AudioFlinger.cpp revision c0b52836d07f823732f0ff98ca5ca9d7f5730cb8
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_INIT; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_INIT; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid cnt\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->sessionid, r->pid, r->cnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 assert(NULL != mPrimaryHardwareDev); 727 assert(NULL != mPrimaryHardwareDev->get_master_volume); 728 729 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 730 mHardwareStatus = AUDIO_HW_IDLE; 731 return ret_val; 732 } 733 734 return mMasterVolume; 735} 736 737status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 738 audio_io_handle_t output) 739{ 740 // check calling permissions 741 if (!settingsAllowed()) { 742 return PERMISSION_DENIED; 743 } 744 745 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 746 ALOGE("setStreamVolume() invalid stream %d", stream); 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 PlaybackThread *thread = NULL; 752 if (output) { 753 thread = checkPlaybackThread_l(output); 754 if (thread == NULL) { 755 return BAD_VALUE; 756 } 757 } 758 759 mStreamTypes[stream].volume = value; 760 761 if (thread == NULL) { 762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 763 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 764 } 765 } else { 766 thread->setStreamVolume(stream, value); 767 } 768 769 return NO_ERROR; 770} 771 772status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 773{ 774 // check calling permissions 775 if (!settingsAllowed()) { 776 return PERMISSION_DENIED; 777 } 778 779 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 780 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 781 ALOGE("setStreamMute() invalid stream %d", stream); 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 mStreamTypes[stream].mute = muted; 787 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 789 790 return NO_ERROR; 791} 792 793float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 794{ 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 return 0.0f; 797 } 798 799 AutoMutex lock(mLock); 800 float volume; 801 if (output) { 802 PlaybackThread *thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return 0.0f; 805 } 806 volume = thread->streamVolume(stream); 807 } else { 808 volume = streamVolume_l(stream); 809 } 810 811 return volume; 812} 813 814bool AudioFlinger::streamMute(audio_stream_type_t stream) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return true; 818 } 819 820 AutoMutex lock(mLock); 821 return streamMute_l(stream); 822} 823 824status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 825{ 826 status_t result; 827 828 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 829 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 830 // check calling permissions 831 if (!settingsAllowed()) { 832 return PERMISSION_DENIED; 833 } 834 835 // ioHandle == 0 means the parameters are global to the audio hardware interface 836 if (ioHandle == 0) { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_SET_PARAMETER; 839 status_t final_result = NO_ERROR; 840 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 841 audio_hw_device_t *dev = mAudioHwDevs[i]; 842 result = dev->set_parameters(dev, keyValuePairs.string()); 843 final_result = result ?: final_result; 844 } 845 mHardwareStatus = AUDIO_HW_IDLE; 846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 847 AudioParameter param = AudioParameter(keyValuePairs); 848 String8 value; 849 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 850 Mutex::Autolock _l(mLock); 851 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 852 if (mBtNrecIsOff != btNrecIsOff) { 853 for (size_t i = 0; i < mRecordThreads.size(); i++) { 854 sp<RecordThread> thread = mRecordThreads.valueAt(i); 855 RecordThread::RecordTrack *track = thread->track(); 856 if (track != NULL) { 857 audio_devices_t device = (audio_devices_t)( 858 thread->device() & AUDIO_DEVICE_IN_ALL); 859 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 860 thread->setEffectSuspended(FX_IID_AEC, 861 suspend, 862 track->sessionId()); 863 thread->setEffectSuspended(FX_IID_NS, 864 suspend, 865 track->sessionId()); 866 } 867 } 868 mBtNrecIsOff = btNrecIsOff; 869 } 870 } 871 return final_result; 872 } 873 874 // hold a strong ref on thread in case closeOutput() or closeInput() is called 875 // and the thread is exited once the lock is released 876 sp<ThreadBase> thread; 877 { 878 Mutex::Autolock _l(mLock); 879 thread = checkPlaybackThread_l(ioHandle); 880 if (thread == NULL) { 881 thread = checkRecordThread_l(ioHandle); 882 } else if (thread == primaryPlaybackThread_l()) { 883 // indicate output device change to all input threads for pre processing 884 AudioParameter param = AudioParameter(keyValuePairs); 885 int value; 886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 887 for (size_t i = 0; i < mRecordThreads.size(); i++) { 888 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 889 } 890 } 891 } 892 } 893 if (thread != 0) { 894 return thread->setParameters(keyValuePairs); 895 } 896 return BAD_VALUE; 897} 898 899String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 900{ 901// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 902// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 903 904 if (ioHandle == 0) { 905 String8 out_s8; 906 907 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 908 audio_hw_device_t *dev = mAudioHwDevs[i]; 909 char *s = dev->get_parameters(dev, keys.string()); 910 out_s8 += String8(s ? s : ""); 911 free(s); 912 } 913 return out_s8; 914 } 915 916 Mutex::Autolock _l(mLock); 917 918 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 919 if (playbackThread != NULL) { 920 return playbackThread->getParameters(keys); 921 } 922 RecordThread *recordThread = checkRecordThread_l(ioHandle); 923 if (recordThread != NULL) { 924 return recordThread->getParameters(keys); 925 } 926 return String8(""); 927} 928 929size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 930{ 931 status_t ret = initCheck(); 932 if (ret != NO_ERROR) { 933 return 0; 934 } 935 936 AutoMutex lock(mHardwareLock); 937 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 938 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 939 mHardwareStatus = AUDIO_HW_IDLE; 940 return size; 941} 942 943unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 944{ 945 if (ioHandle == 0) { 946 return 0; 947 } 948 949 Mutex::Autolock _l(mLock); 950 951 RecordThread *recordThread = checkRecordThread_l(ioHandle); 952 if (recordThread != NULL) { 953 return recordThread->getInputFramesLost(); 954 } 955 return 0; 956} 957 958status_t AudioFlinger::setVoiceVolume(float value) 959{ 960 status_t ret = initCheck(); 961 if (ret != NO_ERROR) { 962 return ret; 963 } 964 965 // check calling permissions 966 if (!settingsAllowed()) { 967 return PERMISSION_DENIED; 968 } 969 970 AutoMutex lock(mHardwareLock); 971 mHardwareStatus = AUDIO_SET_VOICE_VOLUME; 972 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 973 mHardwareStatus = AUDIO_HW_IDLE; 974 975 return ret; 976} 977 978status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 979 audio_io_handle_t output) const 980{ 981 status_t status; 982 983 Mutex::Autolock _l(mLock); 984 985 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 986 if (playbackThread != NULL) { 987 return playbackThread->getRenderPosition(halFrames, dspFrames); 988 } 989 990 return BAD_VALUE; 991} 992 993void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 994{ 995 996 Mutex::Autolock _l(mLock); 997 998 pid_t pid = IPCThreadState::self()->getCallingPid(); 999 if (mNotificationClients.indexOfKey(pid) < 0) { 1000 sp<NotificationClient> notificationClient = new NotificationClient(this, 1001 client, 1002 pid); 1003 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1004 1005 mNotificationClients.add(pid, notificationClient); 1006 1007 sp<IBinder> binder = client->asBinder(); 1008 binder->linkToDeath(notificationClient); 1009 1010 // the config change is always sent from playback or record threads to avoid deadlock 1011 // with AudioSystem::gLock 1012 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1013 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1014 } 1015 1016 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1017 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1018 } 1019 } 1020} 1021 1022void AudioFlinger::removeNotificationClient(pid_t pid) 1023{ 1024 Mutex::Autolock _l(mLock); 1025 1026 ssize_t index = mNotificationClients.indexOfKey(pid); 1027 if (index >= 0) { 1028 sp <NotificationClient> client = mNotificationClients.valueFor(pid); 1029 ALOGV("removeNotificationClient() %p, pid %d", client.get(), pid); 1030 mNotificationClients.removeItem(pid); 1031 } 1032 1033 