AudioFlinger.cpp revision ea7939a079b3600cab955760839b021326f8cfc3
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 for (size_t i = 0; i < mRecordThreads.size(); i++) { 899 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 900 } 901 } 902 } 903 } 904 if (thread != 0) { 905 return thread->setParameters(keyValuePairs); 906 } 907 return BAD_VALUE; 908} 909 910String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 911{ 912// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 913// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 914 915 if (ioHandle == 0) { 916 String8 out_s8; 917 918 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 919 char *s; 920 { 921 AutoMutex lock(mHardwareLock); 922 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 923 audio_hw_device_t *dev = mAudioHwDevs[i]; 924 s = dev->get_parameters(dev, keys.string()); 925 mHardwareStatus = AUDIO_HW_IDLE; 926 } 927 out_s8 += String8(s ? s : ""); 928 free(s); 929 } 930 return out_s8; 931 } 932 933 Mutex::Autolock _l(mLock); 934 935 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 936 if (playbackThread != NULL) { 937 return playbackThread->getParameters(keys); 938 } 939 RecordThread *recordThread = checkRecordThread_l(ioHandle); 940 if (recordThread != NULL) { 941 return recordThread->getParameters(keys); 942 } 943 return String8(""); 944} 945 946size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 947{ 948 status_t ret = initCheck(); 949 if (ret != NO_ERROR) { 950 return 0; 951 } 952 953 AutoMutex lock(mHardwareLock); 954 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 955 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 956 mHardwareStatus = AUDIO_HW_IDLE; 957 return size; 958} 959 960unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 961{ 962 if (ioHandle == 0) { 963 return 0; 964 } 965 966 Mutex::Autolock _l(mLock); 967 968 RecordThread *recordThread = checkRecordThread_l(ioHandle); 969 if (recordThread != NULL) { 970 return recordThread->getInputFramesLost(); 971 } 972 return 0; 973} 974 975status_t AudioFlinger::setVoiceVolume(float value) 976{ 977 status_t ret = initCheck(); 978 if (ret != NO_ERROR) { 979 return ret; 980 } 981 982 // check calling permissions 983 if (!settingsAllowed()) { 984 return PERMISSION_DENIED; 985 } 986 987 AutoMutex lock(mHardwareLock); 988 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 989 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 990 mHardwareStatus = AUDIO_HW_IDLE; 991 992 return ret; 993} 994 995status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 996 audio_io_handle_t output) const 997{ 998 status_t status; 999 1000 Mutex::Autolock _l(mLock); 1001 1002 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1003 if (playbackThread != NULL) { 1004 return playbackThread->getRenderPosition(halFrames, dspFrames); 1005 } 1006 1007 return BAD_VALUE; 1008} 1009 1010void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1011{ 1012 1013 Mutex::Autolock _l(mLock); 1014 1015 pid_t pid = IPCThreadState::self()->getCallingPid(); 1016 if (mNotificationClients.indexOfKey(pid) < 0) { 1017 sp<NotificationClient> notificationClient = new NotificationClient(this, 1018 client, 1019 pid); 1020 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1021 1022 mNotificationClients.add(pid, notificationClient); 1023 1024 sp<IBinder> binder = client->asBinder(); 1025 binder->linkToDeath(notificationClient); 1026 1027 // the config change is always sent from playback or record threads to avoid deadlock 1028 // with AudioSystem::gLock 1029 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1030 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1031 } 1032 1033 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1034 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1035 } 1036 } 1037} 1038 1039void AudioFlinger::removeNotificationClient(pid_t pid) 1040{ 1041 Mutex::Autolock _l(mLock); 1042 1043 mNotificationClients.removeItem(pid); 1044 1045 ALOGV("%d died, releasing its sessions", pid); 1046 size_t num = mAudioSessionRefs.size(); 1047 bool removed = false; 1048 for (size_t i = 0; i< num; ) { 1049 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1050 ALOGV(" pid %d @ %d", ref->mPid, i); 1051 if (ref->mPid == pid) { 1052 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1053 mAudioSessionRefs.removeAt(i); 1054 delete ref; 1055 removed = true; 1056 num--; 1057 } else { 1058 i++; 1059 } 1060 } 1061 if (removed) { 1062 purgeStaleEffects_l(); 1063 } 1064} 1065 1066// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1067void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1068{ 1069 size_t size = mNotificationClients.size(); 1070 for (size_t i = 0; i < size; i++) { 1071 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1072 param2); 1073 } 1074} 1075 1076// removeClient_l() must be called with AudioFlinger::mLock held 1077void AudioFlinger::removeClient_l(pid_t pid) 1078{ 1079 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1080 mClients.removeItem(pid); 1081} 1082 1083 1084// ---------------------------------------------------------------------------- 1085 1086AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1087 uint32_t device, type_t type) 1088 : Thread(false), 1089 mType(type), 1090 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1091 // mChannelMask 1092 mChannelCount(0), 1093 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1094 mParamStatus(NO_ERROR), 1095 mStandby(false), mId(id), 1096 mDevice(device), 1097 mDeathRecipient(new PMDeathRecipient(this)) 1098{ 1099} 1100 1101AudioFlinger::ThreadBase::~ThreadBase() 1102{ 1103 mParamCond.broadcast(); 1104 // do not lock the mutex in destructor 1105 releaseWakeLock_l(); 1106 if (mPowerManager != 0) { 1107 sp<IBinder> binder = mPowerManager->asBinder(); 1108 binder->unlinkToDeath(mDeathRecipient); 1109 } 1110} 1111 1112void AudioFlinger::ThreadBase::exit() 1113{ 1114 ALOGV("ThreadBase::exit"); 1115 { 1116 // This lock prevents the following race in thread (uniprocessor for illustration): 1117 // if (!exitPending()) { 1118 // // context switch from here to exit() 1119 // // exit() calls requestExit(), what exitPending() observes 1120 // // exit() calls signal(), which is dropped since no waiters 1121 // // context switch back from exit() to here 1122 // mWaitWorkCV.wait(...); 1123 // // now thread is hung 1124 // } 1125 AutoMutex lock(mLock); 1126 requestExit(); 1127 mWaitWorkCV.signal(); 1128 } 1129 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1130 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1131 requestExitAndWait(); 1132} 1133 1134status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1135{ 1136 status_t status; 1137 1138 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1139 Mutex::Autolock _l(mLock); 1140 1141 mNewParameters.add(keyValuePairs); 1142 mWaitWorkCV.signal(); 1143 // wait condition with timeout in case the thread loop has exited 1144 // before the request could be processed 1145 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1146 status = mParamStatus; 1147 mWaitWorkCV.signal(); 1148 } else { 1149 status = TIMED_OUT; 1150 } 1151 return status; 1152} 1153 1154void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1155{ 1156 Mutex::Autolock _l(mLock); 1157 sendConfigEvent_l(event, param); 1158} 1159 1160// sendConfigEvent_l() must be called with ThreadBase::mLock held 1161void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1162{ 1163 ConfigEvent configEvent; 1164 configEvent.mEvent = event; 1165 configEvent.mParam = param; 1166 mConfigEvents.add(configEvent); 1167 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1168 mWaitWorkCV.signal(); 1169} 1170 1171void AudioFlinger::ThreadBase::processConfigEvents() 1172{ 1173 mLock.lock(); 1174 while (!mConfigEvents.isEmpty()) { 1175 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1176 ConfigEvent configEvent = mConfigEvents[0]; 1177 mConfigEvents.removeAt(0); 1178 // release mLock before locking AudioFlinger mLock: lock order is always 1179 // AudioFlinger then ThreadBase to avoid cross deadlock 1180 mLock.unlock(); 1181 mAudioFlinger->mLock.lock(); 1182 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1183 mAudioFlinger->mLock.unlock(); 1184 mLock.lock(); 1185 } 1186 mLock.unlock(); 1187} 1188 1189status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1190{ 1191 const size_t SIZE = 256; 1192 char buffer[SIZE]; 1193 String8 result; 1194 1195 bool locked = tryLock(mLock); 1196 if (!locked) { 1197 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1198 write(fd, buffer, strlen(buffer)); 1199 } 1200 1201 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1202 result.append(buffer); 1203 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1204 result.append(buffer); 1205 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1206 result.append(buffer); 1207 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1208 result.append(buffer); 1209 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1210 result.append(buffer); 1211 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1212 result.append(buffer); 1213 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1214 result.append(buffer); 1215 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1216 result.append(buffer); 1217 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1218 result.append(buffer); 1219 1220 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1221 result.append(buffer); 1222 result.append(" Index Command"); 1223 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1224 snprintf(buffer, SIZE, "\n %02d ", i); 1225 result.append(buffer); 1226 result.append(mNewParameters[i]); 1227 } 1228 1229 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1230 result.append(buffer); 1231 snprintf(buffer, SIZE, " Index event param\n"); 1232 result.append(buffer); 1233 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1234 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1235 result.append(buffer); 1236 } 1237 result.append("\n"); 1238 1239 write(fd, result.string(), result.size()); 1240 1241 if (locked) { 1242 mLock.unlock(); 1243 } 1244 return NO_ERROR; 1245} 1246 1247status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1248{ 1249 const size_t SIZE = 256; 1250 char buffer[SIZE]; 1251 String8 result; 1252 1253 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1254 write(fd, buffer, strlen(buffer)); 1255 1256 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1257 sp<EffectChain> chain = mEffectChains[i]; 1258 if (chain != 0) { 1259 chain->dump(fd, args); 1260 } 1261 } 1262 return NO_ERROR; 1263} 1264 1265void AudioFlinger::ThreadBase::acquireWakeLock() 1266{ 1267 Mutex::Autolock _l(mLock); 1268 acquireWakeLock_l(); 1269} 1270 1271void AudioFlinger::ThreadBase::acquireWakeLock_l() 1272{ 1273 if (mPowerManager == 0) { 1274 // use checkService() to avoid blocking if power service is not up yet 1275 sp<IBinder> binder = 1276 defaultServiceManager()->checkService(String16("power")); 1277 if (binder == 0) { 1278 ALOGW("Thread %s cannot connect to the power manager service", mName); 1279 } else { 1280 mPowerManager = interface_cast<IPowerManager>(binder); 1281 binder->linkToDeath(mDeathRecipient); 1282 } 1283 } 1284 if (mPowerManager != 0) { 1285 sp<IBinder> binder = new BBinder(); 1286 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1287 binder, 1288 String16(mName)); 1289 if (status == NO_ERROR) { 1290 mWakeLockToken = binder; 1291 } 1292 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1293 } 1294} 1295 1296void AudioFlinger::ThreadBase::releaseWakeLock() 1297{ 1298 Mutex::Autolock _l(mLock); 1299 releaseWakeLock_l(); 1300} 1301 1302void AudioFlinger::ThreadBase::releaseWakeLock_l() 1303{ 1304 if (mWakeLockToken != 0) { 1305 ALOGV("releaseWakeLock_l() %s", mName); 1306 if (mPowerManager != 0) { 1307 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1308 } 1309 mWakeLockToken.clear(); 1310 } 1311} 1312 1313void AudioFlinger::ThreadBase::clearPowerManager() 1314{ 1315 Mutex::Autolock _l(mLock); 1316 releaseWakeLock_l(); 1317 mPowerManager.clear(); 1318} 1319 1320void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1321{ 1322 sp<ThreadBase> thread = mThread.promote(); 1323 if (thread != 0) { 1324 thread->clearPowerManager(); 1325 } 1326 ALOGW("power manager service died !!!"); 1327} 1328 1329void AudioFlinger::ThreadBase::setEffectSuspended( 1330 const effect_uuid_t *type, bool suspend, int sessionId) 1331{ 1332 Mutex::Autolock _l(mLock); 1333 setEffectSuspended_l(type, suspend, sessionId); 1334} 1335 1336void AudioFlinger::ThreadBase::setEffectSuspended_l( 1337 const effect_uuid_t *type, bool suspend, int sessionId) 1338{ 1339 sp<EffectChain> chain = getEffectChain_l(sessionId); 1340 if (chain != 0) { 1341 if (type != NULL) { 1342 chain->setEffectSuspended_l(type, suspend); 1343 } else { 1344 chain->setEffectSuspendedAll_l(suspend); 1345 } 1346 } 1347 1348 updateSuspendedSessions_l(type, suspend, sessionId); 1349} 1350 1351void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1352{ 1353 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1354 if (index < 0) { 1355 return; 1356 } 1357 1358 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1359 mSuspendedSessions.editValueAt(index); 1360 1361 for (size_t i = 0; i < sessionEffects.size(); i++) { 1362 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1363 for (int j = 0; j < desc->mRefCount; j++) { 1364 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1365 chain->setEffectSuspendedAll_l(true); 1366 } else { 1367 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1368 desc->mType.timeLow); 1369 chain->setEffectSuspended_l(&desc->mType, true); 1370 } 1371 } 1372 } 1373} 1374 1375void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1376 bool suspend, 1377 int sessionId) 1378{ 1379 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1380 1381 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1382 1383 if (suspend) { 1384 if (index >= 0) { 1385 sessionEffects = mSuspendedSessions.editValueAt(index); 1386 } else { 1387 mSuspendedSessions.add(sessionId, sessionEffects); 1388 } 1389 } else { 1390 if (index < 0) { 1391 return; 1392 } 1393 sessionEffects = mSuspendedSessions.editValueAt(index); 1394 } 1395 1396 1397 int key = EffectChain::kKeyForSuspendAll; 1398 if (type != NULL) { 1399 key = type->timeLow; 1400 } 1401 index = sessionEffects.indexOfKey(key); 1402 1403 sp<SuspendedSessionDesc> desc; 1404 if (suspend) { 1405 if (index >= 0) { 1406 desc = sessionEffects.valueAt(index); 1407 } else { 1408 desc = new SuspendedSessionDesc(); 1409 if (type != NULL) { 1410 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1411 } 1412 sessionEffects.add(key, desc); 1413 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1414 } 1415 desc->mRefCount++; 1416 } else { 1417 if (index < 0) { 1418 return; 1419 } 1420 desc = sessionEffects.valueAt(index); 1421 if (--desc->mRefCount == 0) { 1422 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1423 sessionEffects.removeItemsAt(index); 1424 if (sessionEffects.isEmpty()) { 1425 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1426 sessionId); 1427 mSuspendedSessions.removeItem(sessionId); 1428 } 1429 } 1430 } 1431 if (!sessionEffects.isEmpty()) { 1432 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1433 } 1434} 1435 1436void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1437 bool enabled, 1438 int sessionId) 1439{ 1440 Mutex::Autolock _l(mLock); 1441 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1442} 1443 1444void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1445 bool enabled, 1446 int sessionId) 1447{ 1448 if (mType != RECORD) { 1449 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1450 // another session. This gives the priority to well behaved effect control panels 1451 // and applications not using global effects. 1452 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1453 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1454 } 1455 } 1456 1457 sp<EffectChain> chain = getEffectChain_l(sessionId); 1458 if (chain != 0) { 1459 chain->checkSuspendOnEffectEnabled(effect, enabled); 1460 } 1461} 1462 1463// ---------------------------------------------------------------------------- 1464 1465AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1466 AudioStreamOut* output, 1467 audio_io_handle_t id, 1468 uint32_t device, 1469 type_t type) 1470 : ThreadBase(audioFlinger, id, device, type), 1471 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1472 // Assumes constructor is called by AudioFlinger with it's mLock held, 1473 // but it would be safer to explicitly pass initial masterMute as parameter 1474 mMasterMute(audioFlinger->masterMute_l()), 1475 // mStreamTypes[] initialized in constructor body 1476 mOutput(output), 1477 // Assumes constructor is called by AudioFlinger with it's mLock held, 1478 // but it would be safer to explicitly pass initial masterVolume as parameter 1479 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1480 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1481 mMixerStatus(MIXER_IDLE), 1482 mPrevMixerStatus(MIXER_IDLE), 1483 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1484{ 1485 snprintf(mName, kNameLength, "AudioOut_%X", id); 1486 1487 readOutputParameters(); 1488 1489 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1490 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1491 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1492 stream = (audio_stream_type_t) (stream + 1)) { 1493 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1494 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1495 // initialized by stream_type_t default constructor 1496 // mStreamTypes[stream].valid = true; 1497 } 1498 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1499 // because mAudioFlinger doesn't have one to copy from 1500} 1501 1502AudioFlinger::PlaybackThread::~PlaybackThread() 1503{ 1504 delete [] mMixBuffer; 1505} 1506 1507status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1508{ 1509 dumpInternals(fd, args); 1510 dumpTracks(fd, args); 1511 dumpEffectChains(fd, args); 1512 return NO_ERROR; 1513} 1514 1515status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1516{ 1517 const size_t SIZE = 256; 1518 char buffer[SIZE]; 1519 String8 result; 1520 1521 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1522 result.append(buffer); 1523 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1524 for (size_t i = 0; i < mTracks.size(); ++i) { 1525 sp<Track> track = mTracks[i]; 1526 if (track != 0) { 1527 track->dump(buffer, SIZE); 1528 result.append(buffer); 1529 } 1530 } 1531 1532 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1533 result.append(buffer); 1534 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1535 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1536 sp<Track> track = mActiveTracks[i].promote(); 1537 if (track != 0) { 1538 track->dump(buffer, SIZE); 1539 result.append(buffer); 1540 } 1541 } 1542 write(fd, result.string(), result.size()); 1543 return NO_ERROR; 1544} 1545 1546status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1547{ 1548 const size_t SIZE = 256; 1549 char buffer[SIZE]; 1550 String8 result; 1551 1552 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1553 result.append(buffer); 1554 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1555 result.append(buffer); 1556 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1557 result.append(buffer); 1558 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1559 result.append(buffer); 1560 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1561 result.append(buffer); 1562 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1563 result.append(buffer); 1564 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1565 result.append(buffer); 1566 write(fd, result.string(), result.size()); 1567 1568 dumpBase(fd, args); 1569 1570 return NO_ERROR; 1571} 1572 1573// Thread virtuals 1574status_t AudioFlinger::PlaybackThread::readyToRun() 1575{ 1576 status_t status = initCheck(); 1577 if (status == NO_ERROR) { 1578 ALOGI("AudioFlinger's thread %p ready to run", this); 1579 } else { 1580 ALOGE("No working audio driver found."); 1581 } 1582 return status; 1583} 1584 1585void AudioFlinger::PlaybackThread::onFirstRef() 1586{ 1587 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1588} 1589 1590// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1591sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1592 const sp<AudioFlinger::Client>& client, 1593 audio_stream_type_t streamType, 1594 uint32_t sampleRate, 1595 audio_format_t format, 1596 uint32_t channelMask, 1597 int frameCount, 1598 const sp<IMemory>& sharedBuffer, 1599 int sessionId, 1600 bool isTimed, 1601 status_t *status) 1602{ 1603 sp<Track> track; 1604 status_t lStatus; 1605 1606 if (mType == DIRECT) { 1607 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1608 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1609 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1610 "for output %p with format %d", 1611 sampleRate, format, channelMask, mOutput, mFormat); 1612 lStatus = BAD_VALUE; 1613 goto Exit; 1614 } 1615 } 1616 } else { 1617 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1618 if (sampleRate > mSampleRate*2) { 1619 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1620 lStatus = BAD_VALUE; 1621 goto Exit; 1622 } 1623 } 1624 1625 lStatus = initCheck(); 1626 if (lStatus != NO_ERROR) { 1627 ALOGE("Audio driver not initialized."); 1628 goto Exit; 1629 } 1630 1631 { // scope for mLock 1632 Mutex::Autolock _l(mLock); 1633 1634 // all tracks in same audio session must share the same routing strategy otherwise 1635 // conflicts will happen when tracks are moved from one output to another by audio policy 1636 // manager 1637 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1638 for (size_t i = 0; i < mTracks.size(); ++i) { 1639 sp<Track> t = mTracks[i]; 1640 if (t != 0 && !t->isOutputTrack()) { 1641 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1642 if (sessionId == t->sessionId() && strategy != actual) { 1643 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1644 strategy, actual); 1645 lStatus = BAD_VALUE; 1646 goto Exit; 1647 } 1648 } 1649 } 1650 1651 if (!isTimed) { 1652 track = new Track(this, client, streamType, sampleRate, format, 1653 channelMask, frameCount, sharedBuffer, sessionId); 1654 } else { 1655 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1656 channelMask, frameCount, sharedBuffer, sessionId); 1657 } 1658 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1659 lStatus = NO_MEMORY; 1660 goto Exit; 1661 } 1662 mTracks.add(track); 1663 1664 sp<EffectChain> chain = getEffectChain_l(sessionId); 1665 if (chain != 0) { 1666 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1667 track->setMainBuffer(chain->inBuffer()); 1668 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1669 chain->incTrackCnt(); 1670 } 1671 1672 // invalidate track immediately if the stream type was moved to another thread since 1673 // createTrack() was called by the client process. 1674 if (!mStreamTypes[streamType].valid) { 1675 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1676 this, streamType); 1677 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1678 } 1679 } 1680 lStatus = NO_ERROR; 1681 1682Exit: 1683 if (status) { 1684 *status = lStatus; 1685 } 1686 return track; 1687} 1688 1689uint32_t AudioFlinger::PlaybackThread::latency() const 1690{ 1691 Mutex::Autolock _l(mLock); 1692 if (initCheck() == NO_ERROR) { 1693 return mOutput->stream->get_latency(mOutput->stream); 1694 } else { 1695 return 0; 1696 } 1697} 1698 1699void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1700{ 1701 Mutex::Autolock _l(mLock); 1702 mMasterVolume = value; 1703} 1704 1705void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1706{ 1707 Mutex::Autolock _l(mLock); 1708 setMasterMute_l(muted); 1709} 1710 1711void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1712{ 1713 Mutex::Autolock _l(mLock); 1714 mStreamTypes[stream].volume = value; 1715} 1716 1717void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1718{ 1719 Mutex::Autolock _l(mLock); 1720 mStreamTypes[stream].mute = muted; 1721} 1722 1723float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1724{ 1725 Mutex::Autolock _l(mLock); 1726 return mStreamTypes[stream].volume; 1727} 1728 1729// addTrack_l() must be called with ThreadBase::mLock held 1730status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1731{ 1732 status_t status = ALREADY_EXISTS; 1733 1734 // set retry count for buffer fill 1735 track->mRetryCount = kMaxTrackStartupRetries; 1736 if (mActiveTracks.indexOf(track) < 0) { 1737 // the track is newly added, make sure it fills up all its 1738 // buffers before playing. This is to ensure the client will 1739 // effectively get the latency it requested. 1740 track->mFillingUpStatus = Track::FS_FILLING; 1741 track->mResetDone = false; 1742 mActiveTracks.add(track); 1743 if (track->mainBuffer() != mMixBuffer) { 1744 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1745 if (chain != 0) { 1746 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1747 chain->incActiveTrackCnt(); 1748 } 1749 } 1750 1751 status = NO_ERROR; 1752 } 1753 1754 ALOGV("mWaitWorkCV.broadcast"); 1755 mWaitWorkCV.broadcast(); 1756 1757 return status; 1758} 1759 1760// destroyTrack_l() must be called with ThreadBase::mLock held 1761void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1762{ 1763 track->mState = TrackBase::TERMINATED; 1764 if (mActiveTracks.indexOf(track) < 0) { 1765 removeTrack_l(track); 1766 } 1767} 1768 1769void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1770{ 1771 mTracks.remove(track); 1772 deleteTrackName_l(track->name()); 1773 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1774 if (chain != 0) { 1775 chain->decTrackCnt(); 1776 } 1777} 1778 1779String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1780{ 1781 String8 out_s8 = String8(""); 1782 char *s; 1783 1784 Mutex::Autolock _l(mLock); 1785 if (initCheck() != NO_ERROR) { 1786 return out_s8; 1787 } 1788 1789 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1790 out_s8 = String8(s); 1791 free(s); 1792 return out_s8; 1793} 1794 1795// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1796void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1797 AudioSystem::OutputDescriptor desc; 1798 void *param2 = NULL; 1799 1800 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1801 1802 switch (event) { 1803 case AudioSystem::OUTPUT_OPENED: 1804 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1805 desc.channels = mChannelMask; 1806 desc.samplingRate = mSampleRate; 1807 desc.format = mFormat; 1808 desc.frameCount = mFrameCount; 1809 desc.latency = latency(); 1810 param2 = &desc; 1811 break; 1812 1813 case AudioSystem::STREAM_CONFIG_CHANGED: 1814 param2 = ¶m; 1815 case AudioSystem::OUTPUT_CLOSED: 1816 default: 1817 break; 1818 } 1819 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1820} 1821 1822void AudioFlinger::PlaybackThread::readOutputParameters() 1823{ 1824 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1825 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1826 mChannelCount = (uint16_t)popcount(mChannelMask); 1827 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1828 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1829 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1830 1831 // FIXME - Current mixer implementation only supports stereo output: Always 1832 // Allocate a stereo buffer even if HW output is mono. 1833 delete[] mMixBuffer; 1834 mMixBuffer = new int16_t[mFrameCount * 2]; 1835 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1836 1837 // force reconfiguration of effect chains and engines to take new buffer size and audio 1838 // parameters into account 1839 // Note that mLock is not held when readOutputParameters() is called from the constructor 1840 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1841 // matter. 