, descriptor); 5920} 5921 5922status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5923 effect_descriptor_t *descriptor) const 5924{ 5925 Mutex::Autolock _l(mLock); 5926 return EffectGetDescriptor(pUuid, descriptor); 5927} 5928 5929 5930sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5931 effect_descriptor_t *pDesc, 5932 const sp<IEffectClient>& effectClient, 5933 int32_t priority, 5934 audio_io_handle_t io, 5935 int sessionId, 5936 status_t *status, 5937 int *id, 5938 int *enabled) 5939{ 5940 status_t lStatus = NO_ERROR; 5941 sp<EffectHandle> handle; 5942 effect_descriptor_t desc; 5943 5944 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5945 pid, effectClient.get(), priority, sessionId, io); 5946 5947 if (pDesc == NULL) { 5948 lStatus = BAD_VALUE; 5949 goto Exit; 5950 } 5951 5952 // check audio settings permission for global effects 5953 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5954 lStatus = PERMISSION_DENIED; 5955 goto Exit; 5956 } 5957 5958 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5959 // that can only be created by audio policy manager (running in same process) 5960 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5961 lStatus = PERMISSION_DENIED; 5962 goto Exit; 5963 } 5964 5965 if (io == 0) { 5966 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5967 // output must be specified by AudioPolicyManager when using session 5968 // AUDIO_SESSION_OUTPUT_STAGE 5969 lStatus = BAD_VALUE; 5970 goto Exit; 5971 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5972 // if the output returned by getOutputForEffect() is removed before we lock the 5973 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5974 // and we will exit safely 5975 io = AudioSystem::getOutputForEffect(&desc); 5976 } 5977 } 5978 5979 { 5980 Mutex::Autolock _l(mLock); 5981 5982 5983 if (!EffectIsNullUuid(&pDesc->uuid)) { 5984 // if uuid is specified, request effect descriptor 5985 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5986 if (lStatus < 0) { 5987 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5988 goto Exit; 5989 } 5990 } else { 5991 // if uuid is not specified, look for an available implementation 5992 // of the required type in effect factory 5993 if (EffectIsNullUuid(&pDesc->type)) { 5994 ALOGW("createEffect() no effect type"); 5995 lStatus = BAD_VALUE; 5996 goto Exit; 5997 } 5998 uint32_t numEffects = 0; 5999 effect_descriptor_t d; 6000 d.flags = 0; // prevent compiler warning 6001 bool found = false; 6002 6003 lStatus = EffectQueryNumberEffects(&numEffects); 6004 if (lStatus < 0) { 6005 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6006 goto Exit; 6007 } 6008 for (uint32_t i = 0; i < numEffects; i++) { 6009 lStatus = EffectQueryEffect(i, &desc); 6010 if (lStatus < 0) { 6011 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6012 continue; 6013 } 6014 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6015 // If matching type found save effect descriptor. If the session is 6016 // 0 and the effect is not auxiliary, continue enumeration in case 6017 // an auxiliary version of this effect type is available 6018 found = true; 6019 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6020 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6021 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6022 break; 6023 } 6024 } 6025 } 6026 if (!found) { 6027 lStatus = BAD_VALUE; 6028 ALOGW("createEffect() effect not found"); 6029 goto Exit; 6030 } 6031 // For same effect type, chose auxiliary version over insert version if 6032 // connect to output mix (Compliance to OpenSL ES) 6033 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6034 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6035 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6036 } 6037 } 6038 6039 // Do not allow auxiliary effects on a session different from 0 (output mix) 6040 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6041 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6042 lStatus = INVALID_OPERATION; 6043 goto Exit; 6044 } 6045 6046 // check recording permission for visualizer 6047 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6048 !recordingAllowed()) { 6049 lStatus = PERMISSION_DENIED; 6050 goto Exit; 6051 } 6052 6053 // return effect descriptor 6054 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6055 6056 // If output is not specified try to find a matching audio session ID in one of the 6057 // output threads. 6058 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6059 // because of code checking output when entering the function. 6060 // Note: io is never 0 when creating an effect on an input 6061 if (io == 0) { 6062 // look for the thread where the specified audio session is present 6063 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6064 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6065 io = mPlaybackThreads.keyAt(i); 6066 break; 6067 } 6068 } 6069 if (io == 0) { 6070 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6071 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6072 io = mRecordThreads.keyAt(i); 6073 break; 6074 } 6075 } 6076 } 6077 // If no output thread contains the requested session ID, default to 6078 // first output. The effect chain will be moved to the correct output 6079 // thread when a track with the same session ID is created 6080 if (io == 0 && mPlaybackThreads.size()) { 6081 io = mPlaybackThreads.keyAt(0); 6082 } 6083 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6084 } 6085 ThreadBase *thread = checkRecordThread_l(io); 6086 if (thread == NULL) { 6087 thread = checkPlaybackThread_l(io); 6088 if (thread == NULL) { 6089 ALOGE("createEffect() unknown output thread"); 6090 lStatus = BAD_VALUE; 6091 goto Exit; 6092 } 6093 } 6094 6095 sp<Client> client = registerPid_l(pid); 6096 6097 // create effect on selected output thread 6098 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6099 &desc, enabled, &lStatus); 6100 if (handle != 0 && id != NULL) { 6101 *id = handle->id(); 6102 } 6103 } 6104 6105Exit: 6106 if (status != NULL) { 6107 *status = lStatus; 6108 } 6109 return handle; 6110} 6111 6112status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6113 audio_io_handle_t dstOutput) 6114{ 6115 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6116 sessionId, srcOutput, dstOutput); 6117 Mutex::Autolock _l(mLock); 6118 if (srcOutput == dstOutput) { 6119 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6120 return NO_ERROR; 6121 } 6122 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6123 if (srcThread == NULL) { 6124 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6125 return BAD_VALUE; 6126 } 6127 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6128 if (dstThread == NULL) { 6129 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6130 return BAD_VALUE; 6131 } 6132 6133 Mutex::Autolock _dl(dstThread->mLock); 6134 Mutex::Autolock _sl(srcThread->mLock); 6135 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6136 6137 return NO_ERROR; 6138} 6139 6140// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6141status_t AudioFlinger::moveEffectChain_l(int sessionId, 6142 AudioFlinger::PlaybackThread *srcThread, 6143 AudioFlinger::PlaybackThread *dstThread, 6144 bool reRegister) 6145{ 6146 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6147 sessionId, srcThread, dstThread); 6148 6149 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6150 if (chain == 0) { 6151 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6152 sessionId, srcThread); 6153 return INVALID_OPERATION; 6154 } 6155 6156 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6157 // so that a new chain is created with correct parameters when first effect is added. This is 6158 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6159 // removed. 