1842 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1843 Vector< sp<EffectChain> > effectChains = mEffectChains; 1844 for (size_t i = 0; i < effectChains.size(); i ++) { 1845 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1846 } 1847} 1848 1849status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1850{ 1851 if (halFrames == NULL || dspFrames == NULL) { 1852 return BAD_VALUE; 1853 } 1854 Mutex::Autolock _l(mLock); 1855 if (initCheck() != NO_ERROR) { 1856 return INVALID_OPERATION; 1857 } 1858 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1859 1860 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1861} 1862 1863uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1864{ 1865 Mutex::Autolock _l(mLock); 1866 uint32_t result = 0; 1867 if (getEffectChain_l(sessionId) != 0) { 1868 result = EFFECT_SESSION; 1869 } 1870 1871 for (size_t i = 0; i < mTracks.size(); ++i) { 1872 sp<Track> track = mTracks[i]; 1873 if (sessionId == track->sessionId() && 1874 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1875 result |= TRACK_SESSION; 1876 break; 1877 } 1878 } 1879 1880 return result; 1881} 1882 1883uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1884{ 1885 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1886 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1887 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1888 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1889 } 1890 for (size_t i = 0; i < mTracks.size(); i++) { 1891 sp<Track> track = mTracks[i]; 1892 if (sessionId == track->sessionId() && 1893 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1894 return AudioSystem::getStrategyForStream(track->streamType()); 1895 } 1896 } 1897 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1898} 1899 1900 1901AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1902{ 1903 Mutex::Autolock _l(mLock); 1904 return mOutput; 1905} 1906 1907AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1908{ 1909 Mutex::Autolock _l(mLock); 1910 AudioStreamOut *output = mOutput; 1911 mOutput = NULL; 1912 return output; 1913} 1914 1915// this method must always be called either with ThreadBase mLock held or inside the thread loop 1916audio_stream_t* AudioFlinger::PlaybackThread::stream() 1917{ 1918 if (mOutput == NULL) { 1919 return NULL; 1920 } 1921 return &mOutput->stream->common; 1922} 1923 1924uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1925{ 1926 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1927 // decoding and transfer time. So sleeping for half of the latency would likely cause 1928 // underruns 1929 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1930 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1931 } else { 1932 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1933 } 1934} 1935 1936// ---------------------------------------------------------------------------- 1937 1938AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1939 audio_io_handle_t id, uint32_t device, type_t type) 1940 : PlaybackThread(audioFlinger, output, id, device, type) 1941{ 1942 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1943 // FIXME - Current mixer implementation only supports stereo output 1944 if (mChannelCount == 1) { 1945 ALOGE("Invalid audio hardware channel count"); 1946 } 1947} 1948 1949AudioFlinger::MixerThread::~MixerThread() 1950{ 1951 delete mAudioMixer; 1952} 1953 1954class CpuStats { 1955public: 1956 CpuStats(); 1957 void sample(const String8 &title); 1958#ifdef DEBUG_CPU_USAGE 1959private: 1960 ThreadCpuUsage mCpuUsage; // instantaneous thread CPU usage in wall clock ns 1961 CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns 1962 1963 CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles 1964 1965 int mCpuNum; // thread's current CPU number 1966 int mCpukHz; // frequency of thread's current CPU in kHz 1967#endif 1968}; 1969 1970CpuStats::CpuStats() 1971#ifdef DEBUG_CPU_USAGE 1972 : mCpuNum(-1), mCpukHz(-1) 1973#endif 1974{ 1975} 1976 1977void CpuStats::sample(const String8 &title) { 1978#ifdef DEBUG_CPU_USAGE 1979 // get current thread's delta CPU time in wall clock ns 1980 double wcNs; 1981 bool valid = mCpuUsage.sampleAndEnable(wcNs); 1982 1983 // record sample for wall clock statistics 1984 if (valid) { 1985 mWcStats.sample(wcNs); 1986 } 1987 1988 // get the current CPU number 1989 int cpuNum = sched_getcpu(); 1990 1991 // get the current CPU frequency in kHz 1992 int cpukHz = mCpuUsage.getCpukHz(cpuNum); 1993 1994 // check if either CPU number or frequency changed 1995 if (cpuNum != mCpuNum || cpukHz != mCpukHz) { 1996 mCpuNum = cpuNum; 1997 mCpukHz = cpukHz; 1998 // ignore sample for purposes of cycles 1999 valid = false; 2000 } 2001 2002 // if no change in CPU number or frequency, then record sample for cycle statistics 2003 if (valid && mCpukHz > 0) { 2004 double cycles = wcNs * cpukHz * 0.000001; 2005 mHzStats.sample(cycles); 2006 } 2007 2008 unsigned n = mWcStats.n(); 2009 // mCpuUsage.elapsed() is expensive, so don't call it every loop 2010 if ((n & 127) == 1) { 2011 long long elapsed = mCpuUsage.elapsed(); 2012 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 2013 double perLoop = elapsed / (double) n; 2014 double perLoop100 = perLoop * 0.01; 2015 double perLoop1k = perLoop * 0.001; 2016 double mean = mWcStats.mean(); 2017 double stddev = mWcStats.stddev(); 2018 double minimum = mWcStats.minimum(); 2019 double maximum = mWcStats.maximum(); 2020 double meanCycles = mHzStats.mean(); 2021 double stddevCycles = mHzStats.stddev(); 2022 double minCycles = mHzStats.minimum(); 2023 double maxCycles = mHzStats.maximum(); 2024 mCpuUsage.resetElapsed(); 2025 mWcStats.reset(); 2026 mHzStats.reset(); 2027 ALOGD("CPU usage for %s over past %.1f secs\n" 2028 " (%u mixer loops at %.1f mean ms per loop):\n" 2029 " us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n" 2030 " %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f\n" 2031 " MHz: mean=%.1f, stddev=%.1f, min=%.1f max=%.1f", 2032 title.string(), 2033 elapsed * .000000001, n, perLoop * .000001, 2034 mean * .001, 2035 stddev * .001, 2036 minimum * .001, 2037 maximum * .001, 2038 mean / perLoop100, 2039 stddev / perLoop100, 2040 minimum / perLoop100, 2041 maximum / perLoop100, 2042 meanCycles / perLoop1k, 2043 stddevCycles / perLoop1k, 2044 minCycles / perLoop1k, 2045 maxCycles / perLoop1k); 2046 2047 } 2048 } 2049#endif 2050}; 2051 2052void AudioFlinger::PlaybackThread::checkSilentMode_l() 2053{ 2054 if (!mMasterMute) { 2055 char value[PROPERTY_VALUE_MAX]; 2056 if (property_get("ro.audio.silent", value, "0") > 0) { 2057 char *endptr; 2058 unsigned long ul = strtoul(value, &endptr, 0); 2059 if (*endptr == '\0' && ul != 0) { 2060 ALOGD("Silence is golden"); 2061 // The setprop command will not allow a property to be changed after 2062 // the first time it is set, so we don't have to worry about un-muting. 2063 setMasterMute_l(true); 2064 } 2065 } 2066 } 2067} 2068 2069bool AudioFlinger::PlaybackThread::threadLoop() 2070{ 2071 Vector< sp<Track> > tracksToRemove; 2072 2073 standbyTime = systemTime(); 2074 2075 // MIXER 2076 nsecs_t lastWarning = 0; 2077if (mType == MIXER) { 2078 longStandbyExit = false; 2079} 2080 2081 // DUPLICATING 2082 // FIXME could this be made local to while loop? 2083 writeFrames = 0; 2084 2085 cacheParameters_l(); 2086 sleepTime = idleSleepTime; 2087 2088if (mType == MIXER) { 2089 sleepTimeShift = 0; 2090} 2091 2092 CpuStats cpuStats; 2093 const String8 myName(String8::format("thread %p type %d TID %d", this, mType, gettid())); 2094 2095 acquireWakeLock(); 2096 2097 while (!exitPending()) 2098 { 2099 cpuStats.sample(myName); 2100 2101 Vector< sp<EffectChain> > effectChains; 2102 2103 processConfigEvents(); 2104 2105 { // scope for mLock 2106 2107 Mutex::Autolock _l(mLock); 2108 2109 if (checkForNewParameters_l()) { 2110 cacheParameters_l(); 2111 } 2112 2113 saveOutputTracks(); 2114 2115 // put audio hardware into standby after short delay 2116 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2117 mSuspended > 0)) { 2118 if (!mStandby) { 2119 2120 threadLoop_standby(); 2121 2122 mStandby = true; 2123 mBytesWritten = 0; 2124 } 2125 2126 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2127 // we're about to wait, flush the binder command buffer 2128 IPCThreadState::self()->flushCommands(); 2129 2130 clearOutputTracks(); 2131 2132 if (exitPending()) break; 2133 2134 releaseWakeLock_l(); 2135 // wait until we have something to do... 2136 ALOGV("%s going to sleep", myName.string()); 2137 mWaitWorkCV.wait(mLock); 2138 ALOGV("%s waking up", myName.string()); 2139 acquireWakeLock_l(); 2140 2141 mPrevMixerStatus = MIXER_IDLE; 2142 2143 checkSilentMode_l(); 2144 2145 standbyTime = systemTime() + standbyDelay; 2146 sleepTime = idleSleepTime; 2147 if (mType == MIXER) { 2148 sleepTimeShift = 0; 2149 } 2150 2151 continue; 2152 } 2153 } 2154 2155 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2156 // Shift in the new status; this could be a queue if it's 2157 // useful to filter the mixer status over several cycles. 2158 mPrevMixerStatus = mMixerStatus; 2159 mMixerStatus = newMixerStatus; 2160 2161 // prevent any changes in effect chain list and in each effect chain 2162 // during mixing and effect process as the audio buffers could be deleted 2163 // or modified if an effect is created or deleted 2164 lockEffectChains_l(effectChains); 2165 } 2166 2167 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2168 threadLoop_mix(); 2169 } else { 2170 threadLoop_sleepTime(); 2171 } 2172 2173 if (mSuspended > 0) { 2174 sleepTime = suspendSleepTimeUs(); 2175 } 2176 2177 // only process effects if we're going to write 2178 if (sleepTime == 0) { 2179 for (size_t i = 0; i < effectChains.size(); i ++) { 2180 effectChains[i]->process_l(); 2181 } 2182 } 2183 2184 // enable changes in effect chain 2185 unlockEffectChains(effectChains); 2186 2187 // sleepTime == 0 means we must write to audio hardware 2188 if (sleepTime == 0) { 2189 2190 threadLoop_write(); 2191 2192if (mType == MIXER) { 2193 // write blocked detection 2194 nsecs_t now = systemTime(); 2195 nsecs_t delta = now - mLastWriteTime; 2196 if (!mStandby && delta > maxPeriod) { 2197 mNumDelayedWrites++; 2198 if ((now - lastWarning) > kWarningThrottleNs) { 2199 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2200 ns2ms(delta), mNumDelayedWrites, this); 2201 lastWarning = now; 2202 } 2203 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2204 // a different threshold. Or completely removed for what it is worth anyway... 2205 if (mStandby) { 2206 longStandbyExit = true; 2207 } 2208 } 2209} 2210 2211 mStandby = false; 2212 } else { 2213 usleep(sleepTime); 2214 } 2215 2216 // finally let go of removed track(s), without the lock held 2217 // since we can't guarantee the destructors won't acquire that 2218 // same lock. 2219 tracksToRemove.clear(); 2220 2221 // FIXME I don't understand the need for this here; 2222 // it was in the original code but maybe the 2223 // assignment in saveOutputTracks() makes this unnecessary? 2224 clearOutputTracks(); 2225 2226 // Effect chains will be actually deleted here if they were removed from 2227 // mEffectChains list during mixing or effects processing 2228 effectChains.clear(); 2229 2230 // FIXME Note that the above .clear() is no longer necessary since effectChains 2231 // is now local to this block, but will keep it for now (at least until merge done). 2232 } 2233 2234if (mType == MIXER || mType == DIRECT) { 2235 // put output stream into standby mode 2236 if (!mStandby) { 2237 mOutput->stream->common.standby(&mOutput->stream->common); 2238 } 2239} 2240if (mType == DUPLICATING) { 2241 // for DuplicatingThread, standby mode is handled by the outputTracks 2242} 2243 2244 releaseWakeLock(); 2245 2246 ALOGV("Thread %p type %d exiting", this, mType); 2247 return false; 2248} 2249 2250// shared by MIXER and DIRECT, overridden by DUPLICATING 2251void AudioFlinger::PlaybackThread::threadLoop_write() 2252{ 2253 // FIXME rewrite to reduce number of system calls 2254 mLastWriteTime = systemTime(); 2255 mInWrite = true; 2256 mBytesWritten += mixBufferSize; 2257 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2258 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2259 mNumWrites++; 2260 mInWrite = false; 2261} 2262 2263// shared by MIXER and DIRECT, overridden by DUPLICATING 2264void AudioFlinger::PlaybackThread::threadLoop_standby() 2265{ 2266 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2267 mOutput->stream->common.standby(&mOutput->stream->common); 2268} 2269 2270void AudioFlinger::MixerThread::threadLoop_mix() 2271{ 2272 // obtain the presentation timestamp of the next output buffer 2273 int64_t pts; 2274 status_t status = INVALID_OPERATION; 2275 2276 if (NULL != mOutput->stream->get_next_write_timestamp) { 2277 status = mOutput->stream->get_next_write_timestamp( 2278 mOutput->stream, &pts); 2279 } 2280 2281 if (status != NO_ERROR) { 2282 pts = AudioBufferProvider::kInvalidPTS; 2283 } 2284 2285 // mix buffers... 2286 mAudioMixer->process(pts); 2287 // increase sleep time progressively when application underrun condition clears. 2288 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2289 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2290 // such that we would underrun the audio HAL. 2291 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2292 sleepTimeShift--; 2293 } 2294 sleepTime = 0; 2295 standbyTime = systemTime() + standbyDelay; 2296 //TODO: delay standby when effects have a tail 2297} 2298 2299void AudioFlinger::MixerThread::threadLoop_sleepTime() 2300{ 2301 // If no tracks are ready, sleep once for the duration of an output 2302 // buffer size, then write 0s to the output 2303 if (sleepTime == 0) { 2304 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2305 sleepTime = activeSleepTime >> sleepTimeShift; 2306 if (sleepTime < kMinThreadSleepTimeUs) { 2307 sleepTime = kMinThreadSleepTimeUs; 2308 } 2309 // reduce sleep time in case of consecutive application underruns to avoid 2310 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2311 // duration we would end up writing less data than needed by the audio HAL if 2312 // the condition persists. 2313 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2314 sleepTimeShift++; 2315 } 2316 } else { 2317 sleepTime = idleSleepTime; 2318 } 2319 } else if (mBytesWritten != 0 || 2320 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2321 memset (mMixBuffer, 0, mixBufferSize); 2322 sleepTime = 0; 2323 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2324 } 2325 // TODO add standby time extension fct of effect tail 2326} 2327 2328// prepareTracks_l() must be called with ThreadBase::mLock held 2329AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2330 Vector< sp<Track> > *tracksToRemove) 2331{ 2332 2333 mixer_state mixerStatus = MIXER_IDLE; 2334 // find out which tracks need to be processed 2335 size_t count = mActiveTracks.size(); 2336 size_t mixedTracks = 0; 2337 size_t tracksWithEffect = 0; 2338 2339 float masterVolume = mMasterVolume; 2340 bool masterMute = mMasterMute; 2341 2342 if (masterMute) { 2343 masterVolume = 0; 2344 } 2345 // Delegate master volume control to effect in output mix effect chain if needed 2346 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2347 if (chain != 0) { 2348 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2349 chain->setVolume_l(&v, &v); 2350 masterVolume = (float)((v + (1 << 23)) >> 24); 2351 chain.clear(); 2352 } 2353 2354 for (size_t i=0 ; i<count ; i++) { 2355 sp<Track> t = mActiveTracks[i].promote(); 2356 if (t == 0) continue; 2357 2358 // this const just means the local variable doesn't change 2359 Track* const track = t.get(); 2360 audio_track_cblk_t* cblk = track->cblk(); 2361 2362 // The first time a track is added we wait 2363 // for all its buffers to be filled before processing it 2364 int name = track->name(); 2365 // make sure that we have enough frames to mix one full buffer. 2366 // enforce this condition only once to enable draining the buffer in case the client 2367 // app does not call stop() and relies on underrun to stop: 2368 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2369 // during last round 2370 uint32_t minFrames = 1; 2371 if (!track->isStopped() && !track->isPausing() && 2372 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2373 if (t->sampleRate() == (int)mSampleRate) { 2374 minFrames = mFrameCount; 2375 } else { 2376 // +1 for rounding and +1 for additional sample needed for interpolation 2377 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2378 // add frames already consumed but not yet released by the resampler 2379 // because cblk->framesReady() will include these frames 2380 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2381 // the minimum track buffer size is normally twice the number of frames necessary 2382 // to fill one buffer and the resampler should not leave more than one buffer worth 2383 // of unreleased frames after each pass, but just in case... 2384 ALOG_ASSERT(minFrames <= cblk->frameCount); 2385 } 2386 } 2387 if ((track->framesReady() >= minFrames) && track->isReady() && 2388 !track->isPaused() && !track->isTerminated()) 2389 { 2390 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2391 2392 mixedTracks++; 2393 2394 // track->mainBuffer() != mMixBuffer means there is an effect chain 2395 // connected to the track 2396 chain.clear(); 2397 if (track->mainBuffer() != mMixBuffer) { 2398 chain = getEffectChain_l(track->sessionId()); 2399 // Delegate volume control to effect in track effect chain if needed 2400 if (chain != 0) { 2401 tracksWithEffect++; 2402 } else { 2403 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2404 name, track->sessionId()); 2405 } 2406 } 2407 2408 2409 int param = AudioMixer::VOLUME; 2410 if (track->mFillingUpStatus == Track::FS_FILLED) { 2411 // no ramp for the first volume setting 2412 track->mFillingUpStatus = Track::FS_ACTIVE; 2413 if (track->mState == TrackBase::RESUMING) { 2414 track->mState = TrackBase::ACTIVE; 2415 param = AudioMixer::RAMP_VOLUME; 2416 } 2417 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2418 } else if (cblk->server != 0) { 2419 // If the track is stopped before the first frame was mixed, 2420 // do not apply ramp 2421 param = AudioMixer::RAMP_VOLUME; 2422 } 2423 2424 // compute volume for this track 2425 uint32_t vl, vr, va; 2426 if (track->isMuted() || track->isPausing() || 2427 mStreamTypes[track->streamType()].mute) { 2428 vl = vr = va = 0; 2429 if (track->isPausing()) { 2430 track->setPaused(); 2431 } 2432 } else { 2433 2434 // read original volumes with volume control 2435 float typeVolume = mStreamTypes[track->streamType()].volume; 2436 float v = masterVolume * typeVolume; 2437 uint32_t vlr = cblk->getVolumeLR(); 2438 vl = vlr & 0xFFFF; 2439 vr = vlr >> 16; 2440 // track volumes come from shared memory, so can't be trusted and must be clamped 2441 if (vl > MAX_GAIN_INT) { 2442 ALOGV("Track left volume out of range: %04X", vl); 2443 vl = MAX_GAIN_INT; 2444 } 2445 if (vr > MAX_GAIN_INT) { 2446 ALOGV("Track right volume out of range: %04X", vr); 2447 vr = MAX_GAIN_INT; 2448 } 2449 // now apply the master volume and stream type volume 2450 vl = (uint32_t)(v * vl) << 12; 2451 vr = (uint32_t)(v * vr) << 12; 2452 // assuming master volume and stream type volume each go up to 1.0, 2453 // vl and vr are now in 8.24 format 2454 2455 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2456 // send level comes from shared memory and so may be corrupt 2457 if (sendLevel > MAX_GAIN_INT) { 2458 ALOGV("Track send level out of range: %04X", sendLevel); 2459 sendLevel = MAX_GAIN_INT; 2460 } 2461 va = (uint32_t)(v * sendLevel); 2462 } 2463 // Delegate volume control to effect in track effect chain if needed 2464 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2465 // Do not ramp volume if volume is controlled by effect 2466 param = AudioMixer::VOLUME; 2467 track->mHasVolumeController = true; 2468 } else { 2469 // force no volume ramp when volume controller was just disabled or removed 2470 // from effect chain to avoid volume spike 2471 if (track->mHasVolumeController) { 2472 param = AudioMixer::VOLUME; 2473 } 2474 track->mHasVolumeController = false; 2475 } 2476 2477 // Convert volumes from 8.24 to 4.12 format 2478 // This additional clamping is needed in case chain->setVolume_l() overshot 2479 vl = (vl + (1 << 11)) >> 12; 2480 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2481 vr = (vr + (1 << 11)) >> 12; 2482 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2483 2484 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2485 2486 // XXX: these things DON'T need to be done each time 2487 mAudioMixer->setBufferProvider(name, track); 2488 mAudioMixer->enable(name); 2489 2490 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2491 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2492 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2493 mAudioMixer->setParameter( 2494 name, 2495 AudioMixer::TRACK, 2496 AudioMixer::FORMAT, (void *)track->format()); 2497 mAudioMixer->setParameter( 2498 name, 2499 AudioMixer::TRACK, 2500 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2501 mAudioMixer->setParameter( 2502 name, 2503 AudioMixer::RESAMPLE, 2504 AudioMixer::SAMPLE_RATE, 2505 (void *)(cblk->sampleRate)); 2506 mAudioMixer->setParameter( 2507 name, 2508 AudioMixer::TRACK, 2509 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2510 mAudioMixer->setParameter( 2511 name, 2512 AudioMixer::TRACK, 2513 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2514 2515 // reset retry count 2516 track->mRetryCount = kMaxTrackRetries; 2517 2518 // If one track is ready, set the mixer ready if: 2519 // - the mixer was not ready during previous round OR 2520 // - no other track is not ready 2521 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2522 mixerStatus != MIXER_TRACKS_ENABLED) { 2523 mixerStatus = MIXER_TRACKS_READY; 2524 } 2525 } else { 2526 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2527 if (track->isStopped()) { 2528 track->reset(); 2529 } 2530 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2531 // We have consumed all the buffers of this track. 2532 // Remove it from the list of active tracks. 2533 tracksToRemove->add(track); 2534 } else { 2535 // No buffers for this track. Give it a few chances to 2536 // fill a buffer, then remove it from active list. 2537 if (--(track->mRetryCount) <= 0) { 2538 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2539 tracksToRemove->add(track); 2540 // indicate to client process that the track was disabled because of underrun 2541 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2542 // If one track is not ready, mark the mixer also not ready if: 2543 // - the mixer was ready during previous round OR 2544 // - no other track is ready 2545 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2546 mixerStatus != MIXER_TRACKS_READY) { 2547 mixerStatus = MIXER_TRACKS_ENABLED; 2548 } 2549 } 2550 mAudioMixer->disable(name); 2551 } 2552 } 2553 2554 // remove all the tracks that need to be... 2555 count = tracksToRemove->size(); 2556 if (CC_UNLIKELY(count)) { 2557 for (size_t i=0 ; i<count ; i++) { 2558 const sp<Track>& track = tracksToRemove->itemAt(i); 2559 mActiveTracks.remove(track); 2560 if (track->mainBuffer() != mMixBuffer) { 2561 chain = getEffectChain_l(track->sessionId()); 2562 if (chain != 0) { 2563 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2564 chain->decActiveTrackCnt(); 2565 } 2566 } 2567 if (track->isTerminated()) { 2568 removeTrack_l(track); 2569 } 2570 } 2571 } 2572 2573 // mix buffer must be cleared if all tracks are connected to an 2574 // effect chain as in this case the mixer will not write to 2575 // mix buffer and track effects will accumulate into it 2576 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2577 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2578 } 2579 2580 return mixerStatus; 2581} 2582 2583/* 2584The derived values that are cached: 2585 - mixBufferSize from frame count * frame size 2586 - activeSleepTime from activeSleepTimeUs() 2587 - idleSleepTime from idleSleepTimeUs() 2588 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2589 - maxPeriod from frame count and sample rate (MIXER only) 2590 2591The parameters that affect these derived values are: 2592 - frame count 2593 - frame size 2594 - sample rate 2595 - device type: A2DP or not 2596 - device latency 2597 - format: PCM or not 2598 - active sleep time 2599 - idle sleep time 2600*/ 2601 2602void AudioFlinger::PlaybackThread::cacheParameters_l() 2603{ 2604 mixBufferSize = mFrameCount * mFrameSize; 2605 activeSleepTime = activeSleepTimeUs(); 2606 idleSleepTime = idleSleepTimeUs(); 2607} 2608 2609void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2610{ 2611 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2612 this, streamType, mTracks.size()); 2613 Mutex::Autolock _l(mLock); 2614 2615 size_t size = mTracks.size(); 2616 for (size_t i = 0; i < size; i++) { 2617 sp<Track> t = mTracks[i]; 2618 if (t->streamType() == streamType) { 2619 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2620 t->mCblk->cv.signal(); 2621 } 2622 } 2623} 2624 2625void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2626{ 2627 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2628 this, streamType, valid); 2629 Mutex::Autolock _l(mLock); 2630 2631 mStreamTypes[streamType].valid = valid; 2632} 2633 2634// getTrackName_l() must be called with ThreadBase::mLock held 2635int AudioFlinger::MixerThread::getTrackName_l() 2636{ 2637 return mAudioMixer->getTrackName(); 2638} 2639 2640// deleteTrackName_l() must be called with ThreadBase::mLock held 2641void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2642{ 2643 ALOGV("remove track (%d) and delete from mixer", name); 2644 mAudioMixer->deleteTrackName(name); 2645} 2646 2647// checkForNewParameters_l() must be called with ThreadBase::mLock held 2648bool AudioFlinger::MixerThread::checkForNewParameters_l() 2649{ 2650 bool reconfig = false; 2651 2652 while (!mNewParameters.isEmpty()) { 2653 status_t status = NO_ERROR; 2654 String8 keyValuePair = mNewParameters[0]; 2655 AudioParameter param = AudioParameter(keyValuePair); 2656 int value; 2657 2658 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2659 reconfig = true; 2660 } 2661 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2662 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2663 status = BAD_VALUE; 2664 } else { 2665 reconfig = true; 2666 } 2667 } 2668 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2669 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2670 status = BAD_VALUE; 2671 } else { 2672 reconfig = true; 2673 } 2674 } 2675 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2676 // do not accept frame count changes if tracks are open as the track buffer 2677 // size depends on frame count and correct behavior would not be guaranteed 2678 // if frame count is changed after track creation 2679 if (!mTracks.isEmpty()) { 2680 status = INVALID_OPERATION; 2681 } else { 2682 reconfig = true; 2683 } 2684 } 2685 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2686#ifdef ADD_BATTERY_DATA 2687 // when changing the audio output device, call addBatteryData to notify 2688 // the change 2689 if ((int)mDevice != value) { 2690 uint32_t params = 0; 2691 // check whether speaker is on 2692 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2693 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2694 } 2695 2696 int deviceWithoutSpeaker 2697 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2698 // check if any other device (except speaker) is on 2699 if (value & deviceWithoutSpeaker ) { 2700 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2701 } 2702 2703 if (params != 0) { 2704 addBatteryData(params); 2705 } 2706 } 2707#endif 2708 2709 // forward device change to effects that have requested to be 2710 // aware of attached audio device. 