6160 srcThread->removeEffectChain_l(chain); 6161 6162 // transfer all effects one by one so that new effect chain is created on new thread with 6163 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6164 audio_io_handle_t dstOutput = dstThread->id(); 6165 sp<EffectChain> dstChain; 6166 uint32_t strategy = 0; // prevent compiler warning 6167 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6168 while (effect != 0) { 6169 srcThread->removeEffect_l(effect); 6170 dstThread->addEffect_l(effect); 6171 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6172 if (effect->state() == EffectModule::ACTIVE || 6173 effect->state() == EffectModule::STOPPING) { 6174 effect->start(); 6175 } 6176 // if the move request is not received from audio policy manager, the effect must be 6177 // re-registered with the new strategy and output 6178 if (dstChain == 0) { 6179 dstChain = effect->chain().promote(); 6180 if (dstChain == 0) { 6181 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6182 srcThread->addEffect_l(effect); 6183 return NO_INIT; 6184 } 6185 strategy = dstChain->strategy(); 6186 } 6187 if (reRegister) { 6188 AudioSystem::unregisterEffect(effect->id()); 6189 AudioSystem::registerEffect(&effect->desc(), 6190 dstOutput, 6191 strategy, 6192 sessionId, 6193 effect->id()); 6194 } 6195 effect = chain->getEffectFromId_l(0); 6196 } 6197 6198 return NO_ERROR; 6199} 6200 6201 6202// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6203sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6204 const sp<AudioFlinger::Client>& client, 6205 const sp<IEffectClient>& effectClient, 6206 int32_t priority, 6207 int sessionId, 6208 effect_descriptor_t *desc, 6209 int *enabled, 6210 status_t *status 6211 ) 6212{ 6213 sp<EffectModule> effect; 6214 sp<EffectHandle> handle; 6215 status_t lStatus; 6216 sp<EffectChain> chain; 6217 bool chainCreated = false; 6218 bool effectCreated = false; 6219 bool effectRegistered = false; 6220 6221 lStatus = initCheck(); 6222 if (lStatus != NO_ERROR) { 6223 ALOGW("createEffect_l() Audio driver not initialized."); 6224 goto Exit; 6225 } 6226 6227 // Do not allow effects with session ID 0 on direct output or duplicating threads 6228 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6229 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6230 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6231 desc->name, sessionId); 6232 lStatus = BAD_VALUE; 6233 goto Exit; 6234 } 6235 // Only Pre processor effects are allowed on input threads and only on input threads 6236 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6237 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6238 desc->name, desc->flags, mType); 6239 lStatus = BAD_VALUE; 6240 goto Exit; 6241 } 6242 6243 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6244 6245 { // scope for mLock 6246 Mutex::Autolock _l(mLock); 6247 6248 // check for existing effect chain with the requested audio session 6249 chain = getEffectChain_l(sessionId); 6250 if (chain == 0) { 6251 // create a new chain for this session 6252 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6253 chain = new EffectChain(this, sessionId); 6254 addEffectChain_l(chain); 6255 chain->setStrategy(getStrategyForSession_l(sessionId)); 6256 chainCreated = true; 6257 } else { 6258 effect = chain->getEffectFromDesc_l(desc); 6259 } 6260 6261 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6262 6263 if (effect == 0) { 6264 int id = mAudioFlinger->nextUniqueId(); 6265 // Check CPU and memory usage 6266 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6267 if (lStatus != NO_ERROR) { 6268 goto Exit; 6269 } 6270 effectRegistered = true; 6271 // create a new effect module if none present in the chain 6272 effect = new EffectModule(this, chain, desc, id, sessionId); 6273 lStatus = effect->status(); 6274 if (lStatus != NO_ERROR) { 6275 goto Exit; 6276 } 6277 lStatus = chain->addEffect_l(effect); 6278 if (lStatus != NO_ERROR) { 6279 goto Exit; 6280 } 6281 effectCreated = true; 6282 6283 effect->setDevice(mDevice); 6284 effect->setMode(mAudioFlinger->getMode()); 6285 } 6286 // create effect handle and connect it to effect module 6287 handle = new EffectHandle(effect, client, effectClient, priority); 6288 lStatus = effect->addHandle(handle); 6289 if (enabled != NULL) { 6290 *enabled = (int)effect->isEnabled(); 6291 } 6292 } 6293 6294Exit: 6295 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6296 Mutex::Autolock _l(mLock); 6297 if (effectCreated) { 6298 chain->removeEffect_l(effect); 6299 } 6300 if (effectRegistered) { 6301 AudioSystem::unregisterEffect(effect->id()); 6302 } 6303 if (chainCreated) { 6304 removeEffectChain_l(chain); 6305 } 6306 handle.clear(); 6307 } 6308 6309 if (status != NULL) { 6310 *status = lStatus; 6311 } 6312 return handle; 6313} 6314 6315sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6316{ 6317 sp<EffectChain> chain = getEffectChain_l(sessionId); 6318 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6319} 6320 6321// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6322// PlaybackThread::mLock held 6323status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6324{ 6325 // check for existing effect chain with the requested audio session 6326 int sessionId = effect->sessionId(); 6327 sp<EffectChain> chain = getEffectChain_l(sessionId); 6328 bool chainCreated = false; 6329 6330 if (chain == 0) { 6331 // create a new chain for this session 6332 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6333 chain = new EffectChain(this, sessionId); 6334 addEffectChain_l(chain); 6335 chain->setStrategy(getStrategyForSession_l(sessionId)); 6336 chainCreated = true; 6337 } 6338 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6339 6340 if (chain->getEffectFromId_l(effect->id()) != 0) { 6341 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6342 this, effect->desc().name, chain.get()); 6343 return BAD_VALUE; 6344 } 6345 6346 status_t status = chain->addEffect_l(effect); 6347 if (status != NO_ERROR) { 6348 if (chainCreated) { 6349 removeEffectChain_l(chain); 6350 } 6351 return status; 6352 } 6353 6354 effect->setDevice(mDevice); 6355 effect->setMode(mAudioFlinger->getMode()); 6356 return NO_ERROR; 6357} 6358 6359void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6360 6361 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6362 effect_descriptor_t desc = effect->desc(); 6363 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6364 detachAuxEffect_l(effect->id()); 6365 } 6366 6367 sp<EffectChain> chain = effect->chain().promote(); 6368 if (chain != 0) { 6369 // remove effect chain if removing last effect 6370 if (chain->removeEffect_l(effect) == 0) { 6371 removeEffectChain_l(chain); 6372 } 6373 } else { 6374 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6375 } 6376} 6377 6378void AudioFlinger::ThreadBase::lockEffectChains_l( 6379 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6380{ 6381 effectChains = mEffectChains; 6382 for (size_t i = 0; i < mEffectChains.size(); i++) { 6383 mEffectChains[i]->lock(); 6384 } 6385} 6386 6387void AudioFlinger::ThreadBase::unlockEffectChains( 6388 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6389{ 6390 for (size_t i = 0; i < effectChains.size(); i++) { 6391 effectChains[i]->unlock(); 6392 } 6393} 6394 6395sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6396{ 6397 Mutex::Autolock _l(mLock); 6398 return getEffectChain_l(sessionId); 6399} 6400 6401sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6402{ 6403 size_t size = mEffectChains.size(); 6404 for (size_t i = 0; i < size; i++) { 6405 if (mEffectChains[i]->sessionId() == sessionId) { 6406 return mEffectChains[i]; 6407 } 6408 } 6409 return 0; 6410} 6411 6412void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6413{ 6414 Mutex::Autolock _l(mLock); 6415 size_t size = mEffectChains.size(); 6416 for (size_t i = 0; i < size; i++) { 6417 mEffectChains[i]->setMode_l(mode); 6418 } 6419} 6420 6421void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6422 const wp<EffectHandle>& handle, 6423 bool unpinIfLast) { 6424 6425 Mutex::Autolock _l(mLock); 6426 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6427 // delete the effect module if removing last handle on it 6428 if (effect->removeHandle(handle) == 0) { 6429 if (!effect->isPinned() || unpinIfLast) { 6430 removeEffect_l(effect); 6431 AudioSystem::unregisterEffect(effect->id()); 6432 } 6433 } 6434} 6435 6436status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6437{ 6438 int session = chain->sessionId(); 6439 int16_t *buffer = mMixBuffer; 6440 bool ownsBuffer = false; 6441 6442 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6443 if (session > 0) { 6444 // Only one effect chain can be present in direct output thread and it uses 6445 // the mix buffer as input 6446 if (mType != DIRECT) { 6447 size_t numSamples = mFrameCount * mChannelCount; 6448 buffer = new int16_t[numSamples]; 6449 memset(buffer, 0, numSamples * sizeof(int16_t)); 6450 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6451 ownsBuffer = true; 6452 } 6453 6454 // Attach all tracks with same session ID to this chain. 6455 for (size_t i = 0; i < mTracks.size(); ++i) { 6456 sp<Track> track = mTracks[i]; 6457 if (session == track->sessionId()) { 6458 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6459 track->setMainBuffer(buffer); 6460 chain->incTrackCnt(); 6461 } 6462 } 6463 6464 // indicate all active tracks in the chain 6465 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6466 sp<Track> track = mActiveTracks[i].promote(); 6467 if (track == 0) continue; 6468 if (session == track->sessionId()) { 6469 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6470 chain->incActiveTrackCnt(); 6471 } 6472 } 6473 } 6474 6475 chain->setInBuffer(buffer, ownsBuffer); 6476 chain->setOutBuffer(mMixBuffer); 6477 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6478 // chains list in order to be processed last as it contains output stage effects 6479 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6480 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6481 // after track specific effects and before output stage 6482 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6483 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6484 // Effect chain for other sessions are inserted at beginning of effect 6485 // chains list to be processed before output mix effects. Relative order between other 6486 // sessions is not important 6487 size_t size = mEffectChains.size(); 6488 size_t i = 0; 6489 for (i = 0; i < size; i++) { 6490 if (mEffectChains[i]->sessionId() < session) break; 6491 } 6492 mEffectChains.insertAt(chain, i); 6493 checkSuspendOnAddEffectChain_l(chain); 6494 6495 return NO_ERROR; 6496} 6497 6498size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6499{ 6500 int session = chain->sessionId(); 6501 6502 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6503 6504 for (size_t i = 0; i < mEffectChains.size(); i++) { 6505 if (chain == mEffectChains[i]) { 6506 mEffectChains.removeAt(i); 6507 // detach all active tracks from the chain 6508 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6509 sp<Track> track = mActiveTracks[i].promote(); 6510 if (track == 0) continue; 6511 if (session == track->sessionId()) { 6512 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6513 chain.get(), session); 6514 chain->decActiveTrackCnt(); 6515 } 6516 } 6517 6518 // detach all tracks with same session ID from this chain 6519 for (size_t i = 0; i < mTracks.size(); ++i) { 6520 sp<Track> track = mTracks[i]; 6521 if (session == track->sessionId()) { 6522 track->setMainBuffer(mMixBuffer); 6523 chain->decTrackCnt(); 6524 } 6525 } 6526 break; 6527 } 6528 } 6529 return mEffectChains.size(); 6530} 6531 6532status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6533 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6534{ 6535 Mutex::Autolock _l(mLock); 6536 return attachAuxEffect_l(track, EffectId); 6537} 6538 6539status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6540 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6541{ 6542 status_t status = NO_ERROR; 6543 6544 if (EffectId == 0) { 6545 track->setAuxBuffer(0, NULL); 6546 } else { 6547 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6548 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6549 if (effect != 0) { 6550 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6551 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6552 } else { 6553 status = INVALID_OPERATION; 6554 } 6555 } else { 6556 status = BAD_VALUE; 6557 } 6558 } 6559 return status; 6560} 6561 6562void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6563{ 6564 for (size_t i = 0; i < mTracks.size(); ++i) { 6565 sp<Track> track = mTracks[i]; 6566 if (track->auxEffectId() == effectId) { 6567 attachAuxEffect_l(track, 0); 6568 } 6569 } 6570} 6571 6572status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6573{ 6574 // only one chain per input thread 6575 if (mEffectChains.size() != 0) { 6576 return INVALID_OPERATION; 6577 } 6578 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6579 6580 chain->setInBuffer(NULL); 6581 chain->setOutBuffer(NULL); 6582 6583 checkSuspendOnAddEffectChain_l(chain); 6584 6585 mEffectChains.add(chain); 6586 6587 return NO_ERROR; 6588} 6589 6590size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6591{ 6592 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6593 ALOGW_IF(mEffectChains.size() != 1, 6594 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6595 chain.get(), mEffectChains.size(), this); 6596 if (mEffectChains.size() == 1) { 6597 mEffectChains.removeAt(0); 6598 } 6599 return 0; 6600} 6601 6602// ---------------------------------------------------------------------------- 6603// EffectModule implementation 6604// ---------------------------------------------------------------------------- 6605 6606#undef LOG_TAG 6607#define LOG_TAG "AudioFlinger::EffectModule" 6608 6609AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6610 const wp<AudioFlinger::EffectChain>& chain, 6611 effect_descriptor_t *desc, 6612 int id, 6613 int sessionId) 6614 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6615 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6616{ 6617 ALOGV("Constructor %p", this); 6618 int lStatus; 6619 if (thread == NULL) { 6620 return; 6621 } 6622 6623 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6624 6625 // create effect engine from effect factory 6626 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6627 6628 if (mStatus != NO_ERROR) { 6629 return; 6630 } 6631 lStatus = init(); 6632 if (lStatus < 0) { 6633 mStatus = lStatus; 6634 goto Error; 6635 } 6636 6637 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6638 mPinned = true; 6639 } 6640 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6641 return; 6642Error: 6643 EffectRelease(mEffectInterface); 6644 mEffectInterface = NULL; 6645 ALOGV("Constructor Error %d", mStatus); 6646} 6647 6648AudioFlinger::EffectModule::~EffectModule() 6649{ 6650 ALOGV("Destructor %p", this); 6651 if (mEffectInterface != NULL) { 6652 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6653 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6654 sp<ThreadBase> thread = mThread.promote(); 6655 if (thread != 0) { 6656 audio_stream_t *stream = thread->stream(); 6657 if (stream != NULL) { 6658 stream->remove_audio_effect(stream, mEffectInterface); 6659 } 6660 } 6661 } 6662 // release effect engine 6663 EffectRelease(mEffectInterface); 6664 } 6665} 6666 6667status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6668{ 6669 status_t status; 6670 6671 Mutex::Autolock _l(mLock); 6672 int priority = handle->priority(); 6673 size_t size = mHandles.size(); 6674 sp<EffectHandle> h; 6675 size_t i; 6676 for (i = 0; i < size; i++) { 6677 h = mHandles[i].promote(); 6678 if (h == 0) continue; 6679 if (h->priority() <= priority) break; 6680 } 6681 // if inserted in first place, move effect control from previous owner to this handle 6682 if (i == 0) { 6683 bool enabled = false; 6684 if (h != 0) { 6685 enabled = h->enabled(); 6686 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6687 } 6688 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6689 status = NO_ERROR; 6690 } else { 6691 status = ALREADY_EXISTS; 6692 } 6693 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6694 mHandles.insertAt(handle, i); 6695 return status; 6696} 6697 6698size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6699{ 6700 Mutex::Autolock _l(mLock); 6701 size_t size = mHandles.size(); 6702 size_t i; 6703 for (i = 0; i < size; i++) { 6704 if (mHandles[i] == handle) break; 6705 } 6706 if (i == size) { 6707 return size; 6708 } 6709 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6710 6711 bool enabled = false; 6712 EffectHandle *hdl = handle.unsafe_get(); 6713 if (hdl != NULL) { 6714 ALOGV("removeHandle() unsafe_get OK"); 6715 enabled = hdl->enabled(); 6716 } 6717 mHandles.removeAt(i); 6718 size = mHandles.size(); 6719 // if removed from first place, move effect control from this handle to next in line 6720 if (i == 0 && size != 0) { 6721 sp<EffectHandle> h = mHandles[0].promote(); 6722 if (h != 0) { 6723 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6724 } 6725 } 6726 6727 // Prevent calls to process() and other functions on effect interface from now on. 6728 // The effect engine will be released by the destructor when the last strong reference on 6729 // this object is released which can happen after next process is called. 