2711 mDevice = (uint32_t)value; 2712 for (size_t i = 0; i < mEffectChains.size(); i++) { 2713 mEffectChains[i]->setDevice_l(mDevice); 2714 } 2715 } 2716 2717 if (status == NO_ERROR) { 2718 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2719 keyValuePair.string()); 2720 if (!mStandby && status == INVALID_OPERATION) { 2721 mOutput->stream->common.standby(&mOutput->stream->common); 2722 mStandby = true; 2723 mBytesWritten = 0; 2724 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2725 keyValuePair.string()); 2726 } 2727 if (status == NO_ERROR && reconfig) { 2728 delete mAudioMixer; 2729 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2730 mAudioMixer = NULL; 2731 readOutputParameters(); 2732 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2733 for (size_t i = 0; i < mTracks.size() ; i++) { 2734 int name = getTrackName_l(); 2735 if (name < 0) break; 2736 mTracks[i]->mName = name; 2737 // limit track sample rate to 2 x new output sample rate 2738 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2739 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2740 } 2741 } 2742 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2743 } 2744 } 2745 2746 mNewParameters.removeAt(0); 2747 2748 mParamStatus = status; 2749 mParamCond.signal(); 2750 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2751 // already timed out waiting for the status and will never signal the condition. 2752 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2753 } 2754 return reconfig; 2755} 2756 2757status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2758{ 2759 const size_t SIZE = 256; 2760 char buffer[SIZE]; 2761 String8 result; 2762 2763 PlaybackThread::dumpInternals(fd, args); 2764 2765 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2766 result.append(buffer); 2767 write(fd, result.string(), result.size()); 2768 return NO_ERROR; 2769} 2770 2771uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2772{ 2773 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2774} 2775 2776uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2777{ 2778 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2779} 2780 2781void AudioFlinger::MixerThread::cacheParameters_l() 2782{ 2783 PlaybackThread::cacheParameters_l(); 2784 2785 // FIXME: Relaxed timing because of a certain device that can't meet latency 2786 // Should be reduced to 2x after the vendor fixes the driver issue 2787 // increase threshold again due to low power audio mode. The way this warning 2788 // threshold is calculated and its usefulness should be reconsidered anyway. 2789 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2790} 2791 2792// ---------------------------------------------------------------------------- 2793AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2794 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2795 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2796 // mLeftVolFloat, mRightVolFloat 2797 // mLeftVolShort, mRightVolShort 2798{ 2799} 2800 2801AudioFlinger::DirectOutputThread::~DirectOutputThread() 2802{ 2803} 2804 2805AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2806 Vector< sp<Track> > *tracksToRemove 2807) 2808{ 2809 sp<Track> trackToRemove; 2810 2811 mixer_state mixerStatus = MIXER_IDLE; 2812 2813 // find out which tracks need to be processed 2814 if (mActiveTracks.size() != 0) { 2815 sp<Track> t = mActiveTracks[0].promote(); 2816 // The track died recently 2817 if (t == 0) return MIXER_IDLE; 2818 2819 Track* const track = t.get(); 2820 audio_track_cblk_t* cblk = track->cblk(); 2821 2822 // The first time a track is added we wait 2823 // for all its buffers to be filled before processing it 2824 if (cblk->framesReady() && track->isReady() && 2825 !track->isPaused() && !track->isTerminated()) 2826 { 2827 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2828 2829 if (track->mFillingUpStatus == Track::FS_FILLED) { 2830 track->mFillingUpStatus = Track::FS_ACTIVE; 2831 mLeftVolFloat = mRightVolFloat = 0; 2832 mLeftVolShort = mRightVolShort = 0; 2833 if (track->mState == TrackBase::RESUMING) { 2834 track->mState = TrackBase::ACTIVE; 2835 rampVolume = true; 2836 } 2837 } else if (cblk->server != 0) { 2838 // If the track is stopped before the first frame was mixed, 2839 // do not apply ramp 2840 rampVolume = true; 2841 } 2842 // compute volume for this track 2843 float left, right; 2844 if (track->isMuted() || mMasterMute || track->isPausing() || 2845 mStreamTypes[track->streamType()].mute) { 2846 left = right = 0; 2847 if (track->isPausing()) { 2848 track->setPaused(); 2849 } 2850 } else { 2851 float typeVolume = mStreamTypes[track->streamType()].volume; 2852 float v = mMasterVolume * typeVolume; 2853 uint32_t vlr = cblk->getVolumeLR(); 2854 float v_clamped = v * (vlr & 0xFFFF); 2855 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2856 left = v_clamped/MAX_GAIN; 2857 v_clamped = v * (vlr >> 16); 2858 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2859 right = v_clamped/MAX_GAIN; 2860 } 2861 2862 if (left != mLeftVolFloat || right != mRightVolFloat) { 2863 mLeftVolFloat = left; 2864 mRightVolFloat = right; 2865 2866 // If audio HAL implements volume control, 2867 // force software volume to nominal value 2868 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2869 left = 1.0f; 2870 right = 1.0f; 2871 } 2872 2873 // Convert volumes from float to 8.24 2874 uint32_t vl = (uint32_t)(left * (1 << 24)); 2875 uint32_t vr = (uint32_t)(right * (1 << 24)); 2876 2877 // Delegate volume control to effect in track effect chain if needed 2878 // only one effect chain can be present on DirectOutputThread, so if 2879 // there is one, the track is connected to it 2880 if (!mEffectChains.isEmpty()) { 2881 // Do not ramp volume if volume is controlled by effect 2882 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2883 rampVolume = false; 2884 } 2885 } 2886 2887 // Convert volumes from 8.24 to 4.12 format 2888 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2889 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2890 leftVol = (uint16_t)v_clamped; 2891 v_clamped = (vr + (1 << 11)) >> 12; 2892 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2893 rightVol = (uint16_t)v_clamped; 2894 } else { 2895 leftVol = mLeftVolShort; 2896 rightVol = mRightVolShort; 2897 rampVolume = false; 2898 } 2899 2900 // reset retry count 2901 track->mRetryCount = kMaxTrackRetriesDirect; 2902 mActiveTrack = t; 2903 mixerStatus = MIXER_TRACKS_READY; 2904 } else { 2905 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2906 if (track->isStopped()) { 2907 track->reset(); 2908 } 2909 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2910 // We have consumed all the buffers of this track. 2911 // Remove it from the list of active tracks. 2912 trackToRemove = track; 2913 } else { 2914 // No buffers for this track. Give it a few chances to 2915 // fill a buffer, then remove it from active list. 2916 if (--(track->mRetryCount) <= 0) { 2917 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2918 trackToRemove = track; 2919 } else { 2920 mixerStatus = MIXER_TRACKS_ENABLED; 2921 } 2922 } 2923 } 2924 } 2925 2926 // FIXME merge this with similar code for removing multiple tracks 2927 // remove all the tracks that need to be... 2928 if (CC_UNLIKELY(trackToRemove != 0)) { 2929 tracksToRemove->add(trackToRemove); 2930 mActiveTracks.remove(trackToRemove); 2931 if (!mEffectChains.isEmpty()) { 2932 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2933 trackToRemove->sessionId()); 2934 mEffectChains[0]->decActiveTrackCnt(); 2935 } 2936 if (trackToRemove->isTerminated()) { 2937 removeTrack_l(trackToRemove); 2938 } 2939 } 2940 2941 return mixerStatus; 2942} 2943 2944void AudioFlinger::DirectOutputThread::threadLoop_mix() 2945{ 2946 AudioBufferProvider::Buffer buffer; 2947 size_t frameCount = mFrameCount; 2948 int8_t *curBuf = (int8_t *)mMixBuffer; 2949 // output audio to hardware 2950 while (frameCount) { 2951 buffer.frameCount = frameCount; 2952 mActiveTrack->getNextBuffer(&buffer); 2953 if (CC_UNLIKELY(buffer.raw == NULL)) { 2954 memset(curBuf, 0, frameCount * mFrameSize); 2955 break; 2956 } 2957 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2958 frameCount -= buffer.frameCount; 2959 curBuf += buffer.frameCount * mFrameSize; 2960 mActiveTrack->releaseBuffer(&buffer); 2961 } 2962 sleepTime = 0; 2963 standbyTime = systemTime() + standbyDelay; 2964 mActiveTrack.clear(); 2965 2966 // apply volume 2967 2968 // Do not apply volume on compressed audio 2969 if (!audio_is_linear_pcm(mFormat)) { 2970 return; 2971 } 2972 2973 // convert to signed 16 bit before volume calculation 2974 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2975 size_t count = mFrameCount * mChannelCount; 2976 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2977 int16_t *dst = mMixBuffer + count-1; 2978 while (count--) { 2979 *dst-- = (int16_t)(*src--^0x80) << 8; 2980 } 2981 } 2982 2983 frameCount = mFrameCount; 2984 int16_t *out = mMixBuffer; 2985 if (rampVolume) { 2986 if (mChannelCount == 1) { 2987 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2988 int32_t vlInc = d / (int32_t)frameCount; 2989 int32_t vl = ((int32_t)mLeftVolShort << 16); 2990 do { 2991 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2992 out++; 2993 vl += vlInc; 2994 } while (--frameCount); 2995 2996 } else { 2997 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2998 int32_t vlInc = d / (int32_t)frameCount; 2999 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 3000 int32_t vrInc = d / (int32_t)frameCount; 3001 int32_t vl = ((int32_t)mLeftVolShort << 16); 3002 int32_t vr = ((int32_t)mRightVolShort << 16); 3003 do { 3004 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 3005 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 3006 out += 2; 3007 vl += vlInc; 3008 vr += vrInc; 3009 } while (--frameCount); 3010 } 3011 } else { 3012 if (mChannelCount == 1) { 3013 do { 3014 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3015 out++; 3016 } while (--frameCount); 3017 } else { 3018 do { 3019 out[0] = clamp16(mul(out[0], leftVol) >> 12); 3020 out[1] = clamp16(mul(out[1], rightVol) >> 12); 3021 out += 2; 3022 } while (--frameCount); 3023 } 3024 } 3025 3026 // convert back to unsigned 8 bit after volume calculation 3027 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 3028 size_t count = mFrameCount * mChannelCount; 3029 int16_t *src = mMixBuffer; 3030 uint8_t *dst = (uint8_t *)mMixBuffer; 3031 while (count--) { 3032 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 3033 } 3034 } 3035 3036 mLeftVolShort = leftVol; 3037 mRightVolShort = rightVol; 3038} 3039 3040void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 3041{ 3042 if (sleepTime == 0) { 3043 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3044 sleepTime = activeSleepTime; 3045 } else { 3046 sleepTime = idleSleepTime; 3047 } 3048 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3049 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3050 sleepTime = 0; 3051 } 3052} 3053 3054// getTrackName_l() must be called with ThreadBase::mLock held 3055int AudioFlinger::DirectOutputThread::getTrackName_l() 3056{ 3057 return 0; 3058} 3059 3060// deleteTrackName_l() must be called with ThreadBase::mLock held 3061void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3062{ 3063} 3064 3065// checkForNewParameters_l() must be called with ThreadBase::mLock held 3066bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3067{ 3068 bool reconfig = false; 3069 3070 while (!mNewParameters.isEmpty()) { 3071 status_t status = NO_ERROR; 3072 String8 keyValuePair = mNewParameters[0]; 3073 AudioParameter param = AudioParameter(keyValuePair); 3074 int value; 3075 3076 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3077 // do not accept frame count changes if tracks are open as the track buffer 3078 // size depends on frame count and correct behavior would not be garantied 3079 // if frame count is changed after track creation 3080 if (!mTracks.isEmpty()) { 3081 status = INVALID_OPERATION; 3082 } else { 3083 reconfig = true; 3084 } 3085 } 3086 if (status == NO_ERROR) { 3087 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3088 keyValuePair.string()); 3089 if (!mStandby && status == INVALID_OPERATION) { 3090 mOutput->stream->common.standby(&mOutput->stream->common); 3091 mStandby = true; 3092 mBytesWritten = 0; 3093 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3094 keyValuePair.string()); 3095 } 3096 if (status == NO_ERROR && reconfig) { 3097 readOutputParameters(); 3098 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3099 } 3100 } 3101 3102 mNewParameters.removeAt(0); 3103 3104 mParamStatus = status; 3105 mParamCond.signal(); 3106 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3107 // already timed out waiting for the status and will never signal the condition. 3108 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3109 } 3110 return reconfig; 3111} 3112 3113uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3114{ 3115 uint32_t time; 3116 if (audio_is_linear_pcm(mFormat)) { 3117 time = PlaybackThread::activeSleepTimeUs(); 3118 } else { 3119 time = 10000; 3120 } 3121 return time; 3122} 3123 3124uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3125{ 3126 uint32_t time; 3127 if (audio_is_linear_pcm(mFormat)) { 3128 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3129 } else { 3130 time = 10000; 3131 } 3132 return time; 3133} 3134 3135uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3136{ 3137 uint32_t time; 3138 if (audio_is_linear_pcm(mFormat)) { 3139 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3140 } else { 3141 time = 10000; 3142 } 3143 return time; 3144} 3145 3146void AudioFlinger::DirectOutputThread::cacheParameters_l() 3147{ 3148 PlaybackThread::cacheParameters_l(); 3149 3150 // use shorter standby delay as on normal output to release 3151 // hardware resources as soon as possible 3152 standbyDelay = microseconds(activeSleepTime*2); 3153} 3154 3155// ---------------------------------------------------------------------------- 3156 3157AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3158 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3159 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3160 mWaitTimeMs(UINT_MAX) 3161{ 3162 addOutputTrack(mainThread); 3163} 3164 3165AudioFlinger::DuplicatingThread::~DuplicatingThread() 3166{ 3167 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3168 mOutputTracks[i]->destroy(); 3169 } 3170} 3171 3172void AudioFlinger::DuplicatingThread::threadLoop_mix() 3173{ 3174 // mix buffers... 3175 if (outputsReady(outputTracks)) { 3176 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3177 } else { 3178 memset(mMixBuffer, 0, mixBufferSize); 3179 } 3180 sleepTime = 0; 3181 writeFrames = mFrameCount; 3182} 3183 3184void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3185{ 3186 if (sleepTime == 0) { 3187 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3188 sleepTime = activeSleepTime; 3189 } else { 3190 sleepTime = idleSleepTime; 3191 } 3192 } else if (mBytesWritten != 0) { 3193 // flush remaining overflow buffers in output tracks 3194 for (size_t i = 0; i < outputTracks.size(); i++) { 3195 if (outputTracks[i]->isActive()) { 3196 sleepTime = 0; 3197 writeFrames = 0; 3198 memset(mMixBuffer, 0, mixBufferSize); 3199 break; 3200 } 3201 } 3202 } 3203} 3204 3205void AudioFlinger::DuplicatingThread::threadLoop_write() 3206{ 3207 standbyTime = systemTime() + standbyDelay; 3208 for (size_t i = 0; i < outputTracks.size(); i++) { 3209 outputTracks[i]->write(mMixBuffer, writeFrames); 3210 } 3211 mBytesWritten += mixBufferSize; 3212} 3213 3214void AudioFlinger::DuplicatingThread::threadLoop_standby() 3215{ 3216 // DuplicatingThread implements standby by stopping all tracks 3217 for (size_t i = 0; i < outputTracks.size(); i++) { 3218 outputTracks[i]->stop(); 3219 } 3220} 3221 3222void AudioFlinger::DuplicatingThread::saveOutputTracks() 3223{ 3224 outputTracks = mOutputTracks; 3225} 3226 3227void AudioFlinger::DuplicatingThread::clearOutputTracks() 3228{ 3229 outputTracks.clear(); 3230} 3231 3232void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3233{ 3234 Mutex::Autolock _l(mLock); 3235 // FIXME explain this formula 3236 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3237 OutputTrack *outputTrack = new OutputTrack(thread, 3238 this, 3239 mSampleRate, 3240 mFormat, 3241 mChannelMask, 3242 frameCount); 3243 if (outputTrack->cblk() != NULL) { 3244 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3245 mOutputTracks.add(outputTrack); 3246 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3247 updateWaitTime_l(); 3248 } 3249} 3250 3251void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3252{ 3253 Mutex::Autolock _l(mLock); 3254 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3255 if (mOutputTracks[i]->thread() == thread) { 3256 mOutputTracks[i]->destroy(); 3257 mOutputTracks.removeAt(i); 3258 updateWaitTime_l(); 3259 return; 3260 } 3261 } 3262 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3263} 3264 3265// caller must hold mLock 3266void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3267{ 3268 mWaitTimeMs = UINT_MAX; 3269 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3270 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3271 if (strong != 0) { 3272 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3273 if (waitTimeMs < mWaitTimeMs) { 3274 mWaitTimeMs = waitTimeMs; 3275 } 3276 } 3277 } 3278} 3279 3280 3281bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3282{ 3283 for (size_t i = 0; i < outputTracks.size(); i++) { 3284 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3285 if (thread == 0) { 3286 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3287 return false; 3288 } 3289 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3290 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3291 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3292 return false; 3293 } 3294 } 3295 return true; 3296} 3297 3298uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3299{ 3300 return (mWaitTimeMs * 1000) / 2; 3301} 3302 3303void AudioFlinger::DuplicatingThread::cacheParameters_l() 3304{ 3305 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3306 updateWaitTime_l(); 3307 3308 MixerThread::cacheParameters_l(); 3309} 3310 3311// ---------------------------------------------------------------------------- 3312 3313// TrackBase constructor must be called with AudioFlinger::mLock held 3314AudioFlinger::ThreadBase::TrackBase::TrackBase( 3315 ThreadBase *thread, 3316 const sp<Client>& client, 3317 uint32_t sampleRate, 3318 audio_format_t format, 3319 uint32_t channelMask, 3320 int frameCount, 3321 const sp<IMemory>& sharedBuffer, 3322 int sessionId) 3323 : RefBase(), 3324 mThread(thread), 3325 mClient(client), 3326 mCblk(NULL), 3327 // mBuffer 3328 // mBufferEnd 3329 mFrameCount(0), 3330 mState(IDLE), 3331 mFormat(format), 3332 mStepServerFailed(false), 3333 mSessionId(sessionId) 3334 // mChannelCount 3335 // mChannelMask 3336{ 3337 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3338 3339 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3340 size_t size = sizeof(audio_track_cblk_t); 3341 uint8_t channelCount = popcount(channelMask); 3342 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3343 if (sharedBuffer == 0) { 3344 size += bufferSize; 3345 } 3346 3347 if (client != NULL) { 3348 mCblkMemory = client->heap()->allocate(size); 3349 if (mCblkMemory != 0) { 3350 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3351 if (mCblk != NULL) { // construct the shared structure in-place. 3352 new(mCblk) audio_track_cblk_t(); 3353 // clear all buffers 3354 mCblk->frameCount = frameCount; 3355 mCblk->sampleRate = sampleRate; 3356 mChannelCount = channelCount; 3357 mChannelMask = channelMask; 3358 if (sharedBuffer == 0) { 3359 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3360 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3361 // Force underrun condition to avoid false underrun callback until first data is 3362 // written to buffer (other flags are cleared) 3363 mCblk->flags = CBLK_UNDERRUN_ON; 3364 } else { 3365 mBuffer = sharedBuffer->pointer(); 3366 } 3367 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3368 } 3369 } else { 3370 ALOGE("not enough memory for AudioTrack size=%u", size); 3371 client->heap()->dump("AudioTrack"); 3372 return; 3373 } 3374 } else { 3375 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3376 // construct the shared structure in-place. 3377 new(mCblk) audio_track_cblk_t(); 3378 // clear all buffers 3379 mCblk->frameCount = frameCount; 3380 mCblk->sampleRate = sampleRate; 3381 mChannelCount = channelCount; 3382 mChannelMask = channelMask; 3383 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3384 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3385 // Force underrun condition to avoid false underrun callback until first data is 3386 // written to buffer (other flags are cleared) 3387 mCblk->flags = CBLK_UNDERRUN_ON; 3388 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3389 } 3390} 3391 3392AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3393{ 3394 if (mCblk != NULL) { 3395 if (mClient == 0) { 3396 delete mCblk; 3397 } else { 3398 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3399 } 3400 } 3401 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3402 if (mClient != 0) { 3403 // Client destructor must run with AudioFlinger mutex locked 3404 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3405 // If the client's reference count drops to zero, the associated destructor 3406 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3407 // relying on the automatic clear() at end of scope. 3408 mClient.clear(); 3409 } 3410} 3411 3412// AudioBufferProvider interface 3413// getNextBuffer() = 0; 3414// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3415void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3416{ 3417 buffer->raw = NULL; 3418 mFrameCount = buffer->frameCount; 3419 (void) step(); // ignore return value of step() 3420 buffer->frameCount = 0; 3421} 3422 3423bool AudioFlinger::ThreadBase::TrackBase::step() { 3424 bool result; 3425 audio_track_cblk_t* cblk = this->cblk(); 3426 3427 result = cblk->stepServer(mFrameCount); 3428 if (!result) { 3429 ALOGV("stepServer failed acquiring cblk mutex"); 3430 mStepServerFailed = true; 3431 } 3432 return result; 3433} 3434 3435void AudioFlinger::ThreadBase::TrackBase::reset() { 3436 audio_track_cblk_t* cblk = this->cblk(); 3437 3438 cblk->user = 0; 3439 cblk->server = 0; 3440 cblk->userBase = 0; 3441 cblk->serverBase = 0; 3442 mStepServerFailed = false; 3443 ALOGV("TrackBase::reset"); 3444} 3445 3446int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3447 return (int)mCblk->sampleRate; 3448} 3449 3450void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3451 audio_track_cblk_t* cblk = this->cblk(); 3452 size_t frameSize = cblk->frameSize; 3453 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3454 int8_t *bufferEnd = bufferStart + frames * frameSize; 3455 3456 // Check validity of returned pointer in case the track control block would have been corrupted. 3457 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3458 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3459 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3460 server %d, serverBase %d, user %d, userBase %d", 3461 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3462 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3463 return NULL; 3464 } 3465 3466 return bufferStart; 3467} 3468 3469// ---------------------------------------------------------------------------- 3470 3471// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3472AudioFlinger::PlaybackThread::Track::Track( 3473 PlaybackThread *thread, 3474 const sp<Client>& client, 3475 audio_stream_type_t streamType, 3476 uint32_t sampleRate, 3477 audio_format_t format, 3478 uint32_t channelMask, 3479 int frameCount, 3480 const sp<IMemory>& sharedBuffer, 3481 int sessionId) 3482 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3483 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3484 mAuxEffectId(0), mHasVolumeController(false) 3485{ 3486 if (mCblk != NULL) { 3487 if (thread != NULL) { 3488 mName = thread->getTrackName_l(); 3489 mMainBuffer = thread->mixBuffer(); 3490 } 3491 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3492 if (mName < 0) { 3493 ALOGE("no more track names available"); 3494 } 3495 mStreamType = streamType; 3496 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3497 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3498 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3499 } 3500} 3501 3502AudioFlinger::PlaybackThread::Track::~Track() 3503{ 3504 ALOGV("PlaybackThread::Track destructor"); 3505 sp<ThreadBase> thread = mThread.promote(); 3506 if (thread != 0) { 3507 Mutex::Autolock _l(thread->mLock); 3508 mState = TERMINATED; 3509 } 3510} 3511 3512void AudioFlinger::PlaybackThread::Track::destroy() 3513{ 3514 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3515 // by removing it from mTracks vector, so there is a risk that this Tracks's 3516 // destructor is called. As the destructor needs to lock mLock, 3517 // we must acquire a strong reference on this Track before locking mLock 3518 // here so that the destructor is called only when exiting this function. 3519 // On the other hand, as long as Track::destroy() is only called by 3520 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3521 // this Track with its member mTrack. 