6730 if (size == 0 && !mPinned) { 6731 mState = DESTROYED; 6732 } 6733 6734 return size; 6735} 6736 6737sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6738{ 6739 Mutex::Autolock _l(mLock); 6740 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6741} 6742 6743void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6744{ 6745 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6746 // keep a strong reference on this EffectModule to avoid calling the 6747 // destructor before we exit 6748 sp<EffectModule> keep(this); 6749 { 6750 sp<ThreadBase> thread = mThread.promote(); 6751 if (thread != 0) { 6752 thread->disconnectEffect(keep, handle, unpinIfLast); 6753 } 6754 } 6755} 6756 6757void AudioFlinger::EffectModule::updateState() { 6758 Mutex::Autolock _l(mLock); 6759 6760 switch (mState) { 6761 case RESTART: 6762 reset_l(); 6763 // FALL THROUGH 6764 6765 case STARTING: 6766 // clear auxiliary effect input buffer for next accumulation 6767 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6768 memset(mConfig.inputCfg.buffer.raw, 6769 0, 6770 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6771 } 6772 start_l(); 6773 mState = ACTIVE; 6774 break; 6775 case STOPPING: 6776 stop_l(); 6777 mDisableWaitCnt = mMaxDisableWaitCnt; 6778 mState = STOPPED; 6779 break; 6780 case STOPPED: 6781 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6782 // turn off sequence. 6783 if (--mDisableWaitCnt == 0) { 6784 reset_l(); 6785 mState = IDLE; 6786 } 6787 break; 6788 default: //IDLE , ACTIVE, DESTROYED 6789 break; 6790 } 6791} 6792 6793void AudioFlinger::EffectModule::process() 6794{ 6795 Mutex::Autolock _l(mLock); 6796 6797 if (mState == DESTROYED || mEffectInterface == NULL || 6798 mConfig.inputCfg.buffer.raw == NULL || 6799 mConfig.outputCfg.buffer.raw == NULL) { 6800 return; 6801 } 6802 6803 if (isProcessEnabled()) { 6804 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6805 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6806 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6807 mConfig.inputCfg.buffer.s32, 6808 mConfig.inputCfg.buffer.frameCount/2); 6809 } 6810 6811 // do the actual processing in the effect engine 6812 int ret = (*mEffectInterface)->process(mEffectInterface, 6813 &mConfig.inputCfg.buffer, 6814 &mConfig.outputCfg.buffer); 6815 6816 // force transition to IDLE state when engine is ready 6817 if (mState == STOPPED && ret == -ENODATA) { 6818 mDisableWaitCnt = 1; 6819 } 6820 6821 // clear auxiliary effect input buffer for next accumulation 6822 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6823 memset(mConfig.inputCfg.buffer.raw, 0, 6824 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6825 } 6826 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6827 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6828 // If an insert effect is idle and input buffer is different from output buffer, 6829 // accumulate input onto output 6830 sp<EffectChain> chain = mChain.promote(); 6831 if (chain != 0 && chain->activeTrackCnt() != 0) { 6832 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6833 int16_t *in = mConfig.inputCfg.buffer.s16; 6834 int16_t *out = mConfig.outputCfg.buffer.s16; 6835 for (size_t i = 0; i < frameCnt; i++) { 6836 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6837 } 6838 } 6839 } 6840} 6841 6842void AudioFlinger::EffectModule::reset_l() 6843{ 6844 if (mEffectInterface == NULL) { 6845 return; 6846 } 6847 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6848} 6849 6850status_t AudioFlinger::EffectModule::configure() 6851{ 6852 uint32_t channels; 6853 if (mEffectInterface == NULL) { 6854 return NO_INIT; 6855 } 6856 6857 sp<ThreadBase> thread = mThread.promote(); 6858 if (thread == 0) { 6859 return DEAD_OBJECT; 6860 } 6861 6862 // TODO: handle configuration of effects replacing track process 6863 if (thread->channelCount() == 1) { 6864 channels = AUDIO_CHANNEL_OUT_MONO; 6865 } else { 6866 channels = AUDIO_CHANNEL_OUT_STEREO; 6867 } 6868 6869 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6870 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6871 } else { 6872 mConfig.inputCfg.channels = channels; 6873 } 6874 mConfig.outputCfg.channels = channels; 6875 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6876 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6877 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6878 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6879 mConfig.inputCfg.bufferProvider.cookie = NULL; 6880 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6881 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6882 mConfig.outputCfg.bufferProvider.cookie = NULL; 6883 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6884 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6885 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6886 // Insert effect: 6887 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6888 // always overwrites output buffer: input buffer == output buffer 6889 // - in other sessions: 6890 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6891 // other effect: overwrites output buffer: input buffer == output buffer 6892 // Auxiliary effect: 6893 // accumulates in output buffer: input buffer != output buffer 6894 // Therefore: accumulate <=> input buffer != output buffer 6895 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6896 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6897 } else { 6898 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6899 } 6900 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6901 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6902 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6903 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6904 6905 ALOGV("configure() %p thread %p buffer %p framecount %d", 6906 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6907 6908 status_t cmdStatus; 6909 uint32_t size = sizeof(int); 6910 status_t status = (*mEffectInterface)->command(mEffectInterface, 6911 EFFECT_CMD_SET_CONFIG, 6912 sizeof(effect_config_t), 6913 &mConfig, 6914 &size, 6915 &cmdStatus); 6916 if (status == 0) { 6917 status = cmdStatus; 6918 } 6919 6920 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6921 (1000 * mConfig.outputCfg.buffer.frameCount); 6922 6923 return status; 6924} 6925 6926status_t AudioFlinger::EffectModule::init() 6927{ 6928 Mutex::Autolock _l(mLock); 6929 if (mEffectInterface == NULL) { 6930 return NO_INIT; 6931 } 6932 status_t cmdStatus; 6933 uint32_t size = sizeof(status_t); 6934 status_t status = (*mEffectInterface)->command(mEffectInterface, 6935 EFFECT_CMD_INIT, 6936 0, 6937 NULL, 6938 &size, 6939 &cmdStatus); 6940 if (status == 0) { 6941 status = cmdStatus; 6942 } 6943 return status; 6944} 6945 6946status_t AudioFlinger::EffectModule::start() 6947{ 6948 Mutex::Autolock _l(mLock); 6949 return start_l(); 6950} 6951 6952status_t AudioFlinger::EffectModule::start_l() 6953{ 6954 if (mEffectInterface == NULL) { 6955 return NO_INIT; 6956 } 6957 status_t cmdStatus; 6958 uint32_t size = sizeof(status_t); 6959 status_t status = (*mEffectInterface)->command(mEffectInterface, 6960 EFFECT_CMD_ENABLE, 6961 0, 6962 NULL, 6963 &size, 6964 &cmdStatus); 6965 if (status == 0) { 6966 status = cmdStatus; 6967 } 6968 if (status == 0 && 6969 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6970 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6971 sp<ThreadBase> thread = mThread.promote(); 6972 if (thread != 0) { 6973 audio_stream_t *stream = thread->stream(); 6974 if (stream != NULL) { 6975 stream->add_audio_effect(stream, mEffectInterface); 6976 } 6977 } 6978 } 6979 return status; 6980} 6981 6982status_t AudioFlinger::EffectModule::stop() 6983{ 6984 Mutex::Autolock _l(mLock); 6985 return stop_l(); 6986} 6987 6988status_t AudioFlinger::EffectModule::stop_l() 6989{ 6990 if (mEffectInterface == NULL) { 6991 return NO_INIT; 6992 } 6993 status_t cmdStatus; 6994 uint32_t size = sizeof(status_t); 6995 status_t status = (*mEffectInterface)->command(mEffectInterface, 6996 EFFECT_CMD_DISABLE, 6997 0, 6998 NULL, 6999 &size, 7000 &cmdStatus); 7001 if (status == 0) { 7002 status = cmdStatus; 7003 } 7004 if (status == 0 && 7005 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7006 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7007 sp<ThreadBase> thread = mThread.promote(); 7008 if (thread != 0) { 7009 audio_stream_t *stream = thread->stream(); 7010 if (stream != NULL) { 7011 stream->remove_audio_effect(stream, mEffectInterface); 7012 } 7013 } 7014 } 7015 return status; 7016} 7017 7018status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7019 uint32_t cmdSize, 7020 void *pCmdData, 7021 uint32_t *replySize, 7022 void *pReplyData) 7023{ 7024 Mutex::Autolock _l(mLock); 7025// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7026 7027 if (mState == DESTROYED || mEffectInterface == NULL) { 7028 return NO_INIT; 7029 } 7030 status_t status = (*mEffectInterface)->command(mEffectInterface, 7031 cmdCode, 7032 cmdSize, 7033 pCmdData, 7034 replySize, 7035 pReplyData); 7036 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7037 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7038 for (size_t i = 1; i < mHandles.size(); i++) { 7039 sp<EffectHandle> h = mHandles[i].promote(); 7040 if (h != 0) { 7041 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7042 } 7043 } 7044 } 7045 return status; 7046} 7047 7048status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7049{ 7050 7051 Mutex::Autolock _l(mLock); 7052 ALOGV("setEnabled %p enabled %d", this, enabled); 7053 7054 if (enabled != isEnabled()) { 7055 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7056 if (enabled && status != NO_ERROR) { 7057 return status; 7058 } 7059 7060 switch (mState) { 7061 // going from disabled to enabled 7062 case IDLE: 7063 mState = STARTING; 7064 break; 7065 case STOPPED: 7066 mState = RESTART; 7067 break; 7068 case STOPPING: 7069 mState = ACTIVE; 7070 break; 7071 7072 // going from enabled to disabled 7073 case RESTART: 7074 mState = STOPPED; 7075 break; 7076 case STARTING: 7077 mState = IDLE; 7078 break; 7079 case ACTIVE: 7080 mState = STOPPING; 7081 break; 7082 case DESTROYED: 7083 return NO_ERROR; // simply ignore as we are being destroyed 7084 } 7085 for (size_t i = 1; i < mHandles.size(); i++) { 7086 sp<EffectHandle> h = mHandles[i].promote(); 7087 if (h != 0) { 7088 h->setEnabled(enabled); 7089 } 7090 } 7091 } 7092 return NO_ERROR; 7093} 7094 7095bool AudioFlinger::EffectModule::isEnabled() const 7096{ 7097 switch (mState) { 7098 case RESTART: 7099 case STARTING: 7100 case ACTIVE: 7101 return true; 7102 case IDLE: 7103 case STOPPING: 7104 case STOPPED: 7105 case DESTROYED: 7106 default: 7107 return false; 7108 } 7109} 7110 7111bool AudioFlinger::EffectModule::isProcessEnabled() const 7112{ 7113 switch (mState) { 7114 case RESTART: 7115 case ACTIVE: 7116 case STOPPING: 7117 case STOPPED: 7118 return true; 7119 case IDLE: 7120 case STARTING: 7121 case DESTROYED: 7122 default: 7123 return false; 7124 } 7125} 7126 7127status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7128{ 7129 Mutex::Autolock _l(mLock); 7130 status_t status = NO_ERROR; 7131 7132 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7133 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7134 if (isProcessEnabled() && 7135 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7136 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7137 status_t cmdStatus; 7138 uint32_t volume[2]; 7139 uint32_t *pVolume = NULL; 7140 uint32_t size = sizeof(volume); 7141 volume[0] = *left; 7142 volume[1] = *right; 7143 if (controller) { 7144 pVolume = volume; 7145 } 7146 status = (*mEffectInterface)->command(mEffectInterface, 7147 EFFECT_CMD_SET_VOLUME, 7148 size, 7149 volume, 7150 &size, 7151 pVolume); 7152 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7153 *left = volume[0]; 7154 *right = volume[1]; 7155 } 7156 } 7157 return status; 7158} 7159 7160status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7161{ 7162 Mutex::Autolock _l(mLock); 7163 status_t status = NO_ERROR; 7164 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7165 // audio pre processing modules on RecordThread can receive both output and 7166 // input device indication in the same call 7167 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7168 if (dev) { 7169 status_t cmdStatus; 7170 uint32_t size = sizeof(status_t); 7171 7172 status = (*mEffectInterface)->command(mEffectInterface, 7173 EFFECT_CMD_SET_DEVICE, 7174 sizeof(uint32_t), 7175 &dev, 7176 &size, 7177 &cmdStatus); 7178 if (status == NO_ERROR) { 7179 status = cmdStatus; 7180 } 7181 } 7182 dev = device & AUDIO_DEVICE_IN_ALL; 7183 if (dev) { 7184 status_t cmdStatus; 7185 uint32_t size = sizeof(status_t); 7186 7187 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7188 EFFECT_CMD_SET_INPUT_DEVICE, 7189 sizeof(uint32_t), 7190 &dev, 7191 &size, 7192 &cmdStatus); 7193 if (status2 == NO_ERROR) { 7194 status2 = cmdStatus; 7195 } 7196 if (status == NO_ERROR) { 7197 status = status2; 7198 } 7199 } 7200 } 7201 return status; 7202} 7203 7204status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7205{ 7206 Mutex::Autolock _l(mLock); 7207 status_t status = NO_ERROR; 7208 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7209 status_t cmdStatus; 7210 uint32_t size = sizeof(status_t); 7211 status = (*mEffectInterface)->command(mEffectInterface, 7212 EFFECT_CMD_SET_AUDIO_MODE, 7213 sizeof(audio_mode_t), 7214 &mode, 7215 &size, 7216 &cmdStatus); 7217 if (status == NO_ERROR) { 7218 status = cmdStatus; 7219 } 7220 } 7221 return status; 7222} 7223 7224void AudioFlinger::EffectModule::setSuspended(bool suspended) 7225{ 7226 Mutex::Autolock _l(mLock); 7227 mSuspended = suspended; 7228} 7229 7230bool AudioFlinger::EffectModule::suspended() const 7231{ 7232 Mutex::Autolock _l(mLock); 7233 return mSuspended; 7234} 7235 7236status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7237{ 7238 const size_t SIZE = 256; 7239 char buffer[SIZE]; 7240 String8 result; 7241 7242 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7243 result.append(buffer); 7244 7245 bool locked = tryLock(mLock); 7246 // failed to lock - AudioFlinger is probably deadlocked 7247 if (!locked) { 7248 result.append("\t\tCould not lock Fx mutex:\n"); 7249 } 7250 7251 result.append("\t\tSession Status State Engine:\n"); 7252 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7253 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7254 result.append(buffer); 7255 7256 result.append("\t\tDescriptor:\n"); 7257 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7258 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7259 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7260 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7261 result.append(buffer); 7262 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7263 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7264 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7265 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7266 result.append(buffer); 7267 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7268 mDescriptor.apiVersion, 7269 mDescriptor.flags); 7270 result.append(buffer); 7271 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7272 mDescriptor.name); 7273 result.append(buffer); 7274 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7275 mDescriptor.implementor); 7276 result.append(buffer); 7277 7278 result.append("\t\t- Input configuration:\n"); 7279 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7280 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7281 (uint32_t)mConfig.inputCfg.buffer.raw, 7282 mConfig.inputCfg.buffer.frameCount, 7283 mConfig.inputCfg.samplingRate, 7284 mConfig.inputCfg.channels, 7285 mConfig.inputCfg.format); 7286 result.append(buffer); 7287 7288 result.append("\t\t- Output configuration:\n"); 7289 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7290 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7291 (uint32_t)mConfig.outputCfg.buffer.raw, 7292 mConfig.outputCfg.buffer.frameCount, 7293 mConfig.outputCfg.samplingRate, 7294 mConfig.outputCfg.channels, 7295 mConfig.outputCfg.format); 7296 result.append(buffer); 7297 7298 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7299 result.append(buffer); 7300 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7301 for (size_t i = 0; i < mHandles.size(); ++i) { 7302 sp<EffectHandle> handle = mHandles[i].promote(); 7303 if (handle != 0) { 7304 handle->dump(buffer, SIZE); 7305 result.append(buffer); 7306 } 7307 } 7308 7309 result.append("\n"); 7310 7311 