3522 sp<Track> keep(this); 3523 { // scope for mLock 3524 sp<ThreadBase> thread = mThread.promote(); 3525 if (thread != 0) { 3526 if (!isOutputTrack()) { 3527 if (mState == ACTIVE || mState == RESUMING) { 3528 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3529 3530#ifdef ADD_BATTERY_DATA 3531 // to track the speaker usage 3532 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3533#endif 3534 } 3535 AudioSystem::releaseOutput(thread->id()); 3536 } 3537 Mutex::Autolock _l(thread->mLock); 3538 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3539 playbackThread->destroyTrack_l(this); 3540 } 3541 } 3542} 3543 3544void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3545{ 3546 uint32_t vlr = mCblk->getVolumeLR(); 3547 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", 3548 mName - AudioMixer::TRACK0, 3549 (mClient == 0) ? getpid_cached : mClient->pid(), 3550 mStreamType, 3551 mFormat, 3552 mChannelMask, 3553 mSessionId, 3554 mFrameCount, 3555 mState, 3556 mMute, 3557 mFillingUpStatus, 3558 mCblk->sampleRate, 3559 vlr & 0xFFFF, 3560 vlr >> 16, 3561 mCblk->server, 3562 mCblk->user, 3563 (int)mMainBuffer, 3564 (int)mAuxBuffer); 3565} 3566 3567// AudioBufferProvider interface 3568status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3569 AudioBufferProvider::Buffer* buffer, int64_t pts) 3570{ 3571 audio_track_cblk_t* cblk = this->cblk(); 3572 uint32_t framesReady; 3573 uint32_t framesReq = buffer->frameCount; 3574 3575 // Check if last stepServer failed, try to step now 3576 if (mStepServerFailed) { 3577 if (!step()) goto getNextBuffer_exit; 3578 ALOGV("stepServer recovered"); 3579 mStepServerFailed = false; 3580 } 3581 3582 framesReady = cblk->framesReady(); 3583 3584 if (CC_LIKELY(framesReady)) { 3585 uint32_t s = cblk->server; 3586 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3587 3588 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3589 if (framesReq > framesReady) { 3590 framesReq = framesReady; 3591 } 3592 if (s + framesReq > bufferEnd) { 3593 framesReq = bufferEnd - s; 3594 } 3595 3596 buffer->raw = getBuffer(s, framesReq); 3597 if (buffer->raw == NULL) goto getNextBuffer_exit; 3598 3599 buffer->frameCount = framesReq; 3600 return NO_ERROR; 3601 } 3602 3603getNextBuffer_exit: 3604 buffer->raw = NULL; 3605 buffer->frameCount = 0; 3606 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3607 return NOT_ENOUGH_DATA; 3608} 3609 3610uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3611 return mCblk->framesReady(); 3612} 3613 3614bool AudioFlinger::PlaybackThread::Track::isReady() const { 3615 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3616 3617 if (framesReady() >= mCblk->frameCount || 3618 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3619 mFillingUpStatus = FS_FILLED; 3620 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3621 return true; 3622 } 3623 return false; 3624} 3625 3626status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3627{ 3628 status_t status = NO_ERROR; 3629 ALOGV("start(%d), calling pid %d session %d tid %d", 3630 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3631 sp<ThreadBase> thread = mThread.promote(); 3632 if (thread != 0) { 3633 Mutex::Autolock _l(thread->mLock); 3634 track_state state = mState; 3635 // here the track could be either new, or restarted 3636 // in both cases "unstop" the track 3637 if (mState == PAUSED) { 3638 mState = TrackBase::RESUMING; 3639 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3640 } else { 3641 mState = TrackBase::ACTIVE; 3642 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3643 } 3644 3645 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3646 thread->mLock.unlock(); 3647 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3648 thread->mLock.lock(); 3649 3650#ifdef ADD_BATTERY_DATA 3651 // to track the speaker usage 3652 if (status == NO_ERROR) { 3653 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3654 } 3655#endif 3656 } 3657 if (status == NO_ERROR) { 3658 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3659 playbackThread->addTrack_l(this); 3660 } else { 3661 mState = state; 3662 } 3663 } else { 3664 status = BAD_VALUE; 3665 } 3666 return status; 3667} 3668 3669void AudioFlinger::PlaybackThread::Track::stop() 3670{ 3671 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3672 sp<ThreadBase> thread = mThread.promote(); 3673 if (thread != 0) { 3674 Mutex::Autolock _l(thread->mLock); 3675 track_state state = mState; 3676 if (mState > STOPPED) { 3677 mState = STOPPED; 3678 // If the track is not active (PAUSED and buffers full), flush buffers 3679 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3680 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3681 reset(); 3682 } 3683 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3684 } 3685 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3686 thread->mLock.unlock(); 3687 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3688 thread->mLock.lock(); 3689 3690#ifdef ADD_BATTERY_DATA 3691 // to track the speaker usage 3692 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3693#endif 3694 } 3695 } 3696} 3697 3698void AudioFlinger::PlaybackThread::Track::pause() 3699{ 3700 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3701 sp<ThreadBase> thread = mThread.promote(); 3702 if (thread != 0) { 3703 Mutex::Autolock _l(thread->mLock); 3704 if (mState == ACTIVE || mState == RESUMING) { 3705 mState = PAUSING; 3706 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3707 if (!isOutputTrack()) { 3708 thread->mLock.unlock(); 3709 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3710 thread->mLock.lock(); 3711 3712#ifdef ADD_BATTERY_DATA 3713 // to track the speaker usage 3714 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3715#endif 3716 } 3717 } 3718 } 3719} 3720 3721void AudioFlinger::PlaybackThread::Track::flush() 3722{ 3723 ALOGV("flush(%d)", mName); 3724 sp<ThreadBase> thread = mThread.promote(); 3725 if (thread != 0) { 3726 Mutex::Autolock _l(thread->mLock); 3727 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3728 return; 3729 } 3730 // No point remaining in PAUSED state after a flush => go to 3731 // STOPPED state 3732 mState = STOPPED; 3733 3734 // do not reset the track if it is still in the process of being stopped or paused. 3735 // this will be done by prepareTracks_l() when the track is stopped. 3736 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3737 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3738 reset(); 3739 } 3740 } 3741} 3742 3743void AudioFlinger::PlaybackThread::Track::reset() 3744{ 3745 // Do not reset twice to avoid discarding data written just after a flush and before 3746 // the audioflinger thread detects the track is stopped. 3747 if (!mResetDone) { 3748 TrackBase::reset(); 3749 // Force underrun condition to avoid false underrun callback until first data is 3750 // written to buffer 3751 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3752 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3753 mFillingUpStatus = FS_FILLING; 3754 mResetDone = true; 3755 } 3756} 3757 3758void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3759{ 3760 mMute = muted; 3761} 3762 3763status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3764{ 3765 status_t status = DEAD_OBJECT; 3766 sp<ThreadBase> thread = mThread.promote(); 3767 if (thread != 0) { 3768 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3769 status = playbackThread->attachAuxEffect(this, EffectId); 3770 } 3771 return status; 3772} 3773 3774void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3775{ 3776 mAuxEffectId = EffectId; 3777 mAuxBuffer = buffer; 3778} 3779 3780// timed audio tracks 3781 3782sp<AudioFlinger::PlaybackThread::TimedTrack> 3783AudioFlinger::PlaybackThread::TimedTrack::create( 3784 PlaybackThread *thread, 3785 const sp<Client>& client, 3786 audio_stream_type_t streamType, 3787 uint32_t sampleRate, 3788 audio_format_t format, 3789 uint32_t channelMask, 3790 int frameCount, 3791 const sp<IMemory>& sharedBuffer, 3792 int sessionId) { 3793 if (!client->reserveTimedTrack()) 3794 return NULL; 3795 3796 sp<TimedTrack> track = new TimedTrack( 3797 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3798 sharedBuffer, sessionId); 3799 3800 if (track == NULL) { 3801 client->releaseTimedTrack(); 3802 return NULL; 3803 } 3804 3805 return track; 3806} 3807 3808AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3809 PlaybackThread *thread, 3810 const sp<Client>& client, 3811 audio_stream_type_t streamType, 3812 uint32_t sampleRate, 3813 audio_format_t format, 3814 uint32_t channelMask, 3815 int frameCount, 3816 const sp<IMemory>& sharedBuffer, 3817 int sessionId) 3818 : Track(thread, client, streamType, sampleRate, format, channelMask, 3819 frameCount, sharedBuffer, sessionId), 3820 mTimedSilenceBuffer(NULL), 3821 mTimedSilenceBufferSize(0), 3822 mTimedAudioOutputOnTime(false), 3823 mMediaTimeTransformValid(false) 3824{ 3825 LocalClock lc; 3826 mLocalTimeFreq = lc.getLocalFreq(); 3827 3828 mLocalTimeToSampleTransform.a_zero = 0; 3829 mLocalTimeToSampleTransform.b_zero = 0; 3830 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3831 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3832 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3833 &mLocalTimeToSampleTransform.a_to_b_denom); 3834} 3835 3836AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3837 mClient->releaseTimedTrack(); 3838 delete [] mTimedSilenceBuffer; 3839} 3840 3841status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3842 size_t size, sp<IMemory>* buffer) { 3843 3844 Mutex::Autolock _l(mTimedBufferQueueLock); 3845 3846 trimTimedBufferQueue_l(); 3847 3848 // lazily initialize the shared memory heap for timed buffers 3849 if (mTimedMemoryDealer == NULL) { 3850 const int kTimedBufferHeapSize = 512 << 10; 3851 3852 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3853 "AudioFlingerTimed"); 3854 if (mTimedMemoryDealer == NULL) 3855 return NO_MEMORY; 3856 } 3857 3858 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3859 if (newBuffer == NULL) { 3860 newBuffer = mTimedMemoryDealer->allocate(size); 3861 if (newBuffer == NULL) 3862 return NO_MEMORY; 3863 } 3864 3865 *buffer = newBuffer; 3866 return NO_ERROR; 3867} 3868 3869// caller must hold mTimedBufferQueueLock 3870void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3871 int64_t mediaTimeNow; 3872 { 3873 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3874 if (!mMediaTimeTransformValid) 3875 return; 3876 3877 int64_t targetTimeNow; 3878 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3879 ? mCCHelper.getCommonTime(&targetTimeNow) 3880 : mCCHelper.getLocalTime(&targetTimeNow); 3881 3882 if (OK != res) 3883 return; 3884 3885 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3886 &mediaTimeNow)) { 3887 return; 3888 } 3889 } 3890 3891 size_t trimIndex; 3892 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3893 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3894 break; 3895 } 3896 3897 if (trimIndex) { 3898 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3899 } 3900} 3901 3902status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3903 const sp<IMemory>& buffer, int64_t pts) { 3904 3905 { 3906 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3907 if (!mMediaTimeTransformValid) 3908 return INVALID_OPERATION; 3909 } 3910 3911 Mutex::Autolock _l(mTimedBufferQueueLock); 3912 3913 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3914 3915 return NO_ERROR; 3916} 3917 3918status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3919 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3920 3921 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3922 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3923 target); 3924 3925 if (!(target == TimedAudioTrack::LOCAL_TIME || 3926 target == TimedAudioTrack::COMMON_TIME)) { 3927 return BAD_VALUE; 3928 } 3929 3930 Mutex::Autolock lock(mMediaTimeTransformLock); 3931 mMediaTimeTransform = xform; 3932 mMediaTimeTransformTarget = target; 3933 mMediaTimeTransformValid = true; 3934 3935 return NO_ERROR; 3936} 3937 3938#define min(a, b) ((a) < (b) ? (a) : (b)) 3939 3940// implementation of getNextBuffer for tracks whose buffers have timestamps 3941status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3942 AudioBufferProvider::Buffer* buffer, int64_t pts) 3943{ 3944 if (pts == AudioBufferProvider::kInvalidPTS) { 3945 buffer->raw = 0; 3946 buffer->frameCount = 0; 3947 return INVALID_OPERATION; 3948 } 3949 3950 Mutex::Autolock _l(mTimedBufferQueueLock); 3951 3952 while (true) { 3953 3954 // if we have no timed buffers, then fail 3955 if (mTimedBufferQueue.isEmpty()) { 3956 buffer->raw = 0; 3957 buffer->frameCount = 0; 3958 return NOT_ENOUGH_DATA; 3959 } 3960 3961 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3962 3963 // calculate the PTS of the head of the timed buffer queue expressed in 3964 // local time 3965 int64_t headLocalPTS; 3966 { 3967 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3968 3969 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 3970 3971 if (mMediaTimeTransform.a_to_b_denom == 0) { 3972 // the transform represents a pause, so yield silence 3973 timedYieldSilence(buffer->frameCount, buffer); 3974 return NO_ERROR; 3975 } 3976 3977 int64_t transformedPTS; 3978 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3979 &transformedPTS)) { 3980 // the transform failed. this shouldn't happen, but if it does 3981 // then just drop this buffer 3982 ALOGW("timedGetNextBuffer transform failed"); 3983 buffer->raw = 0; 3984 buffer->frameCount = 0; 3985 mTimedBufferQueue.removeAt(0); 3986 return NO_ERROR; 3987 } 3988 3989 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3990 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3991 &headLocalPTS)) { 3992 buffer->raw = 0; 3993 buffer->frameCount = 0; 3994 return INVALID_OPERATION; 3995 } 3996 } else { 3997 headLocalPTS = transformedPTS; 3998 } 3999 } 4000 4001 // adjust the head buffer's PTS to reflect the portion of the head buffer 4002 // that has already been consumed 4003 int64_t effectivePTS = headLocalPTS + 4004 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 4005 4006 // Calculate the delta in samples between the head of the input buffer 4007 // queue and the start of the next output buffer that will be written. 4008 // If the transformation fails because of over or underflow, it means 4009 // that the sample's position in the output stream is so far out of 4010 // whack that it should just be dropped. 4011 int64_t sampleDelta; 4012 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 4013 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 4014 mTimedBufferQueue.removeAt(0); 4015 continue; 4016 } 4017 if (!mLocalTimeToSampleTransform.doForwardTransform( 4018 (effectivePTS - pts) << 32, &sampleDelta)) { 4019 ALOGV("*** too late during sample rate transform: dropped buffer"); 4020 mTimedBufferQueue.removeAt(0); 4021 continue; 4022 } 4023 4024 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4025 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4026 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4027 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4028 4029 // if the delta between the ideal placement for the next input sample and 4030 // the current output position is within this threshold, then we will 4031 // concatenate the next input samples to the previous output 4032 const int64_t kSampleContinuityThreshold = 4033 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4034 4035 // if this is the first buffer of audio that we're emitting from this track 4036 // then it should be almost exactly on time. 4037 const int64_t kSampleStartupThreshold = 1LL << 32; 4038 4039 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4040 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4041 // the next input is close enough to being on time, so concatenate it 4042 // with the last output 4043 timedYieldSamples(buffer); 4044 4045 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4046 return NO_ERROR; 4047 } else if (sampleDelta > 0) { 4048 // the gap between the current output position and the proper start of 4049 // the next input sample is too big, so fill it with silence 4050 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4051 4052 timedYieldSilence(framesUntilNextInput, buffer); 4053 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4054 return NO_ERROR; 4055 } else { 4056 // the next input sample is late 4057 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4058 size_t onTimeSamplePosition = 4059 head.position() + lateFrames * mCblk->frameSize; 4060 4061 if (onTimeSamplePosition > head.buffer()->size()) { 4062 // all the remaining samples in the head are too late, so 4063 // drop it and move on 4064 ALOGV("*** too late: dropped buffer"); 4065 mTimedBufferQueue.removeAt(0); 4066 continue; 4067 } else { 4068 // skip over the late samples 4069 head.setPosition(onTimeSamplePosition); 4070 4071 // yield the available samples 4072 timedYieldSamples(buffer); 4073 4074 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4075 return NO_ERROR; 4076 } 4077 } 4078 } 4079} 4080 4081// Yield samples from the timed buffer queue head up to the given output 4082// buffer's capacity. 4083// 4084// Caller must hold mTimedBufferQueueLock 4085void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4086 AudioBufferProvider::Buffer* buffer) { 4087 4088 const TimedBuffer& head = mTimedBufferQueue[0]; 4089 4090 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4091 head.position()); 4092 4093 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4094 mCblk->frameSize); 4095 size_t framesRequested = buffer->frameCount; 4096 buffer->frameCount = min(framesLeftInHead, framesRequested); 4097 4098 mTimedAudioOutputOnTime = true; 4099} 4100 4101// Yield samples of silence up to the given output buffer's capacity 4102// 4103// Caller must hold mTimedBufferQueueLock 4104void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4105 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4106 4107 // lazily allocate a buffer filled with silence 4108 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4109 delete [] mTimedSilenceBuffer; 4110 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4111 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4112 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4113 } 4114 4115 buffer->raw = mTimedSilenceBuffer; 4116 size_t framesRequested = buffer->frameCount; 4117 buffer->frameCount = min(numFrames, framesRequested); 4118 4119 mTimedAudioOutputOnTime = false; 4120} 4121 4122// AudioBufferProvider interface 4123void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4124 AudioBufferProvider::Buffer* buffer) { 4125 4126 Mutex::Autolock _l(mTimedBufferQueueLock); 4127 4128 // If the buffer which was just released is part of the buffer at the head 4129 // of the queue, be sure to update the amt of the buffer which has been 4130 // consumed. If the buffer being returned is not part of the head of the 4131 // queue, its either because the buffer is part of the silence buffer, or 4132 // because the head of the timed queue was trimmed after the mixer called 4133 // getNextBuffer but before the mixer called releaseBuffer. 4134 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4135 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4136 4137 void* start = head.buffer()->pointer(); 4138 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4139 4140 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4141 head.setPosition(head.position() + 4142 (buffer->frameCount * mCblk->frameSize)); 4143 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4144 mTimedBufferQueue.removeAt(0); 4145 } 4146 } 4147 } 4148 4149 buffer->raw = 0; 4150 buffer->frameCount = 0; 4151} 4152 4153uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4154 Mutex::Autolock _l(mTimedBufferQueueLock); 4155 4156 uint32_t frames = 0; 4157 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4158 const TimedBuffer& tb = mTimedBufferQueue[i]; 4159 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4160 } 4161 4162 return frames; 4163} 4164 4165AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4166 : mPTS(0), mPosition(0) {} 4167 4168AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4169 const sp<IMemory>& buffer, int64_t pts) 4170 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4171 4172// ---------------------------------------------------------------------------- 4173 4174// RecordTrack constructor must be called with AudioFlinger::mLock held 4175AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4176 RecordThread *thread, 4177 const sp<Client>& client, 4178 uint32_t sampleRate, 4179 audio_format_t format, 4180 uint32_t channelMask, 4181 int frameCount, 4182 int sessionId) 4183 : TrackBase(thread, client, sampleRate, format, 4184 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4185 mOverflow(false) 4186{ 4187 if (mCblk != NULL) { 4188 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4189 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4190 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4191 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4192 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4193 } else { 4194 mCblk->frameSize = sizeof(int8_t); 4195 } 4196 } 4197} 4198 4199AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4200{ 4201 sp<ThreadBase> thread = mThread.promote(); 4202 if (thread != 0) { 4203 AudioSystem::releaseInput(thread->id()); 4204 } 4205} 4206 4207// AudioBufferProvider interface 4208status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4209{ 4210 audio_track_cblk_t* cblk = this->cblk(); 4211 uint32_t framesAvail; 4212 uint32_t framesReq = buffer->frameCount; 4213 4214 // Check if last stepServer failed, try to step now 4215 if (mStepServerFailed) { 4216 if (!step()) goto getNextBuffer_exit; 4217 ALOGV("stepServer recovered"); 4218 mStepServerFailed = false; 4219 } 4220 4221 framesAvail = cblk->framesAvailable_l(); 4222 4223 if (CC_LIKELY(framesAvail)) { 4224 uint32_t s = cblk->server; 4225 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4226 4227 if (framesReq > framesAvail) { 4228 framesReq = framesAvail; 4229 } 4230 if (s + framesReq > bufferEnd) { 4231 framesReq = bufferEnd - s; 4232 } 4233 4234 buffer->raw = getBuffer(s, framesReq); 4235 if (buffer->raw == NULL) goto getNextBuffer_exit; 4236 4237 buffer->frameCount = framesReq; 4238 return NO_ERROR; 4239 } 4240 4241getNextBuffer_exit: 4242 buffer->raw = NULL; 4243 buffer->frameCount = 0; 4244 return NOT_ENOUGH_DATA; 4245} 4246 4247status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4248{ 4249 sp<ThreadBase> thread = mThread.promote(); 4250 if (thread != 0) { 4251 RecordThread *recordThread = (RecordThread *)thread.get(); 4252 return recordThread->start(this, tid); 4253 } else { 4254 return BAD_VALUE; 4255 } 4256} 4257 4258void AudioFlinger::RecordThread::RecordTrack::stop() 4259{ 4260 sp<ThreadBase> thread = mThread.promote(); 4261 if (thread != 0) { 4262 RecordThread *recordThread = (RecordThread *)thread.get(); 4263 recordThread->stop(this); 4264 TrackBase::reset(); 4265 // Force overerrun condition to avoid false overrun callback until first data is 4266 // read from buffer 4267 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4268 } 4269} 4270 4271void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4272{ 4273 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4274 (mClient == 0) ? getpid_cached : mClient->pid(), 4275 mFormat, 4276 mChannelMask, 4277 mSessionId, 4278 mFrameCount, 4279 mState, 4280 mCblk->sampleRate, 4281 mCblk->server, 4282 mCblk->user); 4283} 4284 4285 4286// ---------------------------------------------------------------------------- 4287 4288AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4289 PlaybackThread *playbackThread, 4290 DuplicatingThread *sourceThread, 4291 uint32_t sampleRate, 4292 audio_format_t format, 4293 uint32_t channelMask, 4294 int frameCount) 4295 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4296 mActive(false), mSourceThread(sourceThread) 4297{ 4298 4299 if (mCblk != NULL) { 4300 mCblk->flags |= CBLK_DIRECTION_OUT; 4301 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4302 mOutBuffer.frameCount = 0; 4303 playbackThread->mTracks.add(this); 4304 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4305 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4306 mCblk, mBuffer, mCblk->buffers, 4307 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4308 } else { 4309 ALOGW("Error creating output track on thread %p", playbackThread); 4310 } 4311} 4312 4313AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4314{ 4315 clearBufferQueue(); 4316} 4317 4318status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4319{ 4320 status_t status = Track::start(tid); 4321 if (status != NO_ERROR) { 4322 return status; 4323 } 4324 4325 mActive = true; 4326 mRetryCount = 127; 4327 return status; 4328} 4329 4330void AudioFlinger::PlaybackThread::OutputTrack::stop() 4331{ 4332 Track::stop(); 4333 clearBufferQueue(); 4334 mOutBuffer.frameCount = 0; 4335 mActive = false; 4336} 4337 4338bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4339{ 4340 Buffer *pInBuffer; 4341 Buffer inBuffer; 4342 uint32_t channelCount = mChannelCount; 4343 bool outputBufferFull = false; 4344 inBuffer.frameCount = frames; 4345 inBuffer.i16 = data; 4346 4347 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4348 4349 if (!mActive && frames != 0) { 4350 start(0); 4351 sp<ThreadBase> thread = mThread.promote(); 4352 if (thread != 0) { 4353 MixerThread *mixerThread = (MixerThread *)thread.get(); 4354 if (mCblk->frameCount > frames){ 4355 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4356 uint32_t startFrames = (mCblk->frameCount - frames); 4357 pInBuffer = new Buffer; 4358 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4359 pInBuffer->frameCount = startFrames; 4360 pInBuffer->i16 = pInBuffer->mBuffer; 4361 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4362 mBufferQueue.add(pInBuffer); 4363 } else { 4364 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4365 } 4366 } 4367 } 4368 } 4369 4370 while (waitTimeLeftMs) { 4371 // First write pending buffers, then new data 4372 if (mBufferQueue.size()) { 4373 pInBuffer = mBufferQueue.itemAt(0); 4374 } else { 4375 pInBuffer = &inBuffer; 4376 } 4377 4378 if (pInBuffer->frameCount == 0) { 4379 break; 4380 } 4381 4382 if (mOutBuffer.frameCount == 0) { 4383 mOutBuffer.frameCount = pInBuffer->frameCount; 4384 nsecs_t startTime = systemTime(); 4385 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4386 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4387 outputBufferFull = true; 4388 break; 4389 } 4390 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4391 if (waitTimeLeftMs >= waitTimeMs) { 4392 waitTimeLeftMs -= waitTimeMs; 4393 } else { 4394 waitTimeLeftMs = 0; 4395 } 4396 } 4397 4398 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4399 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4400 mCblk->stepUser(outFrames); 4401 pInBuffer->frameCount -= outFrames; 4402 pInBuffer->i16 += outFrames * channelCount; 4403 mOutBuffer.frameCount -= outFrames; 4404 mOutBuffer.i16 += outFrames * channelCount; 4405 4406 if (pInBuffer->frameCount == 0) { 4407 if (mBufferQueue.size()) { 4408 mBufferQueue.removeAt(0); 4409 delete [] pInBuffer->mBuffer; 4410 delete pInBuffer; 4411 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4412 } else { 4413 break; 4414 } 4415 } 4416 } 4417 4418 // If we could not write all frames, allocate a buffer and queue it for next time. 4419 if (inBuffer.frameCount) { 4420 sp<ThreadBase> thread = mThread.promote(); 4421 if (thread != 0 && !thread->standby()) { 4422 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4423 pInBuffer = new Buffer; 4424 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4425 pInBuffer->frameCount = inBuffer.frameCount; 4426 pInBuffer->i16 = pInBuffer->mBuffer; 4427 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4428 mBufferQueue.add(pInBuffer); 4429 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4430 } else { 4431 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4432 } 4433 } 4434 } 4435 4436 // Calling write() with a 0 length buffer, means that no more data will be written: 4437 // If no more buffers are pending, fill output track buffer to make sure it is started 4438 // by output mixer. 