write(fd, result.string(), result.length()); 7312 7313 if (locked) { 7314 mLock.unlock(); 7315 } 7316 7317 return NO_ERROR; 7318} 7319 7320// ---------------------------------------------------------------------------- 7321// EffectHandle implementation 7322// ---------------------------------------------------------------------------- 7323 7324#undef LOG_TAG 7325#define LOG_TAG "AudioFlinger::EffectHandle" 7326 7327AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7328 const sp<AudioFlinger::Client>& client, 7329 const sp<IEffectClient>& effectClient, 7330 int32_t priority) 7331 : BnEffect(), 7332 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7333 mPriority(priority), mHasControl(false), mEnabled(false) 7334{ 7335 ALOGV("constructor %p", this); 7336 7337 if (client == 0) { 7338 return; 7339 } 7340 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7341 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7342 if (mCblkMemory != 0) { 7343 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7344 7345 if (mCblk != NULL) { 7346 new(mCblk) effect_param_cblk_t(); 7347 mBuffer = (uint8_t *)mCblk + bufOffset; 7348 } 7349 } else { 7350 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7351 return; 7352 } 7353} 7354 7355AudioFlinger::EffectHandle::~EffectHandle() 7356{ 7357 ALOGV("Destructor %p", this); 7358 disconnect(false); 7359 ALOGV("Destructor DONE %p", this); 7360} 7361 7362status_t AudioFlinger::EffectHandle::enable() 7363{ 7364 ALOGV("enable %p", this); 7365 if (!mHasControl) return INVALID_OPERATION; 7366 if (mEffect == 0) return DEAD_OBJECT; 7367 7368 if (mEnabled) { 7369 return NO_ERROR; 7370 } 7371 7372 mEnabled = true; 7373 7374 sp<ThreadBase> thread = mEffect->thread().promote(); 7375 if (thread != 0) { 7376 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7377 } 7378 7379 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7380 if (mEffect->suspended()) { 7381 return NO_ERROR; 7382 } 7383 7384 status_t status = mEffect->setEnabled(true); 7385 if (status != NO_ERROR) { 7386 if (thread != 0) { 7387 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7388 } 7389 mEnabled = false; 7390 } 7391 return status; 7392} 7393 7394status_t AudioFlinger::EffectHandle::disable() 7395{ 7396 ALOGV("disable %p", this); 7397 if (!mHasControl) return INVALID_OPERATION; 7398 if (mEffect == 0) return DEAD_OBJECT; 7399 7400 if (!mEnabled) { 7401 return NO_ERROR; 7402 } 7403 mEnabled = false; 7404 7405 if (mEffect->suspended()) { 7406 return NO_ERROR; 7407 } 7408 7409 status_t status = mEffect->setEnabled(false); 7410 7411 sp<ThreadBase> thread = mEffect->thread().promote(); 7412 if (thread != 0) { 7413 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7414 } 7415 7416 return status; 7417} 7418 7419void AudioFlinger::EffectHandle::disconnect() 7420{ 7421 disconnect(true); 7422} 7423 7424void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7425{ 7426 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7427 if (mEffect == 0) { 7428 return; 7429 } 7430 mEffect->disconnect(this, unpinIfLast); 7431 7432 if (mHasControl && mEnabled) { 7433 sp<ThreadBase> thread = mEffect->thread().promote(); 7434 if (thread != 0) { 7435 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7436 } 7437 } 7438 7439 // release sp on module => module destructor can be called now 7440 mEffect.clear(); 7441 if (mClient != 0) { 7442 if (mCblk != NULL) { 7443 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7444 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7445 } 7446 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7447 // Client destructor must run with AudioFlinger mutex locked 7448 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7449 mClient.clear(); 7450 } 7451} 7452 7453status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7454 uint32_t cmdSize, 7455 void *pCmdData, 7456 uint32_t *replySize, 7457 void *pReplyData) 7458{ 7459// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7460// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7461 7462 // only get parameter command is permitted for applications not controlling the effect 7463 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7464 return INVALID_OPERATION; 7465 } 7466 if (mEffect == 0) return DEAD_OBJECT; 7467 if (mClient == 0) return INVALID_OPERATION; 7468 7469 // handle commands that are not forwarded transparently to effect engine 7470 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7471 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7472 // no risk to block the whole media server process or mixer threads is we are stuck here 7473 Mutex::Autolock _l(mCblk->lock); 7474 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7475 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7476 mCblk->serverIndex = 0; 7477 mCblk->clientIndex = 0; 7478 return BAD_VALUE; 7479 } 7480 status_t status = NO_ERROR; 7481 while (mCblk->serverIndex < mCblk->clientIndex) { 7482 int reply; 7483 uint32_t rsize = sizeof(int); 7484 int *p = (int *)(mBuffer + mCblk->serverIndex); 7485 int size = *p++; 7486 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7487 ALOGW("command(): invalid parameter block size"); 7488 break; 7489 } 7490 effect_param_t *param = (effect_param_t *)p; 7491 if (param->psize == 0 || param->vsize == 0) { 7492 ALOGW("command(): null parameter or value size"); 7493 mCblk->serverIndex += size; 7494 continue; 7495 } 7496 uint32_t psize = sizeof(effect_param_t) + 7497 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7498 param->vsize; 7499 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7500 psize, 7501 p, 7502 &rsize, 7503 &reply); 7504 // stop at first error encountered 7505 if (ret != NO_ERROR) { 7506 status = ret; 7507 *(int *)pReplyData = reply; 7508 break; 7509 } else if (reply != NO_ERROR) { 7510 *(int *)pReplyData = reply; 7511 break; 7512 } 7513 mCblk->serverIndex += size; 7514 } 7515 mCblk->serverIndex = 0; 7516 mCblk->clientIndex = 0; 7517 return status; 7518 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7519 *(int *)pReplyData = NO_ERROR; 7520 return enable(); 7521 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7522 *(int *)pReplyData = NO_ERROR; 7523 return disable(); 7524 } 7525 7526 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7527} 7528 7529void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7530{ 7531 ALOGV("setControl %p control %d", this, hasControl); 7532 7533 mHasControl = hasControl; 7534 mEnabled = enabled; 7535 7536 if (signal && mEffectClient != 0) { 7537 mEffectClient->controlStatusChanged(hasControl); 7538 } 7539} 7540 7541void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7542 uint32_t cmdSize, 7543 void *pCmdData, 7544 uint32_t replySize, 7545 void *pReplyData) 7546{ 7547 if (mEffectClient != 0) { 7548 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7549 } 7550} 7551 7552 7553 7554void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7555{ 7556 if (mEffectClient != 0) { 7557 mEffectClient->enableStatusChanged(enabled); 7558 } 7559} 7560 7561status_t AudioFlinger::EffectHandle::onTransact( 7562 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7563{ 7564 return BnEffect::onTransact(code, data, reply, flags); 7565} 7566 7567 7568void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7569{ 7570 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7571 7572 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7573 (mClient == 0) ? getpid_cached : mClient->pid(), 7574 mPriority, 7575 mHasControl, 7576 !locked, 7577 mCblk ? mCblk->clientIndex : 0, 7578 mCblk ? mCblk->serverIndex : 0 7579 ); 7580 7581 if (locked) { 7582 mCblk->lock.unlock(); 7583 } 7584} 7585 7586#undef LOG_TAG 7587#define LOG_TAG "AudioFlinger::EffectChain" 7588 7589AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7590 int sessionId) 7591 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7592 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7593 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7594{ 7595 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7596 if (thread == NULL) { 7597 return; 7598 } 7599 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7600 thread->frameCount(); 7601} 7602 7603AudioFlinger::EffectChain::~EffectChain() 7604{ 7605 if (mOwnInBuffer) { 7606 delete mInBuffer; 7607 } 7608 7609} 7610 7611// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7612sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7613{ 7614 size_t size = mEffects.size(); 7615 7616 for (size_t i = 0; i < size; i++) { 7617 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7618 return mEffects[i]; 7619 } 7620 } 7621 return 0; 7622} 7623 7624// getEffectFromId_l() must be called with ThreadBase::mLock held 7625sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7626{ 7627 size_t size = mEffects.size(); 7628 7629 for (size_t i = 0; i < size; i++) { 7630 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7631 if (id == 0 || mEffects[i]->id() == id) { 7632 return mEffects[i]; 7633 } 7634 } 7635 return 0; 7636} 7637 7638// getEffectFromType_l() must be called with ThreadBase::mLock held 7639sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7640 const effect_uuid_t *type) 7641{ 7642 size_t size = mEffects.size(); 7643 7644 for (size_t i = 0; i < size; i++) { 7645 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7646 return mEffects[i]; 7647 } 7648 } 7649 return 0; 7650} 7651 7652// Must be called with EffectChain::mLock locked 7653void AudioFlinger::EffectChain::process_l() 7654{ 7655 sp<ThreadBase> thread = mThread.promote(); 7656 if (thread == 0) { 7657 ALOGW("process_l(): cannot promote mixer thread"); 7658 return; 7659 } 7660 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7661 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7662 // always process effects unless no more tracks are on the session and the effect tail 7663 // has been rendered 7664 bool doProcess = true; 7665 if (!isGlobalSession) { 7666 bool tracksOnSession = (trackCnt() != 0); 7667 7668 if (!tracksOnSession && mTailBufferCount == 0) { 7669 doProcess = false; 7670 } 7671 7672 if (activeTrackCnt() == 0) { 7673 // if no track is active and the effect tail has not been rendered, 7674 // the input buffer must be cleared here as the mixer process will not do it 7675 if (tracksOnSession || mTailBufferCount > 0) { 7676 size_t numSamples = thread->frameCount() * thread->channelCount(); 7677 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7678 if (mTailBufferCount > 0) { 7679 mTailBufferCount--; 7680 } 7681 } 7682 } 7683 } 7684 7685 size_t size = mEffects.size(); 7686 if (doProcess) { 7687 for (size_t i = 0; i < size; i++) { 7688 mEffects[i]->process(); 7689 } 7690 } 7691 for (size_t i = 0; i < size; i++) { 7692 mEffects[i]->updateState(); 7693 } 7694} 7695 7696// addEffect_l() must be called with PlaybackThread::mLock held 7697status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7698{ 7699 effect_descriptor_t desc = effect->desc(); 7700 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7701 7702 Mutex::Autolock _l(mLock); 7703 effect->setChain(this); 7704 sp<ThreadBase> thread = mThread.promote(); 7705 if (thread == 0) { 7706 return NO_INIT; 7707 } 7708 effect->setThread(thread); 7709 7710 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7711 // Auxiliary effects are inserted at the beginning of mEffects vector as 7712 // they are processed first and accumulated in chain input buffer 7713 mEffects.insertAt(effect, 0); 7714 7715 // the input buffer for auxiliary effect contains mono samples in 7716 // 32 bit format. This is to avoid saturation in AudoMixer 7717 // accumulation stage. Saturation is done in EffectModule::process() before 7718 // calling the process in effect engine 7719 size_t numSamples = thread->frameCount(); 7720 int32_t *buffer = new int32_t[numSamples]; 7721 memset(buffer, 0, numSamples * sizeof(int32_t)); 7722 effect->setInBuffer((int16_t *)buffer); 7723 // auxiliary effects output samples to chain input buffer for further processing 7724 // by insert effects 7725 effect->setOutBuffer(mInBuffer); 7726 } else { 7727 // Insert effects are inserted at the end of mEffects vector as they are processed 7728 // after track and auxiliary effects. 7729 // Insert effect order as a function of indicated preference: 7730 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7731 // another effect is present 7732 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7733 // last effect claiming first position 7734 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7735 // first effect claiming last position 7736 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7737 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7738 // already present 7739 7740 size_t size = mEffects.size(); 7741 size_t idx_insert = size; 7742 ssize_t idx_insert_first = -1; 7743 ssize_t idx_insert_last = -1; 7744 7745 for (size_t i = 0; i < size; i++) { 7746 effect_descriptor_t d = mEffects[i]->desc(); 7747 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7748 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7749 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7750 // check invalid effect chaining combinations 7751 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7752 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7753 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7754 return INVALID_OPERATION; 7755 } 7756 // remember position of first insert effect and by default 7757 // select this as insert position for new effect 7758 if (idx_insert == size) { 7759 idx_insert = i; 7760 } 7761 // remember position of last insert effect claiming 7762 // first position 7763 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7764 idx_insert_first = i; 7765 } 7766 // remember position of first insert effect claiming 7767 // last position 7768 if (iPref == EFFECT_FLAG_INSERT_LAST && 7769 idx_insert_last == -1) { 7770 idx_insert_last = i; 7771 } 7772 } 7773 } 7774 7775 // modify idx_insert from first position if needed 7776 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7777 if (idx_insert_last != -1) { 7778 idx_insert = idx_insert_last; 7779 } else { 7780 idx_insert = size; 7781 } 7782 } else { 7783 if (idx_insert_first != -1) { 7784 idx_insert = idx_insert_first + 1; 7785 } 7786 } 7787 7788 // always read samples from chain input buffer 7789 effect->setInBuffer(mInBuffer); 7790 7791 // if last effect in the chain, output samples to chain 7792 // output buffer, otherwise to chain input buffer 7793 if (idx_insert == size) { 7794 if (idx_insert != 0) { 7795 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7796 mEffects[idx_insert-1]->configure(); 7797 } 7798 effect->setOutBuffer(mOutBuffer); 7799 } else { 7800 effect->setOutBuffer(mInBuffer); 7801 } 7802 mEffects.insertAt(effect, idx_insert); 7803 7804 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7805 } 7806 effect->configure(); 7807 return NO_ERROR; 7808} 7809 7810// removeEffect_l() must be called with PlaybackThread::mLock held 7811size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7812{ 7813 Mutex::Autolock _l(mLock); 7814 size_t size = mEffects.size(); 7815 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7816 7817 for (size_t i = 0; i < size; i++) { 7818 if (effect == mEffects[i]) { 7819 // calling stop here will remove pre-processing effect from the audio HAL. 7820 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7821 // the middle of a read from audio HAL 7822 if (mEffects[i]->state() == EffectModule::ACTIVE || 7823 mEffects[i]->state() == EffectModule::STOPPING) { 7824 mEffects[i]->stop(); 7825 } 7826 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7827 delete[] effect->inBuffer(); 7828 } else { 7829 if (i == size - 1 && i != 0) { 7830 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7831 mEffects[i - 1]->configure(); 7832 } 7833 } 7834 mEffects.removeAt(i); 7835 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7836 break; 7837 } 7838 } 7839 7840 return mEffects.size(); 7841} 7842 7843// setDevice_l() must be called with PlaybackThread::mLock held 7844void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7845{ 7846 size_t size = mEffects.size(); 7847 for (size_t i = 0; i < size; i++) { 7848 mEffects[i]->setDevice(device); 7849 } 7850} 7851 7852// setMode_l() must be called with PlaybackThread::mLock held 7853void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7854{ 7855 size_t size = mEffects.size(); 7856 for (size_t i = 0; i < size; i++) { 7857 mEffects[i]->setMode(mode); 7858 } 7859} 7860 7861// setVolume_l() must be called with PlaybackThread::mLock held 7862bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7863{ 7864 uint32_t newLeft = *left; 7865 uint32_t newRight = *right; 7866 bool hasControl = false; 7867 int ctrlIdx = -1; 7868 size_t size = mEffects.size(); 7869 7870 // first update volume controller 7871 for (size_t i = size; i > 0; i--) { 7872 if (mEffects[i - 1]->isProcessEnabled() && 7873 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7874 ctrlIdx = i - 1; 7875 hasControl = true; 7876 break; 7877 } 7878 } 7879 7880 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7881 if (hasControl) { 7882 *left = mNewLeftVolume; 7883 *right = mNewRightVolume; 7884 } 7885 return hasControl; 7886 } 7887 7888 mVolumeCtrlIdx = ctrlIdx; 7889 mLeftVolume = newLeft; 7890 mRightVolume = newRight; 7891 7892 // second get volume update from volume controller 7893 if (ctrlIdx >= 0) { 7894 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7895 mNewLeftVolume = newLeft; 7896 mNewRightVolume = newRight; 7897 } 7898 // then indicate volume to all other effects in chain. 