4439 if (frames == 0 && mBufferQueue.size() == 0) { 4440 if (mCblk->user < mCblk->frameCount) { 4441 frames = mCblk->frameCount - mCblk->user; 4442 pInBuffer = new Buffer; 4443 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4444 pInBuffer->frameCount = frames; 4445 pInBuffer->i16 = pInBuffer->mBuffer; 4446 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4447 mBufferQueue.add(pInBuffer); 4448 } else if (mActive) { 4449 stop(); 4450 } 4451 } 4452 4453 return outputBufferFull; 4454} 4455 4456status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4457{ 4458 int active; 4459 status_t result; 4460 audio_track_cblk_t* cblk = mCblk; 4461 uint32_t framesReq = buffer->frameCount; 4462 4463// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4464 buffer->frameCount = 0; 4465 4466 uint32_t framesAvail = cblk->framesAvailable(); 4467 4468 4469 if (framesAvail == 0) { 4470 Mutex::Autolock _l(cblk->lock); 4471 goto start_loop_here; 4472 while (framesAvail == 0) { 4473 active = mActive; 4474 if (CC_UNLIKELY(!active)) { 4475 ALOGV("Not active and NO_MORE_BUFFERS"); 4476 return NO_MORE_BUFFERS; 4477 } 4478 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4479 if (result != NO_ERROR) { 4480 return NO_MORE_BUFFERS; 4481 } 4482 // read the server count again 4483 start_loop_here: 4484 framesAvail = cblk->framesAvailable_l(); 4485 } 4486 } 4487 4488// if (framesAvail < framesReq) { 4489// return NO_MORE_BUFFERS; 4490// } 4491 4492 if (framesReq > framesAvail) { 4493 framesReq = framesAvail; 4494 } 4495 4496 uint32_t u = cblk->user; 4497 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4498 4499 if (u + framesReq > bufferEnd) { 4500 framesReq = bufferEnd - u; 4501 } 4502 4503 buffer->frameCount = framesReq; 4504 buffer->raw = (void *)cblk->buffer(u); 4505 return NO_ERROR; 4506} 4507 4508 4509void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4510{ 4511 size_t size = mBufferQueue.size(); 4512 4513 for (size_t i = 0; i < size; i++) { 4514 Buffer *pBuffer = mBufferQueue.itemAt(i); 4515 delete [] pBuffer->mBuffer; 4516 delete pBuffer; 4517 } 4518 mBufferQueue.clear(); 4519} 4520 4521// ---------------------------------------------------------------------------- 4522 4523AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4524 : RefBase(), 4525 mAudioFlinger(audioFlinger), 4526 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4527 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4528 mPid(pid), 4529 mTimedTrackCount(0) 4530{ 4531 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4532} 4533 4534// Client destructor must be called with AudioFlinger::mLock held 4535AudioFlinger::Client::~Client() 4536{ 4537 mAudioFlinger->removeClient_l(mPid); 4538} 4539 4540sp<MemoryDealer> AudioFlinger::Client::heap() const 4541{ 4542 return mMemoryDealer; 4543} 4544 4545// Reserve one of the limited slots for a timed audio track associated 4546// with this client 4547bool AudioFlinger::Client::reserveTimedTrack() 4548{ 4549 const int kMaxTimedTracksPerClient = 4; 4550 4551 Mutex::Autolock _l(mTimedTrackLock); 4552 4553 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4554 ALOGW("can not create timed track - pid %d has exceeded the limit", 4555 mPid); 4556 return false; 4557 } 4558 4559 mTimedTrackCount++; 4560 return true; 4561} 4562 4563// Release a slot for a timed audio track 4564void AudioFlinger::Client::releaseTimedTrack() 4565{ 4566 Mutex::Autolock _l(mTimedTrackLock); 4567 mTimedTrackCount--; 4568} 4569 4570// ---------------------------------------------------------------------------- 4571 4572AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4573 const sp<IAudioFlingerClient>& client, 4574 pid_t pid) 4575 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4576{ 4577} 4578 4579AudioFlinger::NotificationClient::~NotificationClient() 4580{ 4581} 4582 4583void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4584{ 4585 sp<NotificationClient> keep(this); 4586 mAudioFlinger->removeNotificationClient(mPid); 4587} 4588 4589// ---------------------------------------------------------------------------- 4590 4591AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4592 : BnAudioTrack(), 4593 mTrack(track) 4594{ 4595} 4596 4597AudioFlinger::TrackHandle::~TrackHandle() { 4598 // just stop the track on deletion, associated resources 4599 // will be freed from the main thread once all pending buffers have 4600 // been played. Unless it's not in the active track list, in which 4601 // case we free everything now... 4602 mTrack->destroy(); 4603} 4604 4605sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4606 return mTrack->getCblk(); 4607} 4608 4609status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4610 return mTrack->start(tid); 4611} 4612 4613void AudioFlinger::TrackHandle::stop() { 4614 mTrack->stop(); 4615} 4616 4617void AudioFlinger::TrackHandle::flush() { 4618 mTrack->flush(); 4619} 4620 4621void AudioFlinger::TrackHandle::mute(bool e) { 4622 mTrack->mute(e); 4623} 4624 4625void AudioFlinger::TrackHandle::pause() { 4626 mTrack->pause(); 4627} 4628 4629status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4630{ 4631 return mTrack->attachAuxEffect(EffectId); 4632} 4633 4634status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4635 sp<IMemory>* buffer) { 4636 if (!mTrack->isTimedTrack()) 4637 return INVALID_OPERATION; 4638 4639 PlaybackThread::TimedTrack* tt = 4640 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4641 return tt->allocateTimedBuffer(size, buffer); 4642} 4643 4644status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4645 int64_t pts) { 4646 if (!mTrack->isTimedTrack()) 4647 return INVALID_OPERATION; 4648 4649 PlaybackThread::TimedTrack* tt = 4650 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4651 return tt->queueTimedBuffer(buffer, pts); 4652} 4653 4654status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4655 const LinearTransform& xform, int target) { 4656 4657 if (!mTrack->isTimedTrack()) 4658 return INVALID_OPERATION; 4659 4660 PlaybackThread::TimedTrack* tt = 4661 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4662 return tt->setMediaTimeTransform( 4663 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4664} 4665 4666status_t AudioFlinger::TrackHandle::onTransact( 4667 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4668{ 4669 return BnAudioTrack::onTransact(code, data, reply, flags); 4670} 4671 4672// ---------------------------------------------------------------------------- 4673 4674sp<IAudioRecord> AudioFlinger::openRecord( 4675 pid_t pid, 4676 audio_io_handle_t input, 4677 uint32_t sampleRate, 4678 audio_format_t format, 4679 uint32_t channelMask, 4680 int frameCount, 4681 // FIXME dead, remove from IAudioFlinger 4682 uint32_t flags, 4683 int *sessionId, 4684 status_t *status) 4685{ 4686 sp<RecordThread::RecordTrack> recordTrack; 4687 sp<RecordHandle> recordHandle; 4688 sp<Client> client; 4689 status_t lStatus; 4690 RecordThread *thread; 4691 size_t inFrameCount; 4692 int lSessionId; 4693 4694 // check calling permissions 4695 if (!recordingAllowed()) { 4696 lStatus = PERMISSION_DENIED; 4697 goto Exit; 4698 } 4699 4700 // add client to list 4701 { // scope for mLock 4702 Mutex::Autolock _l(mLock); 4703 thread = checkRecordThread_l(input); 4704 if (thread == NULL) { 4705 lStatus = BAD_VALUE; 4706 goto Exit; 4707 } 4708 4709 client = registerPid_l(pid); 4710 4711 // If no audio session id is provided, create one here 4712 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4713 lSessionId = *sessionId; 4714 } else { 4715 lSessionId = nextUniqueId(); 4716 if (sessionId != NULL) { 4717 *sessionId = lSessionId; 4718 } 4719 } 4720 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4721 recordTrack = thread->createRecordTrack_l(client, 4722 sampleRate, 4723 format, 4724 channelMask, 4725 frameCount, 4726 lSessionId, 4727 &lStatus); 4728 } 4729 if (lStatus != NO_ERROR) { 4730 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4731 // destructor is called by the TrackBase destructor with mLock held 4732 client.clear(); 4733 recordTrack.clear(); 4734 goto Exit; 4735 } 4736 4737 // return to handle to client 4738 recordHandle = new RecordHandle(recordTrack); 4739 lStatus = NO_ERROR; 4740 4741Exit: 4742 if (status) { 4743 *status = lStatus; 4744 } 4745 return recordHandle; 4746} 4747 4748// ---------------------------------------------------------------------------- 4749 4750AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4751 : BnAudioRecord(), 4752 mRecordTrack(recordTrack) 4753{ 4754} 4755 4756AudioFlinger::RecordHandle::~RecordHandle() { 4757 stop(); 4758} 4759 4760sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4761 return mRecordTrack->getCblk(); 4762} 4763 4764status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4765 ALOGV("RecordHandle::start()"); 4766 return mRecordTrack->start(tid); 4767} 4768 4769void AudioFlinger::RecordHandle::stop() { 4770 ALOGV("RecordHandle::stop()"); 4771 mRecordTrack->stop(); 4772} 4773 4774status_t AudioFlinger::RecordHandle::onTransact( 4775 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4776{ 4777 return BnAudioRecord::onTransact(code, data, reply, flags); 4778} 4779 4780// ---------------------------------------------------------------------------- 4781 4782AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4783 AudioStreamIn *input, 4784 uint32_t sampleRate, 4785 uint32_t channels, 4786 audio_io_handle_t id, 4787 uint32_t device) : 4788 ThreadBase(audioFlinger, id, device, RECORD), 4789 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4790 // mRsmpInIndex and mInputBytes set by readInputParameters() 4791 mReqChannelCount(popcount(channels)), 4792 mReqSampleRate(sampleRate) 4793 // mBytesRead is only meaningful while active, and so is cleared in start() 4794 // (but might be better to also clear here for dump?) 4795{ 4796 snprintf(mName, kNameLength, "AudioIn_%X", id); 4797 4798 readInputParameters(); 4799} 4800 4801 4802AudioFlinger::RecordThread::~RecordThread() 4803{ 4804 delete[] mRsmpInBuffer; 4805 delete mResampler; 4806 delete[] mRsmpOutBuffer; 4807} 4808 4809void AudioFlinger::RecordThread::onFirstRef() 4810{ 4811 run(mName, PRIORITY_URGENT_AUDIO); 4812} 4813 4814status_t AudioFlinger::RecordThread::readyToRun() 4815{ 4816 status_t status = initCheck(); 4817 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4818 return status; 4819} 4820 4821bool AudioFlinger::RecordThread::threadLoop() 4822{ 4823 AudioBufferProvider::Buffer buffer; 4824 sp<RecordTrack> activeTrack; 4825 Vector< sp<EffectChain> > effectChains; 4826 4827 nsecs_t lastWarning = 0; 4828 4829 acquireWakeLock(); 4830 4831 // start recording 4832 while (!exitPending()) { 4833 4834 processConfigEvents(); 4835 4836 { // scope for mLock 4837 Mutex::Autolock _l(mLock); 4838 checkForNewParameters_l(); 4839 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4840 if (!mStandby) { 4841 mInput->stream->common.standby(&mInput->stream->common); 4842 mStandby = true; 4843 } 4844 4845 if (exitPending()) break; 4846 4847 releaseWakeLock_l(); 4848 ALOGV("RecordThread: loop stopping"); 4849 // go to sleep 4850 mWaitWorkCV.wait(mLock); 4851 ALOGV("RecordThread: loop starting"); 4852 acquireWakeLock_l(); 4853 continue; 4854 } 4855 if (mActiveTrack != 0) { 4856 if (mActiveTrack->mState == TrackBase::PAUSING) { 4857 if (!mStandby) { 4858 mInput->stream->common.standby(&mInput->stream->common); 4859 mStandby = true; 4860 } 4861 mActiveTrack.clear(); 4862 mStartStopCond.broadcast(); 4863 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4864 if (mReqChannelCount != mActiveTrack->channelCount()) { 4865 mActiveTrack.clear(); 4866 mStartStopCond.broadcast(); 4867 } else if (mBytesRead != 0) { 4868 // record start succeeds only if first read from audio input 4869 // succeeds 4870 if (mBytesRead > 0) { 4871 mActiveTrack->mState = TrackBase::ACTIVE; 4872 } else { 4873 mActiveTrack.clear(); 4874 } 4875 mStartStopCond.broadcast(); 4876 } 4877 mStandby = false; 4878 } 4879 } 4880 lockEffectChains_l(effectChains); 4881 } 4882 4883 if (mActiveTrack != 0) { 4884 if (mActiveTrack->mState != TrackBase::ACTIVE && 4885 mActiveTrack->mState != TrackBase::RESUMING) { 4886 unlockEffectChains(effectChains); 4887 usleep(kRecordThreadSleepUs); 4888 continue; 4889 } 4890 for (size_t i = 0; i < effectChains.size(); i ++) { 4891 effectChains[i]->process_l(); 4892 } 4893 4894 buffer.frameCount = mFrameCount; 4895 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4896 size_t framesOut = buffer.frameCount; 4897 if (mResampler == NULL) { 4898 // no resampling 4899 while (framesOut) { 4900 size_t framesIn = mFrameCount - mRsmpInIndex; 4901 if (framesIn) { 4902 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4903 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4904 if (framesIn > framesOut) 4905 framesIn = framesOut; 4906 mRsmpInIndex += framesIn; 4907 framesOut -= framesIn; 4908 if ((int)mChannelCount == mReqChannelCount || 4909 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4910 memcpy(dst, src, framesIn * mFrameSize); 4911 } else { 4912 int16_t *src16 = (int16_t *)src; 4913 int16_t *dst16 = (int16_t *)dst; 4914 if (mChannelCount == 1) { 4915 while (framesIn--) { 4916 *dst16++ = *src16; 4917 *dst16++ = *src16++; 4918 } 4919 } else { 4920 while (framesIn--) { 4921 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4922 src16 += 2; 4923 } 4924 } 4925 } 4926 } 4927 if (framesOut && mFrameCount == mRsmpInIndex) { 4928 if (framesOut == mFrameCount && 4929 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4930 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4931 framesOut = 0; 4932 } else { 4933 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4934 mRsmpInIndex = 0; 4935 } 4936 if (mBytesRead < 0) { 4937 ALOGE("Error reading audio input"); 4938 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4939 // Force input into standby so that it tries to 4940 // recover at next read attempt 4941 mInput->stream->common.standby(&mInput->stream->common); 4942 usleep(kRecordThreadSleepUs); 4943 } 4944 mRsmpInIndex = mFrameCount; 4945 framesOut = 0; 4946 buffer.frameCount = 0; 4947 } 4948 } 4949 } 4950 } else { 4951 // resampling 4952 4953 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4954 // alter output frame count as if we were expecting stereo samples 4955 if (mChannelCount == 1 && mReqChannelCount == 1) { 4956 framesOut >>= 1; 4957 } 4958 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4959 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4960 // are 32 bit aligned which should be always true. 4961 if (mChannelCount == 2 && mReqChannelCount == 1) { 4962 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4963 // the resampler always outputs stereo samples: do post stereo to mono conversion 4964 int16_t *src = (int16_t *)mRsmpOutBuffer; 4965 int16_t *dst = buffer.i16; 4966 while (framesOut--) { 4967 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4968 src += 2; 4969 } 4970 } else { 4971 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4972 } 4973 4974 } 4975 mActiveTrack->releaseBuffer(&buffer); 4976 mActiveTrack->overflow(); 4977 } 4978 // client isn't retrieving buffers fast enough 4979 else { 4980 if (!mActiveTrack->setOverflow()) { 4981 nsecs_t now = systemTime(); 4982 if ((now - lastWarning) > kWarningThrottleNs) { 4983 ALOGW("RecordThread: buffer overflow"); 4984 lastWarning = now; 4985 } 4986 } 4987 // Release the processor for a while before asking for a new buffer. 4988 // This will give the application more chance to read from the buffer and 4989 // clear the overflow. 4990 usleep(kRecordThreadSleepUs); 4991 } 4992 } 4993 // enable changes in effect chain 4994 unlockEffectChains(effectChains); 4995 effectChains.clear(); 4996 } 4997 4998 if (!mStandby) { 4999 mInput->stream->common.standby(&mInput->stream->common); 5000 } 5001 mActiveTrack.clear(); 5002 5003 mStartStopCond.broadcast(); 5004 5005 releaseWakeLock(); 5006 5007 ALOGV("RecordThread %p exiting", this); 5008 return false; 5009} 5010 5011 5012sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 5013 const sp<AudioFlinger::Client>& client, 5014 uint32_t sampleRate, 5015 audio_format_t format, 5016 int channelMask, 5017 int frameCount, 5018 int sessionId, 5019 status_t *status) 5020{ 5021 sp<RecordTrack> track; 5022 status_t lStatus; 5023 5024 lStatus = initCheck(); 5025 if (lStatus != NO_ERROR) { 5026 ALOGE("Audio driver not initialized."); 5027 goto Exit; 5028 } 5029 5030 { // scope for mLock 5031 Mutex::Autolock _l(mLock); 5032 5033 track = new RecordTrack(this, client, sampleRate, 5034 format, channelMask, frameCount, sessionId); 5035 5036 if (track->getCblk() == 0) { 5037 lStatus = NO_MEMORY; 5038 goto Exit; 5039 } 5040 5041 mTrack = track.get(); 5042 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5043 bool suspend = audio_is_bluetooth_sco_device( 5044 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5045 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5046 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5047 } 5048 lStatus = NO_ERROR; 5049 5050Exit: 5051 if (status) { 5052 *status = lStatus; 5053 } 5054 return track; 5055} 5056 5057status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5058{ 5059 ALOGV("RecordThread::start tid=%d", tid); 5060 sp<ThreadBase> strongMe = this; 5061 status_t status = NO_ERROR; 5062 { 5063 AutoMutex lock(mLock); 5064 if (mActiveTrack != 0) { 5065 if (recordTrack != mActiveTrack.get()) { 5066 status = -EBUSY; 5067 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5068 mActiveTrack->mState = TrackBase::ACTIVE; 5069 } 5070 return status; 5071 } 5072 5073 recordTrack->mState = TrackBase::IDLE; 5074 mActiveTrack = recordTrack; 5075 mLock.unlock(); 5076 status_t status = AudioSystem::startInput(mId); 5077 mLock.lock(); 5078 if (status != NO_ERROR) { 5079 mActiveTrack.clear(); 5080 return status; 5081 } 5082 mRsmpInIndex = mFrameCount; 5083 mBytesRead = 0; 5084 if (mResampler != NULL) { 5085 mResampler->reset(); 5086 } 5087 mActiveTrack->mState = TrackBase::RESUMING; 5088 // signal thread to start 5089 ALOGV("Signal record thread"); 5090 mWaitWorkCV.signal(); 5091 // do not wait for mStartStopCond if exiting 5092 if (exitPending()) { 5093 mActiveTrack.clear(); 5094 status = INVALID_OPERATION; 5095 goto startError; 5096 } 5097 mStartStopCond.wait(mLock); 5098 if (mActiveTrack == 0) { 5099 ALOGV("Record failed to start"); 5100 status = BAD_VALUE; 5101 goto startError; 5102 } 5103 ALOGV("Record started OK"); 5104 return status; 5105 } 5106startError: 5107 AudioSystem::stopInput(mId); 5108 return status; 5109} 5110 5111void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5112 ALOGV("RecordThread::stop"); 5113 sp<ThreadBase> strongMe = this; 5114 { 5115 AutoMutex lock(mLock); 5116 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5117 mActiveTrack->mState = TrackBase::PAUSING; 5118 // do not wait for mStartStopCond if exiting 5119 if (exitPending()) { 5120 return; 5121 } 5122 mStartStopCond.wait(mLock); 5123 // if we have been restarted, recordTrack == mActiveTrack.get() here 5124 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5125 mLock.unlock(); 5126 AudioSystem::stopInput(mId); 5127 mLock.lock(); 5128 ALOGV("Record stopped OK"); 5129 } 5130 } 5131 } 5132} 5133 5134status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5135{ 5136 const size_t SIZE = 256; 5137 char buffer[SIZE]; 5138 String8 result; 5139 5140 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5141 result.append(buffer); 5142 5143 if (mActiveTrack != 0) { 5144 result.append("Active Track:\n"); 5145 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5146 mActiveTrack->dump(buffer, SIZE); 5147 result.append(buffer); 5148 5149 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5150 result.append(buffer); 5151 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5152 result.append(buffer); 5153 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5154 result.append(buffer); 5155 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5156 result.append(buffer); 5157 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5158 result.append(buffer); 5159 5160 5161 } else { 5162 result.append("No record client\n"); 5163 } 5164 write(fd, result.string(), result.size()); 5165 5166 dumpBase(fd, args); 5167 dumpEffectChains(fd, args); 5168 5169 return NO_ERROR; 5170} 5171 5172// AudioBufferProvider interface 5173status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5174{ 5175 size_t framesReq = buffer->frameCount; 5176 size_t framesReady = mFrameCount - mRsmpInIndex; 5177 int channelCount; 5178 5179 if (framesReady == 0) { 5180 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5181 if (mBytesRead < 0) { 5182 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5183 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5184 // Force input into standby so that it tries to 5185 // recover at next read attempt 5186 mInput->stream->common.standby(&mInput->stream->common); 5187 usleep(kRecordThreadSleepUs); 5188 } 5189 buffer->raw = NULL; 5190 buffer->frameCount = 0; 5191 return NOT_ENOUGH_DATA; 5192 } 5193 mRsmpInIndex = 0; 5194 framesReady = mFrameCount; 5195 } 5196 5197 if (framesReq > framesReady) { 5198 framesReq = framesReady; 5199 } 5200 5201 if (mChannelCount == 1 && mReqChannelCount == 2) { 5202 channelCount = 1; 5203 } else { 5204 channelCount = 2; 5205 } 5206 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5207 buffer->frameCount = framesReq; 5208 return NO_ERROR; 5209} 5210 5211// AudioBufferProvider interface 5212void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5213{ 5214 mRsmpInIndex += buffer->frameCount; 5215 buffer->frameCount = 0; 5216} 5217 5218bool AudioFlinger::RecordThread::checkForNewParameters_l() 5219{ 5220 bool reconfig = false; 5221 5222 while (!mNewParameters.isEmpty()) { 5223 status_t status = NO_ERROR; 5224 String8 keyValuePair = mNewParameters[0]; 5225 AudioParameter param = AudioParameter(keyValuePair); 5226 int value; 5227 audio_format_t reqFormat = mFormat; 5228 int reqSamplingRate = mReqSampleRate; 5229 int reqChannelCount = mReqChannelCount; 5230 5231 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5232 reqSamplingRate = value; 5233 reconfig = true; 5234 } 5235 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5236 reqFormat = (audio_format_t) value; 5237 reconfig = true; 5238 } 5239 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5240 reqChannelCount = popcount(value); 5241 reconfig = true; 5242 } 5243 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5244 // do not accept frame count changes if tracks are open as the track buffer 5245 // size depends on frame count and correct behavior would not be guaranteed 5246 // if frame count is changed after track creation 5247 if (mActiveTrack != 0) { 5248 status = INVALID_OPERATION; 5249 } else { 5250 reconfig = true; 5251 } 5252 } 5253 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5254 // forward device change to effects that have requested to be 5255 // aware of attached audio device. 5256 for (size_t i = 0; i < mEffectChains.size(); i++) { 5257 mEffectChains[i]->setDevice_l(value); 5258 } 5259 // store input device and output device but do not forward output device to audio HAL. 5260 // Note that status is ignored by the caller for output device 5261 // (see AudioFlinger::setParameters() 5262 if (value & AUDIO_DEVICE_OUT_ALL) { 5263 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5264 status = BAD_VALUE; 5265 } else { 5266 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5267 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5268 if (mTrack != NULL) { 5269 bool suspend = audio_is_bluetooth_sco_device( 5270 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5271 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5272 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5273 } 5274 } 5275 mDevice |= (uint32_t)value; 5276 } 5277 if (status == NO_ERROR) { 5278 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5279 if (status == INVALID_OPERATION) { 5280 mInput->stream->common.standby(&mInput->stream->common); 5281 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5282 keyValuePair.string()); 5283 } 5284 if (reconfig) { 5285 if (status == BAD_VALUE && 5286 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5287 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5288 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5289 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5290 (reqChannelCount <= FCC_2)) { 5291 status = NO_ERROR; 5292 } 5293 if (status == NO_ERROR) { 5294 readInputParameters(); 5295 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5296 } 5297 } 5298 } 5299 5300 mNewParameters.removeAt(0); 5301 5302 mParamStatus = status; 5303 mParamCond.signal(); 5304 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5305 // already timed out waiting for the status and will never signal the condition. 5306 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5307 } 5308 return reconfig; 5309} 5310 5311String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5312{ 5313 char *s; 5314 String8 out_s8 = String8(); 5315 5316 Mutex::Autolock _l(mLock); 5317 if (initCheck() != NO_ERROR) { 5318 return out_s8; 5319 } 5320 5321 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5322 out_s8 = String8(s); 5323 free(s); 5324 return out_s8; 5325} 5326 5327void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5328 AudioSystem::OutputDescriptor desc; 5329 void *param2 = NULL; 5330 5331 switch (event) { 5332 case AudioSystem::INPUT_OPENED: 5333 case AudioSystem::INPUT_CONFIG_CHANGED: 5334 desc.channels = mChannelMask; 5335 desc.samplingRate = mSampleRate; 5336 desc.format = mFormat; 5337 desc.frameCount = mFrameCount; 5338 desc.latency = 0; 5339 param2 = &desc; 5340 break; 5341 5342 case AudioSystem::INPUT_CLOSED: 5343 default: 5344 break; 5345 } 5346 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5347} 5348 5349void AudioFlinger::RecordThread::readInputParameters() 5350{ 5351 delete mRsmpInBuffer; 5352 // mRsmpInBuffer is always assigned a new[] below 5353 delete mRsmpOutBuffer; 5354 mRsmpOutBuffer = NULL; 5355 delete mResampler; 5356 mResampler = NULL; 5357 5358 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5359 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5360 mChannelCount = (uint16_t)popcount(mChannelMask); 5361 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5362 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5363 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5364 mFrameCount = mInputBytes / mFrameSize; 5365 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5366 5367 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5368 { 5369 int channelCount; 5370 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5371 // stereo to mono post process as the resampler always outputs stereo. 