7899 // Pass altered volume to effects before volume controller 7900 // and requested volume to effects after controller 7901 uint32_t lVol = newLeft; 7902 uint32_t rVol = newRight; 7903 7904 for (size_t i = 0; i < size; i++) { 7905 if ((int)i == ctrlIdx) continue; 7906 // this also works for ctrlIdx == -1 when there is no volume controller 7907 if ((int)i > ctrlIdx) { 7908 lVol = *left; 7909 rVol = *right; 7910 } 7911 mEffects[i]->setVolume(&lVol, &rVol, false); 7912 } 7913 *left = newLeft; 7914 *right = newRight; 7915 7916 return hasControl; 7917} 7918 7919status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7920{ 7921 const size_t SIZE = 256; 7922 char buffer[SIZE]; 7923 String8 result; 7924 7925 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7926 result.append(buffer); 7927 7928 bool locked = tryLock(mLock); 7929 // failed to lock - AudioFlinger is probably deadlocked 7930 if (!locked) { 7931 result.append("\tCould not lock mutex:\n"); 7932 } 7933 7934 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7935 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7936 mEffects.size(), 7937 (uint32_t)mInBuffer, 7938 (uint32_t)mOutBuffer, 7939 mActiveTrackCnt); 7940 result.append(buffer); 7941 write(fd, result.string(), result.size()); 7942 7943 for (size_t i = 0; i < mEffects.size(); ++i) { 7944 sp<EffectModule> effect = mEffects[i]; 7945 if (effect != 0) { 7946 effect->dump(fd, args); 7947 } 7948 } 7949 7950 if (locked) { 7951 mLock.unlock(); 7952 } 7953 7954 return NO_ERROR; 7955} 7956 7957// must be called with ThreadBase::mLock held 7958void AudioFlinger::EffectChain::setEffectSuspended_l( 7959 const effect_uuid_t *type, bool suspend) 7960{ 7961 sp<SuspendedEffectDesc> desc; 7962 // use effect type UUID timelow as key as there is no real risk of identical 7963 // timeLow fields among effect type UUIDs. 7964 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7965 if (suspend) { 7966 if (index >= 0) { 7967 desc = mSuspendedEffects.valueAt(index); 7968 } else { 7969 desc = new SuspendedEffectDesc(); 7970 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7971 mSuspendedEffects.add(type->timeLow, desc); 7972 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7973 } 7974 if (desc->mRefCount++ == 0) { 7975 sp<EffectModule> effect = getEffectIfEnabled(type); 7976 if (effect != 0) { 7977 desc->mEffect = effect; 7978 effect->setSuspended(true); 7979 effect->setEnabled(false); 7980 } 7981 } 7982 } else { 7983 if (index < 0) { 7984 return; 7985 } 7986 desc = mSuspendedEffects.valueAt(index); 7987 if (desc->mRefCount <= 0) { 7988 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7989 desc->mRefCount = 1; 7990 } 7991 if (--desc->mRefCount == 0) { 7992 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7993 if (desc->mEffect != 0) { 7994 sp<EffectModule> effect = desc->mEffect.promote(); 7995 if (effect != 0) { 7996 effect->setSuspended(false); 7997 sp<EffectHandle> handle = effect->controlHandle(); 7998 if (handle != 0) { 7999 effect->setEnabled(handle->enabled()); 8000 } 8001 } 8002 desc->mEffect.clear(); 8003 } 8004 mSuspendedEffects.removeItemsAt(index); 8005 } 8006 } 8007} 8008 8009// must be called with ThreadBase::mLock held 8010void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8011{ 8012 sp<SuspendedEffectDesc> desc; 8013 8014 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8015 if (suspend) { 8016 if (index >= 0) { 8017 desc = mSuspendedEffects.valueAt(index); 8018 } else { 8019 desc = new SuspendedEffectDesc(); 8020 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8021 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8022 } 8023 if (desc->mRefCount++ == 0) { 8024 Vector< sp<EffectModule> > effects; 8025 getSuspendEligibleEffects(effects); 8026 for (size_t i = 0; i < effects.size(); i++) { 8027 setEffectSuspended_l(&effects[i]->desc().type, true); 8028 } 8029 } 8030 } else { 8031 if (index < 0) { 8032 return; 8033 } 8034 desc = mSuspendedEffects.valueAt(index); 8035 if (desc->mRefCount <= 0) { 8036 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8037 desc->mRefCount = 1; 8038 } 8039 if (--desc->mRefCount == 0) { 8040 Vector<const effect_uuid_t *> types; 8041 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8042 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8043 continue; 8044 } 8045 types.add(&mSuspendedEffects.valueAt(i)->mType); 8046 } 8047 for (size_t i = 0; i < types.size(); i++) { 8048 setEffectSuspended_l(types[i], false); 8049 } 8050 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8051 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8052 } 8053 } 8054} 8055 8056 8057// The volume effect is used for automated tests only 8058#ifndef OPENSL_ES_H_ 8059static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8060 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8061const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8062#endif //OPENSL_ES_H_ 8063 8064bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8065{ 8066 // auxiliary effects and visualizer are never suspended on output mix 8067 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8068 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8069 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8070 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8071 return false; 8072 } 8073 return true; 8074} 8075 8076void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8077{ 8078 effects.clear(); 8079 for (size_t i = 0; i < mEffects.size(); i++) { 8080 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8081 effects.add(mEffects[i]); 8082 } 8083 } 8084} 8085 8086sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8087 const effect_uuid_t *type) 8088{ 8089 sp<EffectModule> effect = getEffectFromType_l(type); 8090 return effect != 0 && effect->isEnabled() ? effect : 0; 8091} 8092 8093void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8094 bool enabled) 8095{ 8096 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8097 if (enabled) { 8098 if (index < 0) { 8099 // if the effect is not suspend check if all effects are suspended 8100 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8101 if (index < 0) { 8102 return; 8103 } 8104 if (!isEffectEligibleForSuspend(effect->desc())) { 8105 return; 8106 } 8107 setEffectSuspended_l(&effect->desc().type, enabled); 8108 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8109 if (index < 0) { 8110 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8111 return; 8112 } 8113 } 8114 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8115 effect->desc().type.timeLow); 8116 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8117 // if effect is requested to suspended but was not yet enabled, supend it now. 8118 if (desc->mEffect == 0) { 8119 desc->mEffect = effect; 8120 effect->setEnabled(false); 8121 effect->setSuspended(true); 8122 } 8123 } else { 8124 if (index < 0) { 8125 return; 8126 } 8127 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8128 effect->desc().type.timeLow); 8129 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8130 desc->mEffect.clear(); 8131 effect->setSuspended(false); 8132 } 8133} 8134 8135#undef LOG_TAG 8136#define LOG_TAG "AudioFlinger" 8137 8138// ---------------------------------------------------------------------------- 8139 8140status_t AudioFlinger::onTransact( 8141 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8142{ 8143 return BnAudioFlinger::onTransact(code, data, reply, flags); 8144} 8145 8146}; // namespace android 8147