5372 if (mChannelCount == 1 && mReqChannelCount == 2) { 5373 channelCount = 1; 5374 } else { 5375 channelCount = 2; 5376 } 5377 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5378 mResampler->setSampleRate(mSampleRate); 5379 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5380 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5381 5382 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5383 if (mChannelCount == 1 && mReqChannelCount == 1) { 5384 mFrameCount >>= 1; 5385 } 5386 5387 } 5388 mRsmpInIndex = mFrameCount; 5389} 5390 5391unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5392{ 5393 Mutex::Autolock _l(mLock); 5394 if (initCheck() != NO_ERROR) { 5395 return 0; 5396 } 5397 5398 return mInput->stream->get_input_frames_lost(mInput->stream); 5399} 5400 5401uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5402{ 5403 Mutex::Autolock _l(mLock); 5404 uint32_t result = 0; 5405 if (getEffectChain_l(sessionId) != 0) { 5406 result = EFFECT_SESSION; 5407 } 5408 5409 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5410 result |= TRACK_SESSION; 5411 } 5412 5413 return result; 5414} 5415 5416AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5417{ 5418 Mutex::Autolock _l(mLock); 5419 return mTrack; 5420} 5421 5422AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5423{ 5424 Mutex::Autolock _l(mLock); 5425 return mInput; 5426} 5427 5428AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5429{ 5430 Mutex::Autolock _l(mLock); 5431 AudioStreamIn *input = mInput; 5432 mInput = NULL; 5433 return input; 5434} 5435 5436// this method must always be called either with ThreadBase mLock held or inside the thread loop 5437audio_stream_t* AudioFlinger::RecordThread::stream() 5438{ 5439 if (mInput == NULL) { 5440 return NULL; 5441 } 5442 return &mInput->stream->common; 5443} 5444 5445 5446// ---------------------------------------------------------------------------- 5447 5448audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5449 uint32_t *pSamplingRate, 5450 audio_format_t *pFormat, 5451 uint32_t *pChannels, 5452 uint32_t *pLatencyMs, 5453 audio_policy_output_flags_t flags) 5454{ 5455 status_t status; 5456 PlaybackThread *thread = NULL; 5457 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5458 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5459 uint32_t channels = pChannels ? *pChannels : 0; 5460 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5461 audio_stream_out_t *outStream; 5462 audio_hw_device_t *outHwDev; 5463 5464 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5465 pDevices ? *pDevices : 0, 5466 samplingRate, 5467 format, 5468 channels, 5469 flags); 5470 5471 if (pDevices == NULL || *pDevices == 0) { 5472 return 0; 5473 } 5474 5475 Mutex::Autolock _l(mLock); 5476 5477 outHwDev = findSuitableHwDev_l(*pDevices); 5478 if (outHwDev == NULL) 5479 return 0; 5480 5481 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5482 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5483 &channels, &samplingRate, &outStream); 5484 mHardwareStatus = AUDIO_HW_IDLE; 5485 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5486 outStream, 5487 samplingRate, 5488 format, 5489 channels, 5490 status); 5491 5492 if (outStream != NULL) { 5493 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5494 audio_io_handle_t id = nextUniqueId(); 5495 5496 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5497 (format != AUDIO_FORMAT_PCM_16_BIT) || 5498 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5499 thread = new DirectOutputThread(this, output, id, *pDevices); 5500 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5501 } else { 5502 thread = new MixerThread(this, output, id, *pDevices); 5503 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5504 } 5505 mPlaybackThreads.add(id, thread); 5506 5507 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5508 if (pFormat != NULL) *pFormat = format; 5509 if (pChannels != NULL) *pChannels = channels; 5510 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5511 5512 // notify client processes of the new output creation 5513 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5514 return id; 5515 } 5516 5517 return 0; 5518} 5519 5520audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5521 audio_io_handle_t output2) 5522{ 5523 Mutex::Autolock _l(mLock); 5524 MixerThread *thread1 = checkMixerThread_l(output1); 5525 MixerThread *thread2 = checkMixerThread_l(output2); 5526 5527 if (thread1 == NULL || thread2 == NULL) { 5528 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5529 return 0; 5530 } 5531 5532 audio_io_handle_t id = nextUniqueId(); 5533 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5534 thread->addOutputTrack(thread2); 5535 mPlaybackThreads.add(id, thread); 5536 // notify client processes of the new output creation 5537 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5538 return id; 5539} 5540 5541status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5542{ 5543 // keep strong reference on the playback thread so that 5544 // it is not destroyed while exit() is executed 5545 sp<PlaybackThread> thread; 5546 { 5547 Mutex::Autolock _l(mLock); 5548 thread = checkPlaybackThread_l(output); 5549 if (thread == NULL) { 5550 return BAD_VALUE; 5551 } 5552 5553 ALOGV("closeOutput() %d", output); 5554 5555 if (thread->type() == ThreadBase::MIXER) { 5556 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5557 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5558 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5559 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5560 } 5561 } 5562 } 5563 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5564 mPlaybackThreads.removeItem(output); 5565 } 5566 thread->exit(); 5567 // The thread entity (active unit of execution) is no longer running here, 5568 // but the ThreadBase container still exists. 5569 5570 if (thread->type() != ThreadBase::DUPLICATING) { 5571 AudioStreamOut *out = thread->clearOutput(); 5572 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5573 // from now on thread->mOutput is NULL 5574 out->hwDev->close_output_stream(out->hwDev, out->stream); 5575 delete out; 5576 } 5577 return NO_ERROR; 5578} 5579 5580status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5581{ 5582 Mutex::Autolock _l(mLock); 5583 PlaybackThread *thread = checkPlaybackThread_l(output); 5584 5585 if (thread == NULL) { 5586 return BAD_VALUE; 5587 } 5588 5589 ALOGV("suspendOutput() %d", output); 5590 thread->suspend(); 5591 5592 return NO_ERROR; 5593} 5594 5595status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5596{ 5597 Mutex::Autolock _l(mLock); 5598 PlaybackThread *thread = checkPlaybackThread_l(output); 5599 5600 if (thread == NULL) { 5601 return BAD_VALUE; 5602 } 5603 5604 ALOGV("restoreOutput() %d", output); 5605 5606 thread->restore(); 5607 5608 return NO_ERROR; 5609} 5610 5611audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5612 uint32_t *pSamplingRate, 5613 audio_format_t *pFormat, 5614 uint32_t *pChannels, 5615 audio_in_acoustics_t acoustics) 5616{ 5617 status_t status; 5618 RecordThread *thread = NULL; 5619 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5620 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5621 uint32_t channels = pChannels ? *pChannels : 0; 5622 uint32_t reqSamplingRate = samplingRate; 5623 audio_format_t reqFormat = format; 5624 uint32_t reqChannels = channels; 5625 audio_stream_in_t *inStream; 5626 audio_hw_device_t *inHwDev; 5627 5628 if (pDevices == NULL || *pDevices == 0) { 5629 return 0; 5630 } 5631 5632 Mutex::Autolock _l(mLock); 5633 5634 inHwDev = findSuitableHwDev_l(*pDevices); 5635 if (inHwDev == NULL) 5636 return 0; 5637 5638 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5639 &channels, &samplingRate, 5640 acoustics, 5641 &inStream); 5642 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5643 inStream, 5644 samplingRate, 5645 format, 5646 channels, 5647 acoustics, 5648 status); 5649 5650 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5651 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5652 // or stereo to mono conversions on 16 bit PCM inputs. 5653 if (inStream == NULL && status == BAD_VALUE && 5654 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5655 (samplingRate <= 2 * reqSamplingRate) && 5656 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5657 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5658 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5659 &channels, &samplingRate, 5660 acoustics, 5661 &inStream); 5662 } 5663 5664 if (inStream != NULL) { 5665 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5666 5667 audio_io_handle_t id = nextUniqueId(); 5668 // Start record thread 5669 // RecorThread require both input and output device indication to forward to audio 5670 // pre processing modules 5671 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5672 thread = new RecordThread(this, 5673 input, 5674 reqSamplingRate, 5675 reqChannels, 5676 id, 5677 device); 5678 mRecordThreads.add(id, thread); 5679 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5680 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5681 if (pFormat != NULL) *pFormat = format; 5682 if (pChannels != NULL) *pChannels = reqChannels; 5683 5684 input->stream->common.standby(&input->stream->common); 5685 5686 // notify client processes of the new input creation 5687 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5688 return id; 5689 } 5690 5691 return 0; 5692} 5693 5694status_t AudioFlinger::closeInput(audio_io_handle_t input) 5695{ 5696 // keep strong reference on the record thread so that 5697 // it is not destroyed while exit() is executed 5698 sp<RecordThread> thread; 5699 { 5700 Mutex::Autolock _l(mLock); 5701 thread = checkRecordThread_l(input); 5702 if (thread == NULL) { 5703 return BAD_VALUE; 5704 } 5705 5706 ALOGV("closeInput() %d", input); 5707 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5708 mRecordThreads.removeItem(input); 5709 } 5710 thread->exit(); 5711 // The thread entity (active unit of execution) is no longer running here, 5712 // but the ThreadBase container still exists. 5713 5714 AudioStreamIn *in = thread->clearInput(); 5715 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5716 // from now on thread->mInput is NULL 5717 in->hwDev->close_input_stream(in->hwDev, in->stream); 5718 delete in; 5719 5720 return NO_ERROR; 5721} 5722 5723status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5724{ 5725 Mutex::Autolock _l(mLock); 5726 MixerThread *dstThread = checkMixerThread_l(output); 5727 if (dstThread == NULL) { 5728 ALOGW("setStreamOutput() bad output id %d", output); 5729 return BAD_VALUE; 5730 } 5731 5732 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5733 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5734 5735 dstThread->setStreamValid(stream, true); 5736 5737 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5738 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5739 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5740 MixerThread *srcThread = (MixerThread *)thread; 5741 srcThread->setStreamValid(stream, false); 5742 srcThread->invalidateTracks(stream); 5743 } 5744 } 5745 5746 return NO_ERROR; 5747} 5748 5749 5750int AudioFlinger::newAudioSessionId() 5751{ 5752 return nextUniqueId(); 5753} 5754 5755void AudioFlinger::acquireAudioSessionId(int audioSession) 5756{ 5757 Mutex::Autolock _l(mLock); 5758 pid_t caller = IPCThreadState::self()->getCallingPid(); 5759 ALOGV("acquiring %d from %d", audioSession, caller); 5760 size_t num = mAudioSessionRefs.size(); 5761 for (size_t i = 0; i< num; i++) { 5762 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5763 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5764 ref->mCnt++; 5765 ALOGV(" incremented refcount to %d", ref->mCnt); 5766 return; 5767 } 5768 } 5769 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5770 ALOGV(" added new entry for %d", audioSession); 5771} 5772 5773void AudioFlinger::releaseAudioSessionId(int audioSession) 5774{ 5775 Mutex::Autolock _l(mLock); 5776 pid_t caller = IPCThreadState::self()->getCallingPid(); 5777 ALOGV("releasing %d from %d", audioSession, caller); 5778 size_t num = mAudioSessionRefs.size(); 5779 for (size_t i = 0; i< num; i++) { 5780 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5781 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5782 ref->mCnt--; 5783 ALOGV(" decremented refcount to %d", ref->mCnt); 5784 if (ref->mCnt == 0) { 5785 mAudioSessionRefs.removeAt(i); 5786 delete ref; 5787 purgeStaleEffects_l(); 5788 } 5789 return; 5790 } 5791 } 5792 ALOGW("session id %d not found for pid %d", audioSession, caller); 5793} 5794 5795void AudioFlinger::purgeStaleEffects_l() { 5796 5797 ALOGV("purging stale effects"); 5798 5799 Vector< sp<EffectChain> > chains; 5800 5801 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5802 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5803 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5804 sp<EffectChain> ec = t->mEffectChains[j]; 5805 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5806 chains.push(ec); 5807 } 5808 } 5809 } 5810 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5811 sp<RecordThread> t = mRecordThreads.valueAt(i); 5812 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5813 sp<EffectChain> ec = t->mEffectChains[j]; 5814 chains.push(ec); 5815 } 5816 } 5817 5818 for (size_t i = 0; i < chains.size(); i++) { 5819 sp<EffectChain> ec = chains[i]; 5820 int sessionid = ec->sessionId(); 5821 sp<ThreadBase> t = ec->mThread.promote(); 5822 if (t == 0) { 5823 continue; 5824 } 5825 size_t numsessionrefs = mAudioSessionRefs.size(); 5826 bool found = false; 5827 for (size_t k = 0; k < numsessionrefs; k++) { 5828 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5829 if (ref->mSessionid == sessionid) { 5830 ALOGV(" session %d still exists for %d with %d refs", 5831 sessionid, ref->mPid, ref->mCnt); 5832 found = true; 5833 break; 5834 } 5835 } 5836 if (!found) { 5837 // remove all effects from the chain 5838 while (ec->mEffects.size()) { 5839 sp<EffectModule> effect = ec->mEffects[0]; 5840 effect->unPin(); 5841 Mutex::Autolock _l (t->mLock); 5842 t->removeEffect_l(effect); 5843 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5844 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5845 if (handle != 0) { 5846 handle->mEffect.clear(); 5847 if (handle->mHasControl && handle->mEnabled) { 5848 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5849 } 5850 } 5851 } 5852 AudioSystem::unregisterEffect(effect->id()); 5853 } 5854 } 5855 } 5856 return; 5857} 5858 5859// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5860AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5861{ 5862 return mPlaybackThreads.valueFor(output).get(); 5863} 5864 5865// checkMixerThread_l() must be called with AudioFlinger::mLock held 5866AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5867{ 5868 PlaybackThread *thread = checkPlaybackThread_l(output); 5869 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5870} 5871 5872// checkRecordThread_l() must be called with AudioFlinger::mLock held 5873AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5874{ 5875 return mRecordThreads.valueFor(input).get(); 5876} 5877 5878uint32_t AudioFlinger::nextUniqueId() 5879{ 5880 return android_atomic_inc(&mNextUniqueId); 5881} 5882 5883AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5884{ 5885 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5886 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5887 AudioStreamOut *output = thread->getOutput(); 5888 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5889 return thread; 5890 } 5891 } 5892 return NULL; 5893} 5894 5895uint32_t AudioFlinger::primaryOutputDevice_l() const 5896{ 5897 PlaybackThread *thread = primaryPlaybackThread_l(); 5898 5899 if (thread == NULL) { 5900 return 0; 5901 } 5902 5903 return thread->device(); 5904} 5905 5906 5907// ---------------------------------------------------------------------------- 5908// Effect management 5909// ---------------------------------------------------------------------------- 5910 5911 5912status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5913{ 5914 Mutex::Autolock _l(mLock); 5915 return EffectQueryNumberEffects(numEffects); 5916} 5917 5918status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5919{ 5920 Mutex::Autolock _l(mLock); 5921 return EffectQueryEffect(index, descriptor); 5922} 5923 5924status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5925 effect_descriptor_t *descriptor) const 5926{ 5927 Mutex::Autolock _l(mLock); 5928 return EffectGetDescriptor(pUuid, descriptor); 5929} 5930 5931 5932sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5933 effect_descriptor_t *pDesc, 5934 const sp<IEffectClient>& effectClient, 5935 int32_t priority, 5936 audio_io_handle_t io, 5937 int sessionId, 5938 status_t *status, 5939 int *id, 5940 int *enabled) 5941{ 5942 status_t lStatus = NO_ERROR; 5943 sp<EffectHandle> handle; 5944 effect_descriptor_t desc; 5945 5946 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5947 pid, effectClient.get(), priority, sessionId, io); 5948 5949 if (pDesc == NULL) { 5950 lStatus = BAD_VALUE; 5951 goto Exit; 5952 } 5953 5954 // check audio settings permission for global effects 5955 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5956 lStatus = PERMISSION_DENIED; 5957 goto Exit; 5958 } 5959 5960 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5961 // that can only be created by audio policy manager (running in same process) 5962 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5963 lStatus = PERMISSION_DENIED; 5964 goto Exit; 5965 } 5966 5967 if (io == 0) { 5968 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5969 // output must be specified by AudioPolicyManager when using session 5970 // AUDIO_SESSION_OUTPUT_STAGE 5971 lStatus = BAD_VALUE; 5972 goto Exit; 5973 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5974 // if the output returned by getOutputForEffect() is removed before we lock the 5975 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5976 // and we will exit safely 5977 io = AudioSystem::getOutputForEffect(&desc); 5978 } 5979 } 5980 5981 { 5982 Mutex::Autolock _l(mLock); 5983 5984 5985 if (!EffectIsNullUuid(&pDesc->uuid)) { 5986 // if uuid is specified, request effect descriptor 5987 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5988 if (lStatus < 0) { 5989 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5990 goto Exit; 5991 } 5992 } else { 5993 // if uuid is not specified, look for an available implementation 5994 // of the required type in effect factory 5995 if (EffectIsNullUuid(&pDesc->type)) { 5996 ALOGW("createEffect() no effect type"); 5997 lStatus = BAD_VALUE; 5998 goto Exit; 5999 } 6000 uint32_t numEffects = 0; 6001 effect_descriptor_t d; 6002 d.flags = 0; // prevent compiler warning 6003 bool found = false; 6004 6005 lStatus = EffectQueryNumberEffects(&numEffects); 6006 if (lStatus < 0) { 6007 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 6008 goto Exit; 6009 } 6010 for (uint32_t i = 0; i < numEffects; i++) { 6011 lStatus = EffectQueryEffect(i, &desc); 6012 if (lStatus < 0) { 6013 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 6014 continue; 6015 } 6016 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 6017 // If matching type found save effect descriptor. If the session is 6018 // 0 and the effect is not auxiliary, continue enumeration in case 6019 // an auxiliary version of this effect type is available 6020 found = true; 6021 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6022 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6023 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6024 break; 6025 } 6026 } 6027 } 6028 if (!found) { 6029 lStatus = BAD_VALUE; 6030 ALOGW("createEffect() effect not found"); 6031 goto Exit; 6032 } 6033 // For same effect type, chose auxiliary version over insert version if 6034 // connect to output mix (Compliance to OpenSL ES) 6035 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6036 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6037 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6038 } 6039 } 6040 6041 // Do not allow auxiliary effects on a session different from 0 (output mix) 6042 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6043 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6044 lStatus = INVALID_OPERATION; 6045 goto Exit; 6046 } 6047 6048 // check recording permission for visualizer 6049 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6050 !recordingAllowed()) { 6051 lStatus = PERMISSION_DENIED; 6052 goto Exit; 6053 } 6054 6055 // return effect descriptor 6056 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6057 6058 // If output is not specified try to find a matching audio session ID in one of the 6059 // output threads. 6060 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6061 // because of code checking output when entering the function. 6062 // Note: io is never 0 when creating an effect on an input 6063 if (io == 0) { 6064 // look for the thread where the specified audio session is present 6065 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6066 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6067 io = mPlaybackThreads.keyAt(i); 6068 break; 6069 } 6070 } 6071 if (io == 0) { 6072 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6073 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6074 io = mRecordThreads.keyAt(i); 6075 break; 6076 } 6077 } 6078 } 6079 // If no output thread contains the requested session ID, default to 6080 // first output. The effect chain will be moved to the correct output 6081 // thread when a track with the same session ID is created 6082 if (io == 0 && mPlaybackThreads.size()) { 6083 io = mPlaybackThreads.keyAt(0); 6084 } 6085 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6086 } 6087 ThreadBase *thread = checkRecordThread_l(io); 6088 if (thread == NULL) { 6089 thread = checkPlaybackThread_l(io); 6090 if (thread == NULL) { 6091 ALOGE("createEffect() unknown output thread"); 6092 lStatus = BAD_VALUE; 6093 goto Exit; 6094 } 6095 } 6096 6097 sp<Client> client = registerPid_l(pid); 6098 6099 // create effect on selected output thread 6100 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6101 &desc, enabled, &lStatus); 6102 if (handle != 0 && id != NULL) { 6103 *id = handle->id(); 6104 } 6105 } 6106 6107Exit: 6108 if (status != NULL) { 6109 *status = lStatus; 6110 } 6111 return handle; 6112} 6113 6114status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6115 audio_io_handle_t dstOutput) 6116{ 6117 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6118 sessionId, srcOutput, dstOutput); 6119 Mutex::Autolock _l(mLock); 6120 if (srcOutput == dstOutput) { 6121 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6122 return NO_ERROR; 6123 } 6124 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6125 if (srcThread == NULL) { 6126 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6127 return BAD_VALUE; 6128 } 6129 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6130 if (dstThread == NULL) { 6131 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6132 return BAD_VALUE; 6133 } 6134 6135 Mutex::Autolock _dl(dstThread->mLock); 6136 Mutex::Autolock _sl(srcThread->mLock); 6137 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6138 6139 return NO_ERROR; 6140} 6141 6142// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6143status_t AudioFlinger::moveEffectChain_l(int sessionId, 6144 AudioFlinger::PlaybackThread *srcThread, 6145 AudioFlinger::PlaybackThread *dstThread, 6146 bool reRegister) 6147{ 6148 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6149 sessionId, srcThread, dstThread); 6150 6151 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6152 if (chain == 0) { 6153 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6154 sessionId, srcThread); 6155 return INVALID_OPERATION; 6156 } 6157 6158 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6159 // so that a new chain is created with correct parameters when first effect is added. This is 6160 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6161 // removed. 6162 srcThread->removeEffectChain_l(chain); 6163 6164 // transfer all effects one by one so that new effect chain is created on new thread with 6165 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6166 audio_io_handle_t dstOutput = dstThread->id(); 6167 sp<EffectChain> dstChain; 6168 uint32_t strategy = 0; // prevent compiler warning 6169 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6170 while (effect != 0) { 6171 srcThread->removeEffect_l(effect); 6172 dstThread->addEffect_l(effect); 6173 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6174 if (effect->state() == EffectModule::ACTIVE || 6175 effect->state() == EffectModule::STOPPING) { 6176 effect->start(); 6177 } 6178 // if the move request is not received from audio policy manager, the effect must be 6179 // re-registered with the new strategy and output 6180 if (dstChain == 0) { 6181 dstChain = effect->chain().promote(); 6182 if (dstChain == 0) { 6183 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6184 srcThread->addEffect_l(effect); 6185 return NO_INIT; 6186 } 6187 strategy = dstChain->strategy(); 6188 } 6189 if (reRegister) { 6190 AudioSystem::unregisterEffect(effect->id()); 6191 AudioSystem::registerEffect(&effect->desc(), 6192 dstOutput, 6193 strategy, 6194 sessionId, 6195 effect->id()); 6196 } 6197 effect = chain->getEffectFromId_l(0); 6198 } 6199 6200 return NO_ERROR; 6201} 6202 6203 6204// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6205sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6206 const sp<AudioFlinger::Client>& client, 6207 const sp<IEffectClient>& effectClient, 6208 int32_t priority, 6209 int sessionId, 6210 effect_descriptor_t *desc, 6211 int *enabled, 6212 status_t *status 6213 ) 6214{ 6215 sp<EffectModule> effect; 6216 sp<EffectHandle> handle; 6217 status_t lStatus; 6218 sp<EffectChain> chain; 6219 bool chainCreated = false; 6220 bool effectCreated = false; 6221 bool effectRegistered = false; 6222 6223 lStatus = initCheck(); 6224 if (lStatus != NO_ERROR) { 6225 ALOGW("createEffect_l() Audio driver not initialized."); 6226 goto Exit; 6227 } 6228 6229 // Do not allow effects with session ID 0 on direct output or duplicating threads 6230 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6231 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6232 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6233 desc->name, sessionId); 6234 lStatus = BAD_VALUE; 6235 goto Exit; 6236 } 6237 // Only Pre processor effects are allowed on input threads and only on input threads 6238 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6239 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6240 desc->name, desc->flags, mType); 6241 lStatus = BAD_VALUE; 6242 goto Exit; 6243 } 6244 6245 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6246 6247 { // scope for mLock 6248 Mutex::Autolock _l(mLock); 6249 6250 // check for existing effect chain with the requested audio session 6251 chain = getEffectChain_l(sessionId); 6252 if (chain == 0) { 6253 // create a new chain for this session 6254 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6255 chain = new EffectChain(this, sessionId); 6256 addEffectChain_l(chain); 6257 chain->setStrategy(getStrategyForSession_l(sessionId)); 6258 chainCreated = true; 6259 } else { 6260 effect = chain->getEffectFromDesc_l(desc); 6261 } 6262 6263 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6264 6265 if (effect == 0) { 6266 int id = mAudioFlinger->nextUniqueId(); 6267 // Check CPU and memory usage 6268 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6269 if (lStatus != NO_ERROR) { 6270 goto Exit; 6271 } 6272 effectRegistered = true; 6273 // create a new effect module if none present in the chain 6274 effect = new EffectModule(this, chain, desc, id, sessionId); 6275 lStatus = effect->status(); 6276 if (lStatus != NO_ERROR) { 6277 goto Exit; 6278 } 6279 lStatus = chain->addEffect_l(effect); 6280 if (lStatus != NO_ERROR) { 6281 goto Exit; 6282 } 6283 effectCreated = true; 6284 6285 effect->setDevice(mDevice); 6286 effect->setMode(mAudioFlinger->getMode()); 6287 } 6288 // create effect handle and connect it to effect module 6289 handle = new EffectHandle(effect, client, effectClient, priority); 6290 lStatus = effect->addHandle(handle); 6291 if (enabled != NULL) { 6292 *enabled = (int)effect->isEnabled(); 6293 } 6294 } 6295 6296Exit: 6297 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6298 Mutex::Autolock _l(mLock); 6299 if (effectCreated) { 6300 chain->removeEffect_l(effect); 6301 } 6302 if (effectRegistered) { 6303 AudioSystem::unregisterEffect(effect->id()); 6304 } 6305 if (chainCreated) { 6306 removeEffectChain_l(chain); 6307 } 6308 handle.clear(); 6309 } 6310 6311 if (status != NULL) { 6312 *status = lStatus; 6313 } 6314 return handle; 6315} 6316 6317sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6318{ 6319 sp<EffectChain> chain = getEffectChain_l(sessionId); 6320 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6321} 6322 6323// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6324// PlaybackThread::mLock held 6325status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6326{ 6327 // check for existing effect chain with the requested audio session 6328 int sessionId = effect->sessionId(); 6329 sp<EffectChain> chain = getEffectChain_l(sessionId); 6330 bool chainCreated = false; 6331 6332 if (chain == 0) { 6333 // create a new chain for this session 6334 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6335 chain = new EffectChain(this, sessionId); 6336 addEffectChain_l(chain); 6337 chain->setStrategy(getStrategyForSession_l(sessionId)); 6338 chainCreated = true; 6339 } 6340 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6341 6342 if (chain->getEffectFromId_l(effect->id()) != 0) { 6343 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6344 this, effect->desc().name, chain.get()); 6345 return BAD_VALUE; 6346 } 6347 6348 status_t status = chain->addEffect_l(effect); 6349 if (status != NO_ERROR) { 6350 if (chainCreated) { 6351 removeEffectChain_l(chain); 6352 } 6353 return status; 6354 } 6355 6356 effect->setDevice(mDevice); 6357 effect->setMode(mAudioFlinger->getMode()); 6358 return NO_ERROR; 6359} 6360 6361void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6362 6363 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6364 effect_descriptor_t desc = effect->desc(); 6365 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6366 detachAuxEffect_l(effect->id()); 6367 } 6368 6369 sp<EffectChain> chain = effect->chain().promote(); 6370 if (chain != 0) { 6371 // remove effect chain if removing last effect 6372 if (chain->removeEffect_l(effect) == 0) { 6373 removeEffectChain_l(chain); 6374 } 6375 } else { 6376 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6377 } 6378} 6379 6380void AudioFlinger::ThreadBase::lockEffectChains_l( 6381 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6382{ 6383 effectChains = mEffectChains; 6384 for (size_t i = 0; i < mEffectChains.size(); i++) { 6385 mEffectChains[i]->lock(); 6386 } 6387} 6388 6389void AudioFlinger::ThreadBase::unlockEffectChains( 6390 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6391{ 6392 for (size_t i = 0; i < effectChains.size(); i++) { 6393 effectChains[i]->unlock(); 6394 } 6395} 6396 6397sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6398{ 6399 Mutex::Autolock _l(mLock); 6400 return getEffectChain_l(sessionId); 6401} 6402 6403sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6404{ 6405 size_t size = mEffectChains.size(); 6406 for (size_t i = 0; i < size; i++) { 6407 if (mEffectChains[i]->sessionId() == sessionId) { 6408 return mEffectChains[i]; 6409 } 6410 } 6411 return 0; 6412} 6413 6414void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6415{ 6416 Mutex::Autolock _l(mLock); 6417 size_t size = mEffectChains.size(); 6418 for (size_t i = 0; i < size; i++) { 6419 mEffectChains[i]->setMode_l(mode); 6420 } 6421} 6422 6423void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6424 const wp<EffectHandle>& handle, 6425 bool unpinIfLast) { 6426 6427 Mutex::Autolock _l(mLock); 6428 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6429 // delete the effect module if removing last handle on it 6430 if (effect->removeHandle(handle) == 0) { 6431 if (!effect->isPinned() || unpinIfLast) { 6432 removeEffect_l(effect); 6433 AudioSystem::unregisterEffect(effect->id()); 6434 } 6435 } 6436} 6437 6438status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6439{ 6440 int session = chain->sessionId(); 6441 int16_t *buffer = mMixBuffer; 6442 bool ownsBuffer = false; 6443 6444 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6445 if (session > 0) { 6446 // Only one effect chain can be present in direct output thread and it uses 6447 // the mix buffer as input 6448 if (mType != DIRECT) { 6449 size_t numSamples = mFrameCount * mChannelCount; 6450 buffer = new int16_t[numSamples]; 6451 memset(buffer, 0, numSamples * sizeof(int16_t)); 6452 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6453 ownsBuffer = true; 6454 } 6455 6456 // Attach all tracks with same session ID to this chain. 6457 for (size_t i = 0; i < mTracks.size(); ++i) { 6458 sp<Track> track = mTracks[i]; 6459 if (session == track->sessionId()) { 6460 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6461 track->setMainBuffer(buffer); 6462 chain->incTrackCnt(); 6463 } 6464 } 6465 6466 // indicate all active tracks in the chain 6467 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6468 sp<Track> track = mActiveTracks[i].promote(); 6469 if (track == 0) continue; 6470 if (session == track->sessionId()) { 6471 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6472 chain->incActiveTrackCnt(); 6473 } 6474 } 6475 } 6476 6477 chain->setInBuffer(buffer, ownsBuffer); 6478 chain->setOutBuffer(mMixBuffer); 6479 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6480 // chains list in order to be processed last as it contains output stage effects 6481 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6482 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6483 // after track specific effects and before output stage 6484 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6485 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6486 // Effect chain for other sessions are inserted at beginning of effect 6487 // chains list to be processed before output mix effects. Relative order between other 6488 // sessions is not important 6489 size_t size = mEffectChains.size(); 6490 size_t i = 0; 6491 for (i = 0; i < size; i++) { 6492 if (mEffectChains[i]->sessionId() < session) break; 6493 } 6494 mEffectChains.insertAt(chain, i); 6495 checkSuspendOnAddEffectChain_l(chain); 6496 6497 return NO_ERROR; 6498} 6499 6500size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6501{ 6502 int session = chain->sessionId(); 6503 6504 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6505 6506 for (size_t i = 0; i < mEffectChains.size(); i++) { 6507 if (chain == mEffectChains[i]) { 6508 mEffectChains.removeAt(i); 6509 // detach all active tracks from the chain 6510 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6511 sp<Track> track = mActiveTracks[i].promote(); 6512 if (track == 0) continue; 6513 if (session == track->sessionId()) { 6514 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6515 chain.get(), session); 6516 chain->decActiveTrackCnt(); 6517 } 6518 } 6519 6520 // detach all tracks with same session ID from this chain 6521 for (size_t i = 0; i < mTracks.size(); ++i) { 6522 sp<Track> track = mTracks[i]; 6523 if (session == track->sessionId()) { 6524 track->setMainBuffer(mMixBuffer); 6525 chain->decTrackCnt(); 6526 } 6527 } 6528 break; 6529 } 6530 } 6531 return mEffectChains.size(); 6532} 6533 6534status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6535 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6536{ 6537 Mutex::Autolock _l(mLock); 6538 return attachAuxEffect_l(track, EffectId); 6539} 6540 6541status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6542 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6543{ 6544 status_t status = NO_ERROR; 6545 6546 if (EffectId == 0) { 6547 track->setAuxBuffer(0, NULL); 6548 } else { 6549 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6550 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6551 if (effect != 0) { 6552 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6553 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6554 } else { 6555 status = INVALID_OPERATION; 6556 } 6557 } else { 6558 status = BAD_VALUE; 6559 } 6560 } 6561 return status; 6562} 6563 6564void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6565{ 6566 for (size_t i = 0; i < mTracks.size(); ++i) { 6567 sp<Track> track = mTracks[i]; 6568 if (track->auxEffectId() == effectId) { 6569 attachAuxEffect_l(track, 0); 6570 } 6571 } 6572} 6573 6574status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6575{ 6576 // only one chain per input thread 6577 if (mEffectChains.size() != 0) { 6578 return INVALID_OPERATION; 6579 } 6580 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6581 6582 chain->setInBuffer(NULL); 6583 chain->setOutBuffer(NULL); 6584 6585 checkSuspendOnAddEffectChain_l(chain); 6586 6587 mEffectChains.add(chain); 6588 6589 return NO_ERROR; 6590} 6591 6592size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6593{ 6594 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6595 ALOGW_IF(mEffectChains.size() != 1, 6596 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6597 chain.get(), mEffectChains.size(), this); 6598 if (mEffectChains.size() == 1) { 6599 mEffectChains.removeAt(0); 6600 } 6601 return 0; 6602} 6603 6604// ---------------------------------------------------------------------------- 6605// EffectModule implementation 6606// ---------------------------------------------------------------------------- 6607 6608#undef LOG_TAG 6609#define LOG_TAG "AudioFlinger::EffectModule" 6610 6611AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6612 const wp<AudioFlinger::EffectChain>& chain, 6613 effect_descriptor_t *desc, 6614 int id, 6615 int sessionId) 6616 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6617 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6618{ 6619 ALOGV("Constructor %p", this); 6620 int lStatus; 6621 if (thread == NULL) { 6622 return; 6623 } 6624 6625 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6626 6627 // create effect engine from effect factory 6628 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6629 6630 if (mStatus != NO_ERROR) { 6631 return; 6632 } 6633 lStatus = init(); 6634 if (lStatus < 0) { 6635 mStatus = lStatus; 6636 goto Error; 6637 } 6638 6639 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6640 mPinned = true; 6641 } 6642 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6643 return; 6644Error: 6645 EffectRelease(mEffectInterface); 6646 mEffectInterface = NULL; 6647 ALOGV("Constructor Error %d", mStatus); 6648} 6649 6650AudioFlinger::EffectModule::~EffectModule() 6651{ 6652 ALOGV("Destructor %p", this); 6653 if (mEffectInterface != NULL) { 6654 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6655 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6656 sp<ThreadBase> thread = mThread.promote(); 6657 if (thread != 0) { 6658 audio_stream_t *stream = thread->stream(); 6659 if (stream != NULL) { 6660 stream->remove_audio_effect(stream, mEffectInterface); 6661 } 6662 } 6663 } 6664 // release effect engine 6665 EffectRelease(mEffectInterface); 6666 } 6667} 6668 6669status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6670{ 6671 status_t status; 6672 6673 Mutex::Autolock _l(mLock); 6674 int priority = handle->priority(); 6675 size_t size = mHandles.size(); 6676 sp<EffectHandle> h; 6677 size_t i; 6678 for (i = 0; i < size; i++) { 6679 h = mHandles[i].promote(); 6680 if (h == 0) continue; 6681 if (h->priority() <= priority) break; 6682 } 6683 // if inserted in first place, move effect control from previous owner to this handle 6684 if (i == 0) { 6685 bool enabled = false; 6686 if (h != 0) { 6687 enabled = h->enabled(); 6688 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6689 } 6690 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6691 status = NO_ERROR; 6692 } else { 6693 status = ALREADY_EXISTS; 6694 } 6695 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6696 mHandles.insertAt(handle, i); 6697 return status; 6698} 6699 6700size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6701{ 6702 Mutex::Autolock _l(mLock); 6703 size_t size = mHandles.size(); 6704 size_t i; 6705 for (i = 0; i < size; i++) { 6706 if (mHandles[i] == handle) break; 6707 } 6708 if (i == size) { 6709 return size; 6710 } 6711 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6712 6713 bool enabled = false; 6714 EffectHandle *hdl = handle.unsafe_get(); 6715 if (hdl != NULL) { 6716 ALOGV("removeHandle() unsafe_get OK"); 6717 enabled = hdl->enabled(); 6718 } 6719 mHandles.removeAt(i); 6720 size = mHandles.size(); 6721 // if removed from first place, move effect control from this handle to next in line 6722 if (i == 0 && size != 0) { 6723 sp<EffectHandle> h = mHandles[0].promote(); 6724 if (h != 0) { 6725 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6726 } 6727 } 6728 6729 // Prevent calls to process() and other functions on effect interface from now on. 6730 // The effect engine will be released by the destructor when the last strong reference on 6731 // this object is released which can happen after next process is called. 6732 if (size == 0 && !mPinned) { 6733 mState = DESTROYED; 6734 } 6735 6736 return size; 6737} 6738 6739sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6740{ 6741 Mutex::Autolock _l(mLock); 6742 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6743} 6744 6745void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6746{ 6747 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6748 // keep a strong reference on this EffectModule to avoid calling the 6749 // destructor before we exit 6750 sp<EffectModule> keep(this); 6751 { 6752 sp<ThreadBase> thread = mThread.promote(); 6753 if (thread != 0) { 6754 thread->disconnectEffect(keep, handle, unpinIfLast); 6755 } 6756 } 6757} 6758 6759void AudioFlinger::EffectModule::updateState() { 6760 Mutex::Autolock _l(mLock); 6761 6762 switch (mState) { 6763 case RESTART: 6764 reset_l(); 6765 // FALL THROUGH 6766 6767 case STARTING: 6768 // clear auxiliary effect input buffer for next accumulation 6769 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6770 memset(mConfig.inputCfg.buffer.raw, 6771 0, 6772 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6773 } 6774 start_l(); 6775 mState = ACTIVE; 6776 break; 6777 case STOPPING: 6778 stop_l(); 6779 mDisableWaitCnt = mMaxDisableWaitCnt; 6780 mState = STOPPED; 6781 break; 6782 case STOPPED: 6783 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6784 // turn off sequence. 6785 if (--mDisableWaitCnt == 0) { 6786 reset_l(); 6787 mState = IDLE; 6788 } 6789 break; 6790 default: //IDLE , ACTIVE, DESTROYED 6791 break; 6792 } 6793} 6794 6795void AudioFlinger::EffectModule::process() 6796{ 6797 Mutex::Autolock _l(mLock); 6798 6799 if (mState == DESTROYED || mEffectInterface == NULL || 6800 mConfig.inputCfg.buffer.raw == NULL || 6801 mConfig.outputCfg.buffer.raw == NULL) { 6802 return; 6803 } 6804 6805 if (isProcessEnabled()) { 6806 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6807 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6808 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6809 mConfig.inputCfg.buffer.s32, 6810 mConfig.inputCfg.buffer.frameCount/2); 6811 } 6812 6813 // do the actual processing in the effect engine 6814 int ret = (*mEffectInterface)->process(mEffectInterface, 6815 &mConfig.inputCfg.buffer, 6816 &mConfig.outputCfg.buffer); 6817 6818 // force transition to IDLE state when engine is ready 6819 if (mState == STOPPED && ret == -ENODATA) { 6820 mDisableWaitCnt = 1; 6821 } 6822 6823 // clear auxiliary effect input buffer for next accumulation 6824 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6825 memset(mConfig.inputCfg.buffer.raw, 0, 6826 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6827 } 6828 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6829 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6830 // If an insert effect is idle and input buffer is different from output buffer, 6831 // accumulate input onto output 6832 sp<EffectChain> chain = mChain.promote(); 6833 if (chain != 0 && chain->activeTrackCnt() != 0) { 6834 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6835 int16_t *in = mConfig.inputCfg.buffer.s16; 6836 int16_t *out = mConfig.outputCfg.buffer.s16; 6837 for (size_t i = 0; i < frameCnt; i++) { 6838 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6839 } 6840 } 6841 } 6842} 6843 6844void AudioFlinger::EffectModule::reset_l() 6845{ 6846 if (mEffectInterface == NULL) { 6847 return; 6848 } 6849 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6850} 6851 6852status_t AudioFlinger::EffectModule::configure() 6853{ 6854 uint32_t channels; 6855 if (mEffectInterface == NULL) { 6856 return NO_INIT; 6857 } 6858 6859 sp<ThreadBase> thread = mThread.promote(); 6860 if (thread == 0) { 6861 return DEAD_OBJECT; 6862 } 6863 6864 // TODO: handle configuration of effects replacing track process 6865 if (thread->channelCount() == 1) { 6866 channels = AUDIO_CHANNEL_OUT_MONO; 6867 } else { 6868 channels = AUDIO_CHANNEL_OUT_STEREO; 6869 } 6870 6871 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6872 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6873 } else { 6874 mConfig.inputCfg.channels = channels; 6875 } 6876 mConfig.outputCfg.channels = channels; 6877 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6878 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6879 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6880 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6881 mConfig.inputCfg.bufferProvider.cookie = NULL; 6882 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6883 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6884 mConfig.outputCfg.bufferProvider.cookie = NULL; 6885 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6886 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6887 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6888 // Insert effect: 6889 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6890 // always overwrites output buffer: input buffer == output buffer 6891 // - in other sessions: 6892 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6893 // other effect: overwrites output buffer: input buffer == output buffer 6894 // Auxiliary effect: 6895 // accumulates in output buffer: input buffer != output buffer 6896 // Therefore: accumulate <=> input buffer != output buffer 6897 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6898 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6899 } else { 6900 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6901 } 6902 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6903 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6904 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6905 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6906 6907 ALOGV("configure() %p thread %p buffer %p framecount %d", 6908 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6909 6910 status_t cmdStatus; 6911 uint32_t size = sizeof(int); 6912 status_t status = (*mEffectInterface)->command(mEffectInterface, 6913 EFFECT_CMD_SET_CONFIG, 6914 sizeof(effect_config_t), 6915 &mConfig, 6916 &size, 6917 &cmdStatus); 6918 if (status == 0) { 6919 status = cmdStatus; 6920 } 6921 6922 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6923 (1000 * mConfig.outputCfg.buffer.frameCount); 6924 6925 return status; 6926} 6927 6928status_t AudioFlinger::EffectModule::init() 6929{ 6930 Mutex::Autolock _l(mLock); 6931 if (mEffectInterface == NULL) { 6932 return NO_INIT; 6933 } 6934 status_t cmdStatus; 6935 uint32_t size = sizeof(status_t); 6936 status_t status = (*mEffectInterface)->command(mEffectInterface, 6937 EFFECT_CMD_INIT, 6938 0, 6939 NULL, 6940 &size, 6941 &cmdStatus); 6942 if (status == 0) { 6943 status = cmdStatus; 6944 } 6945 return status; 6946} 6947 6948status_t AudioFlinger::EffectModule::start() 6949{ 6950 Mutex::Autolock _l(mLock); 6951 return start_l(); 6952} 6953 6954status_t AudioFlinger::EffectModule::start_l() 6955{ 6956 if (mEffectInterface == NULL) { 6957 return NO_INIT; 6958 } 6959 status_t cmdStatus; 6960 uint32_t size = sizeof(status_t); 6961 status_t status = (*mEffectInterface)->command(mEffectInterface, 6962 EFFECT_CMD_ENABLE, 6963 0, 6964 NULL, 6965 &size, 6966 &cmdStatus); 6967 if (status == 0) { 6968 status = cmdStatus; 6969 } 6970 if (status == 0 && 6971 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6972 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6973 sp<ThreadBase> thread = mThread.promote(); 6974 if (thread != 0) { 6975 audio_stream_t *stream = thread->stream(); 6976 if (stream != NULL) { 6977 stream->add_audio_effect(stream, mEffectInterface); 6978 } 6979 } 6980 } 6981 return status; 6982} 6983 6984status_t AudioFlinger::EffectModule::stop() 6985{ 6986 Mutex::Autolock _l(mLock); 6987 return stop_l(); 6988} 6989 6990status_t AudioFlinger::EffectModule::stop_l() 6991{ 6992 if (mEffectInterface == NULL) { 6993 return NO_INIT; 6994 } 6995 status_t cmdStatus; 6996 uint32_t size = sizeof(status_t); 6997 status_t status = (*mEffectInterface)->command(mEffectInterface, 6998 EFFECT_CMD_DISABLE, 6999 0, 7000 NULL, 7001 &size, 7002 &cmdStatus); 7003 if (status == 0) { 7004 status = cmdStatus; 7005 } 7006 if (status == 0 && 7007 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 7008 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 7009 sp<ThreadBase> thread = mThread.promote(); 7010 if (thread != 0) { 7011 audio_stream_t *stream = thread->stream(); 7012 if (stream != NULL) { 7013 stream->remove_audio_effect(stream, mEffectInterface); 7014 } 7015 } 7016 } 7017 return status; 7018} 7019 7020status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7021 uint32_t cmdSize, 7022 void *pCmdData, 7023 uint32_t *replySize, 7024 void *pReplyData) 7025{ 7026 Mutex::Autolock _l(mLock); 7027// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7028 7029 if (mState == DESTROYED || mEffectInterface == NULL) { 7030 return NO_INIT; 7031 } 7032 status_t status = (*mEffectInterface)->command(mEffectInterface, 7033 cmdCode, 7034 cmdSize, 7035 pCmdData, 7036 replySize, 7037 pReplyData); 7038 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7039 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7040 for (size_t i = 1; i < mHandles.size(); i++) { 7041 sp<EffectHandle> h = mHandles[i].promote(); 7042 if (h != 0) { 7043 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7044 } 7045 } 7046 } 7047 return status; 7048} 7049 7050status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7051{ 7052 7053 Mutex::Autolock _l(mLock); 7054 ALOGV("setEnabled %p enabled %d", this, enabled); 7055 7056 if (enabled != isEnabled()) { 7057 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7058 if (enabled && status != NO_ERROR) { 7059 return status; 7060 } 7061 7062 switch (mState) { 7063 // going from disabled to enabled 7064 case IDLE: 7065 mState = STARTING; 7066 break; 7067 case STOPPED: 7068 mState = RESTART; 7069 break; 7070 case STOPPING: 7071 mState = ACTIVE; 7072 break; 7073 7074 // going from enabled to disabled 7075 case RESTART: 7076 mState = STOPPED; 7077 break; 7078 case STARTING: 7079 mState = IDLE; 7080 break; 7081 case ACTIVE: 7082 mState = STOPPING; 7083 break; 7084 case DESTROYED: 7085 return NO_ERROR; // simply ignore as we are being destroyed 7086 } 7087 for (size_t i = 1; i < mHandles.size(); i++) { 7088 sp<EffectHandle> h = mHandles[i].promote(); 7089 if (h != 0) { 7090 h->setEnabled(enabled); 7091 } 7092 } 7093 } 7094 return NO_ERROR; 7095} 7096 7097bool AudioFlinger::EffectModule::isEnabled() const 7098{ 7099 switch (mState) { 7100 case RESTART: 7101 case STARTING: 7102 case ACTIVE: 7103 return true; 7104 case IDLE: 7105 case STOPPING: 7106 case STOPPED: 7107 case DESTROYED: 7108 default: 7109 return false; 7110 } 7111} 7112 7113bool AudioFlinger::EffectModule::isProcessEnabled() const 7114{ 7115 switch (mState) { 7116 case RESTART: 7117 case ACTIVE: 7118 case STOPPING: 7119 case STOPPED: 7120 return true; 7121 case IDLE: 7122 case STARTING: 7123 case DESTROYED: 7124 default: 7125 return false; 7126 } 7127} 7128 7129status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7130{ 7131 Mutex::Autolock _l(mLock); 7132 status_t status = NO_ERROR; 7133 7134 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7135 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7136 if (isProcessEnabled() && 7137 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7138 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7139 status_t cmdStatus; 7140 uint32_t volume[2]; 7141 uint32_t *pVolume = NULL; 7142 uint32_t size = sizeof(volume); 7143 volume[0] = *left; 7144 volume[1] = *right; 7145 if (controller) { 7146 pVolume = volume; 7147 } 7148 status = (*mEffectInterface)->command(mEffectInterface, 7149 EFFECT_CMD_SET_VOLUME, 7150 size, 7151 volume, 7152 &size, 7153 pVolume); 7154 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7155 *left = volume[0]; 7156 *right = volume[1]; 7157 } 7158 } 7159 return status; 7160} 7161 7162status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7163{ 7164 Mutex::Autolock _l(mLock); 7165 status_t status = NO_ERROR; 7166 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7167 // audio pre processing modules on RecordThread can receive both output and 7168 // input device indication in the same call 7169 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7170 if (dev) { 7171 status_t cmdStatus; 7172 uint32_t size = sizeof(status_t); 7173 7174 status = (*mEffectInterface)->command(mEffectInterface, 7175 EFFECT_CMD_SET_DEVICE, 7176 sizeof(uint32_t), 7177 &dev, 7178 &size, 7179 &cmdStatus); 7180 if (status == NO_ERROR) { 7181 status = cmdStatus; 7182 } 7183 } 7184 dev = device & AUDIO_DEVICE_IN_ALL; 7185 if (dev) { 7186 status_t cmdStatus; 7187 uint32_t size = sizeof(status_t); 7188 7189 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7190 EFFECT_CMD_SET_INPUT_DEVICE, 7191 sizeof(uint32_t), 7192 &dev, 7193 &size, 7194 &cmdStatus); 7195 if (status2 == NO_ERROR) { 7196 status2 = cmdStatus; 7197 } 7198 if (status == NO_ERROR) { 7199 status = status2; 7200 } 7201 } 7202 } 7203 return status; 7204} 7205 7206status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7207{ 7208 Mutex::Autolock _l(mLock); 7209 status_t status = NO_ERROR; 7210 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7211 status_t cmdStatus; 7212 uint32_t size = sizeof(status_t); 7213 status = (*mEffectInterface)->command(mEffectInterface, 7214 EFFECT_CMD_SET_AUDIO_MODE, 7215 sizeof(audio_mode_t), 7216 &mode, 7217 &size, 7218 &cmdStatus); 7219 if (status == NO_ERROR) { 7220 status = cmdStatus; 7221 } 7222 } 7223 return status; 7224} 7225 7226void AudioFlinger::EffectModule::setSuspended(bool suspended) 7227{ 7228 Mutex::Autolock _l(mLock); 7229 mSuspended = suspended; 7230} 7231 7232bool AudioFlinger::EffectModule::suspended() const 7233{ 7234 Mutex::Autolock _l(mLock); 7235 return mSuspended; 7236} 7237 7238status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7239{ 7240 const size_t SIZE = 256; 7241 char buffer[SIZE]; 7242 String8 result; 7243 7244 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7245 result.append(buffer); 7246 7247 bool locked = tryLock(mLock); 7248 // failed to lock - AudioFlinger is probably deadlocked 7249 if (!locked) { 7250 result.append("\t\tCould not lock Fx mutex:\n"); 7251 } 7252 7253 result.append("\t\tSession Status State Engine:\n"); 7254 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7255 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7256 result.append(buffer); 7257 7258 result.append("\t\tDescriptor:\n"); 7259 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7260 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7261 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7262 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7263 result.append(buffer); 7264 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7265 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7266 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7267 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7268 result.append(buffer); 7269 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7270 mDescriptor.apiVersion, 7271 mDescriptor.flags); 7272 result.append(buffer); 7273 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7274 mDescriptor.name); 7275 result.append(buffer); 7276 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7277 mDescriptor.implementor); 7278 result.append(buffer); 7279 7280 result.append("\t\t- Input configuration:\n"); 7281 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7282 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7283 (uint32_t)mConfig.inputCfg.buffer.raw, 7284 mConfig.inputCfg.buffer.frameCount, 7285 mConfig.inputCfg.samplingRate, 7286 mConfig.inputCfg.channels, 7287 mConfig.inputCfg.format); 7288 result.append(buffer); 7289 7290 result.append("\t\t- Output configuration:\n"); 7291 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7292 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7293 (uint32_t)mConfig.outputCfg.buffer.raw, 7294 mConfig.outputCfg.buffer.frameCount, 7295 mConfig.outputCfg.samplingRate, 7296 mConfig.outputCfg.channels, 7297 mConfig.outputCfg.format); 7298 result.append(buffer); 7299 7300 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7301 result.append(buffer); 7302 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7303 for (size_t i = 0; i < mHandles.size(); ++i) { 7304 sp<EffectHandle> handle = mHandles[i].promote(); 7305 if (handle != 0) { 7306 handle->dump(buffer, SIZE); 7307 result.append(buffer); 7308 } 7309 } 7310 7311 result.append("\n"); 7312 7313 write(fd, result.string(), result.length()); 7314 7315 if (locked) { 7316 mLock.unlock(); 7317 } 7318 7319 return NO_ERROR; 7320} 7321 7322// ---------------------------------------------------------------------------- 7323// EffectHandle implementation 7324// ---------------------------------------------------------------------------- 7325 7326#undef LOG_TAG 7327#define LOG_TAG "AudioFlinger::EffectHandle" 7328 7329AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7330 const sp<AudioFlinger::Client>& client, 7331 const sp<IEffectClient>& effectClient, 7332 int32_t priority) 7333 : BnEffect(), 7334 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7335 mPriority(priority), mHasControl(false), mEnabled(false) 7336{ 7337 ALOGV("constructor %p", this); 7338 7339 if (client == 0) { 7340 return; 7341 } 7342 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7343 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7344 if (mCblkMemory != 0) { 7345 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7346 7347 if (mCblk != NULL) { 7348 new(mCblk) effect_param_cblk_t(); 7349 mBuffer = (uint8_t *)mCblk + bufOffset; 7350 } 7351 } else { 7352 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7353 return; 7354 } 7355} 7356 7357AudioFlinger::EffectHandle::~EffectHandle() 7358{ 7359 ALOGV("Destructor %p", this); 7360 disconnect(false); 7361 ALOGV("Destructor DONE %p", this); 7362} 7363 7364status_t AudioFlinger::EffectHandle::enable() 7365{ 7366 ALOGV("enable %p", this); 7367 if (!mHasControl) return INVALID_OPERATION; 7368 if (mEffect == 0) return DEAD_OBJECT; 7369 7370 if (mEnabled) { 7371 return NO_ERROR; 7372 } 7373 7374 mEnabled = true; 7375 7376 sp<ThreadBase> thread = mEffect->thread().promote(); 7377 if (thread != 0) { 7378 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7379 } 7380 7381 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7382 if (mEffect->suspended()) { 7383 return NO_ERROR; 7384 } 7385 7386 status_t status = mEffect->setEnabled(true); 7387 if (status != NO_ERROR) { 7388 if (thread != 0) { 7389 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7390 } 7391 mEnabled = false; 7392 } 7393 return status; 7394} 7395 7396status_t AudioFlinger::EffectHandle::disable() 7397{ 7398 ALOGV("disable %p", this); 7399 if (!mHasControl) return INVALID_OPERATION; 7400 if (mEffect == 0) return DEAD_OBJECT; 7401 7402 if (!mEnabled) { 7403 return NO_ERROR; 7404 } 7405 mEnabled = false; 7406 7407 if (mEffect->suspended()) { 7408 return NO_ERROR; 7409 } 7410 7411 status_t status = mEffect->setEnabled(false); 7412 7413 sp<ThreadBase> thread = mEffect->thread().promote(); 7414 if (thread != 0) { 7415 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7416 } 7417 7418 return status; 7419} 7420 7421void AudioFlinger::EffectHandle::disconnect() 7422{ 7423 disconnect(true); 7424} 7425 7426void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7427{ 7428 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7429 if (mEffect == 0) { 7430 return; 7431 } 7432 mEffect->disconnect(this, unpinIfLast); 7433 7434 if (mHasControl && mEnabled) { 7435 sp<ThreadBase> thread = mEffect->thread().promote(); 7436 if (thread != 0) { 7437 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7438 } 7439 } 7440 7441 // release sp on module => module destructor can be called now 7442 mEffect.clear(); 7443 if (mClient != 0) { 7444 if (mCblk != NULL) { 7445 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7446 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7447 } 7448 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7449 // Client destructor must run with AudioFlinger mutex locked 7450 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7451 mClient.clear(); 7452 } 7453} 7454 7455status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7456 uint32_t cmdSize, 7457 void *pCmdData, 7458 uint32_t *replySize, 7459 void *pReplyData) 7460{ 7461// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7462// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7463 7464 // only get parameter command is permitted for applications not controlling the effect 7465 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7466 return INVALID_OPERATION; 7467 } 7468 if (mEffect == 0) return DEAD_OBJECT; 7469 if (mClient == 0) return INVALID_OPERATION; 7470 7471 // handle commands that are not forwarded transparently to effect engine 7472 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7473 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7474 // no risk to block the whole media server process or mixer threads is we are stuck here 7475 Mutex::Autolock _l(mCblk->lock); 7476 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7477 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7478 mCblk->serverIndex = 0; 7479 mCblk->clientIndex = 0; 7480 return BAD_VALUE; 7481 } 7482 status_t status = NO_ERROR; 7483 while (mCblk->serverIndex < mCblk->clientIndex) { 7484 int reply; 7485 uint32_t rsize = sizeof(int); 7486 int *p = (int *)(mBuffer + mCblk->serverIndex); 7487 int size = *p++; 7488 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7489 ALOGW("command(): invalid parameter block size"); 7490 break; 7491 } 7492 effect_param_t *param = (effect_param_t *)p; 7493 if (param->psize == 0 || param->vsize == 0) { 7494 ALOGW("command(): null parameter or value size"); 7495 mCblk->serverIndex += size; 7496 continue; 7497 } 7498 uint32_t psize = sizeof(effect_param_t) + 7499 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7500 param->vsize; 7501 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7502 psize, 7503 p, 7504 &rsize, 7505 &reply); 7506 // stop at first error encountered 7507 if (ret != NO_ERROR) { 7508 status = ret; 7509 *(int *)pReplyData = reply; 7510 break; 7511 } else if (reply != NO_ERROR) { 7512 *(int *)pReplyData = reply; 7513 break; 7514 } 7515 mCblk->serverIndex += size; 7516 } 7517 mCblk->serverIndex = 0; 7518 mCblk->clientIndex = 0; 7519 return status; 7520 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7521 *(int *)pReplyData = NO_ERROR; 7522 return enable(); 7523 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7524 *(int *)pReplyData = NO_ERROR; 7525 return disable(); 7526 } 7527 7528 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7529} 7530 7531void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7532{ 7533 ALOGV("setControl %p control %d", this, hasControl); 7534 7535 mHasControl = hasControl; 7536 mEnabled = enabled; 7537 7538 if (signal && mEffectClient != 0) { 7539 mEffectClient->controlStatusChanged(hasControl); 7540 } 7541} 7542 7543void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7544 uint32_t cmdSize, 7545 void *pCmdData, 7546 uint32_t replySize, 7547 void *pReplyData) 7548{ 7549 if (mEffectClient != 0) { 7550 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7551 } 7552} 7553 7554 7555 7556void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7557{ 7558 if (mEffectClient != 0) { 7559 mEffectClient->enableStatusChanged(enabled); 7560 } 7561} 7562 7563status_t AudioFlinger::EffectHandle::onTransact( 7564 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7565{ 7566 return BnEffect::onTransact(code, data, reply, flags); 7567} 7568 7569 7570void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7571{ 7572 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7573 7574 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7575 (mClient == 0) ? getpid_cached : mClient->pid(), 7576 mPriority, 7577 mHasControl, 7578 !locked, 7579 mCblk ? mCblk->clientIndex : 0, 7580 mCblk ? mCblk->serverIndex : 0 7581 ); 7582 7583 if (locked) { 7584 mCblk->lock.unlock(); 7585 } 7586} 7587 7588#undef LOG_TAG 7589#define LOG_TAG "AudioFlinger::EffectChain" 7590 7591AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7592 int sessionId) 7593 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7594 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7595 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7596{ 7597 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7598 if (thread == NULL) { 7599 return; 7600 } 7601 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7602 thread->frameCount(); 7603} 7604 7605AudioFlinger::EffectChain::~EffectChain() 7606{ 7607 if (mOwnInBuffer) { 7608 delete mInBuffer; 7609 } 7610 7611} 7612 7613// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7614sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7615{ 7616 size_t size = mEffects.size(); 7617 7618 for (size_t i = 0; i < size; i++) { 7619 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7620 return mEffects[i]; 7621 } 7622 } 7623 return 0; 7624} 7625 7626// getEffectFromId_l() must be called with ThreadBase::mLock held 7627sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7628{ 7629 size_t size = mEffects.size(); 7630 7631 for (size_t i = 0; i < size; i++) { 7632 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7633 if (id == 0 || mEffects[i]->id() == id) { 7634 return mEffects[i]; 7635 } 7636 } 7637 return 0; 7638} 7639 7640// getEffectFromType_l() must be called with ThreadBase::mLock held 7641sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7642 const effect_uuid_t *type) 7643{ 7644 size_t size = mEffects.size(); 7645 7646 for (size_t i = 0; i < size; i++) { 7647 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7648 return mEffects[i]; 7649 } 7650 } 7651 return 0; 7652} 7653 7654// Must be called with EffectChain::mLock locked 7655void AudioFlinger::EffectChain::process_l() 7656{ 7657 sp<ThreadBase> thread = mThread.promote(); 7658 if (thread == 0) { 7659 ALOGW("process_l(): cannot promote mixer thread"); 7660 return; 7661 } 7662 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7663 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7664 // always process effects unless no more tracks are on the session and the effect tail 7665 // has been rendered 7666 bool doProcess = true; 7667 if (!isGlobalSession) { 7668 bool tracksOnSession = (trackCnt() != 0); 7669 7670 if (!tracksOnSession && mTailBufferCount == 0) { 7671 doProcess = false; 7672 } 7673 7674 if (activeTrackCnt() == 0) { 7675 // if no track is active and the effect tail has not been rendered, 7676 // the input buffer must be cleared here as the mixer process will not do it 7677 if (tracksOnSession || mTailBufferCount > 0) { 7678 size_t numSamples = thread->frameCount() * thread->channelCount(); 7679 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7680 if (mTailBufferCount > 0) { 7681 mTailBufferCount--; 7682 } 7683 } 7684 } 7685 } 7686 7687 size_t size = mEffects.size(); 7688 if (doProcess) { 7689 for (size_t i = 0; i < size; i++) { 7690 mEffects[i]->process(); 7691 } 7692 } 7693 for (size_t i = 0; i < size; i++) { 7694 mEffects[i]->updateState(); 7695 } 7696} 7697 7698// addEffect_l() must be called with PlaybackThread::mLock held 7699status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7700{ 7701 effect_descriptor_t desc = effect->desc(); 7702 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7703 7704 Mutex::Autolock _l(mLock); 7705 effect->setChain(this); 7706 sp<ThreadBase> thread = mThread.promote(); 7707 if (thread == 0) { 7708 return NO_INIT; 7709 } 7710 effect->setThread(thread); 7711 7712 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7713 // Auxiliary effects are inserted at the beginning of mEffects vector as 7714 // they are processed first and accumulated in chain input buffer 7715 mEffects.insertAt(effect, 0); 7716 7717 // the input buffer for auxiliary effect contains mono samples in 7718 // 32 bit format. This is to avoid saturation in AudoMixer 7719 // accumulation stage. Saturation is done in EffectModule::process() before 7720 // calling the process in effect engine 7721 size_t numSamples = thread->frameCount(); 7722 int32_t *buffer = new int32_t[numSamples]; 7723 memset(buffer, 0, numSamples * sizeof(int32_t)); 7724 effect->setInBuffer((int16_t *)buffer); 7725 // auxiliary effects output samples to chain input buffer for further processing 7726 // by insert effects 7727 effect->setOutBuffer(mInBuffer); 7728 } else { 7729 // Insert effects are inserted at the end of mEffects vector as they are processed 7730 // after track and auxiliary effects. 7731 // Insert effect order as a function of indicated preference: 7732 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7733 // another effect is present 7734 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7735 // last effect claiming first position 7736 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7737 // first effect claiming last position 7738 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7739 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7740 // already present 7741 7742 size_t size = mEffects.size(); 7743 size_t idx_insert = size; 7744 ssize_t idx_insert_first = -1; 7745 ssize_t idx_insert_last = -1; 7746 7747 for (size_t i = 0; i < size; i++) { 7748 effect_descriptor_t d = mEffects[i]->desc(); 7749 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7750 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7751 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7752 // check invalid effect chaining combinations 7753 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7754 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7755 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7756 return INVALID_OPERATION; 7757 } 7758 // remember position of first insert effect and by default 7759 // select this as insert position for new effect 7760 if (idx_insert == size) { 7761 idx_insert = i; 7762 } 7763 // remember position of last insert effect claiming 7764 // first position 7765 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7766 idx_insert_first = i; 7767 } 7768 // remember position of first insert effect claiming 7769 // last position 7770 if (iPref == EFFECT_FLAG_INSERT_LAST && 7771 idx_insert_last == -1) { 7772 idx_insert_last = i; 7773 } 7774 } 7775 } 7776 7777 // modify idx_insert from first position if needed 7778 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7779 if (idx_insert_last != -1) { 7780 idx_insert = idx_insert_last; 7781 } else { 7782 idx_insert = size; 7783 } 7784 } else { 7785 if (idx_insert_first != -1) { 7786 idx_insert = idx_insert_first + 1; 7787 } 7788 } 7789 7790 // always read samples from chain input buffer 7791 effect->setInBuffer(mInBuffer); 7792 7793 // if last effect in the chain, output samples to chain 7794 // output buffer, otherwise to chain input buffer 7795 if (idx_insert == size) { 7796 if (idx_insert != 0) { 7797 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7798 mEffects[idx_insert-1]->configure(); 7799 } 7800 effect->setOutBuffer(mOutBuffer); 7801 } else { 7802 effect->setOutBuffer(mInBuffer); 7803 } 7804 mEffects.insertAt(effect, idx_insert); 7805 7806 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7807 } 7808 effect->configure(); 7809 return NO_ERROR; 7810} 7811 7812// removeEffect_l() must be called with PlaybackThread::mLock held 7813size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7814{ 7815 Mutex::Autolock _l(mLock); 7816 size_t size = mEffects.size(); 7817 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7818 7819 for (size_t i = 0; i < size; i++) { 7820 if (effect == mEffects[i]) { 7821 // calling stop here will remove pre-processing effect from the audio HAL. 7822 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7823 // the middle of a read from audio HAL 7824 if (mEffects[i]->state() == EffectModule::ACTIVE || 7825 mEffects[i]->state() == EffectModule::STOPPING) { 7826 mEffects[i]->stop(); 7827 } 7828 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7829 delete[] effect->inBuffer(); 7830 } else { 7831 if (i == size - 1 && i != 0) { 7832 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7833 mEffects[i - 1]->configure(); 7834 } 7835 } 7836 mEffects.removeAt(i); 7837 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7838 break; 7839 } 7840 } 7841 7842 return mEffects.size(); 7843} 7844 7845// setDevice_l() must be called with PlaybackThread::mLock held 7846void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7847{ 7848 size_t size = mEffects.size(); 7849 for (size_t i = 0; i < size; i++) { 7850 mEffects[i]->setDevice(device); 7851 } 7852} 7853 7854// setMode_l() must be called with PlaybackThread::mLock held 7855void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7856{ 7857 size_t size = mEffects.size(); 7858 for (size_t i = 0; i < size; i++) { 7859 mEffects[i]->setMode(mode); 7860 } 7861} 7862 7863// setVolume_l() must be called with PlaybackThread::mLock held 7864bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7865{ 7866 uint32_t newLeft = *left; 7867 uint32_t newRight = *right; 7868 bool hasControl = false; 7869 int ctrlIdx = -1; 7870 size_t size = mEffects.size(); 7871 7872 // first update volume controller 7873 for (size_t i = size; i > 0; i--) { 7874 if (mEffects[i - 1]->isProcessEnabled() && 7875 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7876 ctrlIdx = i - 1; 7877 hasControl = true; 7878 break; 7879 } 7880 } 7881 7882 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7883 if (hasControl) { 7884 *left = mNewLeftVolume; 7885 *right = mNewRightVolume; 7886 } 7887 return hasControl; 7888 } 7889 7890 mVolumeCtrlIdx = ctrlIdx; 7891 mLeftVolume = newLeft; 7892 mRightVolume = newRight; 7893 7894 // second get volume update from volume controller 7895 if (ctrlIdx >= 0) { 7896 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7897 mNewLeftVolume = newLeft; 7898 mNewRightVolume = newRight; 7899 } 7900 // then indicate volume to all other effects in chain. 7901 // Pass altered volume to effects before volume controller 7902 // and requested volume to effects after controller 7903 uint32_t lVol = newLeft; 7904 uint32_t rVol = newRight; 7905 7906 for (size_t i = 0; i < size; i++) { 7907 if ((int)i == ctrlIdx) continue; 7908 // this also works for ctrlIdx == -1 when there is no volume controller 7909 if ((int)i > ctrlIdx) { 7910 lVol = *left; 7911 rVol = *right; 7912 } 7913 mEffects[i]->setVolume(&lVol, &rVol, false); 7914 } 7915 *left = newLeft; 7916 *right = newRight; 7917 7918 return hasControl; 7919} 7920 7921status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7922{ 7923 const size_t SIZE = 256; 7924 char buffer[SIZE]; 7925 String8 result; 7926 7927 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7928 result.append(buffer); 7929 7930 bool locked = tryLock(mLock); 7931 // failed to lock - AudioFlinger is probably deadlocked 7932 if (!locked) { 7933 result.append("\tCould not lock mutex:\n"); 7934 } 7935 7936 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7937 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7938 mEffects.size(), 7939 (uint32_t)mInBuffer, 7940 (uint32_t)mOutBuffer, 7941 mActiveTrackCnt); 7942 result.append(buffer); 7943 write(fd, result.string(), result.size()); 7944 7945 for (size_t i = 0; i < mEffects.size(); ++i) { 7946 sp<EffectModule> effect = mEffects[i]; 7947 if (effect != 0) { 7948 effect->dump(fd, args); 7949 } 7950 } 7951 7952 if (locked) { 7953 mLock.unlock(); 7954 } 7955 7956 return NO_ERROR; 7957} 7958 7959// must be called with ThreadBase::mLock held 7960void AudioFlinger::EffectChain::setEffectSuspended_l( 7961 const effect_uuid_t *type, bool suspend) 7962{ 7963 sp<SuspendedEffectDesc> desc; 7964 // use effect type UUID timelow as key as there is no real risk of identical 7965 // timeLow fields among effect type UUIDs. 7966 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7967 if (suspend) { 7968 if (index >= 0) { 7969 desc = mSuspendedEffects.valueAt(index); 7970 } else { 7971 desc = new SuspendedEffectDesc(); 7972 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7973 mSuspendedEffects.add(type->timeLow, desc); 7974 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7975 } 7976 if (desc->mRefCount++ == 0) { 7977 sp<EffectModule> effect = getEffectIfEnabled(type); 7978 if (effect != 0) { 7979 desc->mEffect = effect; 7980 effect->setSuspended(true); 7981 effect->setEnabled(false); 7982 } 7983 } 7984 } else { 7985 if (index < 0) { 7986 return; 7987 } 7988 desc = mSuspendedEffects.valueAt(index); 7989 if (desc->mRefCount <= 0) { 7990 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7991 desc->mRefCount = 1; 7992 } 7993 if (--desc->mRefCount == 0) { 7994 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7995 if (desc->mEffect != 0) { 7996 sp<EffectModule> effect = desc->mEffect.promote(); 7997 if (effect != 0) { 7998 effect->setSuspended(false); 7999 sp<EffectHandle> handle = effect->controlHandle(); 8000 if (handle != 0) { 8001 effect->setEnabled(handle->enabled()); 8002 } 8003 } 8004 desc->mEffect.clear(); 8005 } 8006 mSuspendedEffects.removeItemsAt(index); 8007 } 8008 } 8009} 8010 8011// must be called with ThreadBase::mLock held 8012void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 8013{ 8014 sp<SuspendedEffectDesc> desc; 8015 8016 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8017 if (suspend) { 8018 if (index >= 0) { 8019 desc = mSuspendedEffects.valueAt(index); 8020 } else { 8021 desc = new SuspendedEffectDesc(); 8022 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8023 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8024 } 8025 if (desc->mRefCount++ == 0) { 8026 Vector< sp<EffectModule> > effects; 8027 getSuspendEligibleEffects(effects); 8028 for (size_t i = 0; i < effects.size(); i++) { 8029 setEffectSuspended_l(&effects[i]->desc().type, true); 8030 } 8031 } 8032 } else { 8033 if (index < 0) { 8034 return; 8035 } 8036 desc = mSuspendedEffects.valueAt(index); 8037 if (desc->mRefCount <= 0) { 8038 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8039 desc->mRefCount = 1; 8040 } 8041 if (--desc->mRefCount == 0) { 8042 Vector<const effect_uuid_t *> types; 8043 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8044 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8045 continue; 8046 } 8047 types.add(&mSuspendedEffects.valueAt(i)->mType); 8048 } 8049 for (size_t i = 0; i < types.size(); i++) { 8050 setEffectSuspended_l(types[i], false); 8051 } 8052 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8053 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8054 } 8055 } 8056} 8057 8058 8059// The volume effect is used for automated tests only 8060#ifndef OPENSL_ES_H_ 8061static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8062 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8063const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8064#endif //OPENSL_ES_H_ 8065 8066bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8067{ 8068 // auxiliary effects and visualizer are never suspended on output mix 8069 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8070 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8071 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8072 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8073 return false; 8074 } 8075 return true; 8076} 8077 8078void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8079{ 8080 effects.clear(); 8081 for (size_t i = 0; i < mEffects.size(); i++) { 8082 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8083 effects.add(mEffects[i]); 8084 } 8085 } 8086} 8087 8088sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8089 const effect_uuid_t *type) 8090{ 8091 sp<EffectModule> effect = getEffectFromType_l(type); 8092 return effect != 0 && effect->isEnabled() ? effect : 0; 8093} 8094 8095void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8096 bool enabled) 8097{ 8098 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8099 if (enabled) { 8100 if (index < 0) { 8101 // if the effect is not suspend check if all effects are suspended 8102 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8103 if (index < 0) { 8104 return; 8105 } 8106 if (!isEffectEligibleForSuspend(effect->desc())) { 8107 return; 8108 } 8109 setEffectSuspended_l(&effect->desc().type, enabled); 8110 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8111 if (index < 0) { 8112 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8113 return; 8114 } 8115 } 8116 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8117 effect->desc().type.timeLow); 8118 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8119 // if effect is requested to suspended but was not yet enabled, supend it now. 8120 if (desc->mEffect == 0) { 8121 desc->mEffect = effect; 8122 effect->setEnabled(false); 8123 effect->setSuspended(true); 8124 } 8125 } else { 8126 if (index < 0) { 8127 return; 8128 } 8129 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8130 effect->desc().type.timeLow); 8131 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8132 desc->mEffect.clear(); 8133 effect->setSuspended(false); 8134 } 8135} 8136 8137#undef LOG_TAG 8138#define LOG_TAG "AudioFlinger" 8139 8140// ---------------------------------------------------------------------------- 8141 8142status_t AudioFlinger::onTransact( 8143 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8144{ 8145 return BnAudioFlinger::onTransact(code, data, reply, flags); 8146} 8147 8148}; // namespace android 8149