ALOGV("%d died, releasing its sessions", pid); 1034 size_t num = mAudioSessionRefs.size(); 1035 bool removed = false; 1036 for (size_t i = 0; i< num; ) { 1037 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1038 ALOGV(" pid %d @ %d", ref->pid, i); 1039 if (ref->pid == pid) { 1040 ALOGV(" removing entry for pid %d session %d", pid, ref->sessionid); 1041 mAudioSessionRefs.removeAt(i); 1042 delete ref; 1043 removed = true; 1044 num--; 1045 } else { 1046 i++; 1047 } 1048 } 1049 if (removed) { 1050 purgeStaleEffects_l(); 1051 } 1052} 1053 1054// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1055void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1056{ 1057 size_t size = mNotificationClients.size(); 1058 for (size_t i = 0; i < size; i++) { 1059 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1060 param2); 1061 } 1062} 1063 1064// removeClient_l() must be called with AudioFlinger::mLock held 1065void AudioFlinger::removeClient_l(pid_t pid) 1066{ 1067 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1068 mClients.removeItem(pid); 1069} 1070 1071 1072// ---------------------------------------------------------------------------- 1073 1074AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1075 uint32_t device, type_t type) 1076 : Thread(false), 1077 mType(type), 1078 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1079 // mChannelMask 1080 mChannelCount(0), 1081 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1082 mParamStatus(NO_ERROR), 1083 mStandby(false), mId(id), 1084 mDevice(device), 1085 mDeathRecipient(new PMDeathRecipient(this)) 1086{ 1087} 1088 1089AudioFlinger::ThreadBase::~ThreadBase() 1090{ 1091 mParamCond.broadcast(); 1092 // do not lock the mutex in destructor 1093 releaseWakeLock_l(); 1094 if (mPowerManager != 0) { 1095 sp<IBinder> binder = mPowerManager->asBinder(); 1096 binder->unlinkToDeath(mDeathRecipient); 1097 } 1098} 1099 1100void AudioFlinger::ThreadBase::exit() 1101{ 1102 ALOGV("ThreadBase::exit"); 1103 { 1104 // This lock prevents the following race in thread (uniprocessor for illustration): 1105 // if (!exitPending()) { 1106 // // context switch from here to exit() 1107 // // exit() calls requestExit(), what exitPending() observes 1108 // // exit() calls signal(), which is dropped since no waiters 1109 // // context switch back from exit() to here 1110 // mWaitWorkCV.wait(...); 1111 // // now thread is hung 1112 // } 1113 AutoMutex lock(mLock); 1114 requestExit(); 1115 mWaitWorkCV.signal(); 1116 } 1117 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1118 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1119 requestExitAndWait(); 1120} 1121 1122status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1123{ 1124 status_t status; 1125 1126 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1127 Mutex::Autolock _l(mLock); 1128 1129 mNewParameters.add(keyValuePairs); 1130 mWaitWorkCV.signal(); 1131 // wait condition with timeout in case the thread loop has exited 1132 // before the request could be processed 1133 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1134 status = mParamStatus; 1135 mWaitWorkCV.signal(); 1136 } else { 1137 status = TIMED_OUT; 1138 } 1139 return status; 1140} 1141 1142void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1143{ 1144 Mutex::Autolock _l(mLock); 1145 sendConfigEvent_l(event, param); 1146} 1147 1148// sendConfigEvent_l() must be called with ThreadBase::mLock held 1149void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1150{ 1151 ConfigEvent configEvent; 1152 configEvent.mEvent = event; 1153 configEvent.mParam = param; 1154 mConfigEvents.add(configEvent); 1155 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1156 mWaitWorkCV.signal(); 1157} 1158 1159void AudioFlinger::ThreadBase::processConfigEvents() 1160{ 1161 mLock.lock(); 1162 while(!mConfigEvents.isEmpty()) { 1163 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1164 ConfigEvent configEvent = mConfigEvents[0]; 1165 mConfigEvents.removeAt(0); 1166 // release mLock before locking AudioFlinger mLock: lock order is always 1167 // AudioFlinger then ThreadBase to avoid cross deadlock 1168 mLock.unlock(); 1169 mAudioFlinger->mLock.lock(); 1170 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1171 mAudioFlinger->mLock.unlock(); 1172 mLock.lock(); 1173 } 1174 mLock.unlock(); 1175} 1176 1177status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1178{ 1179 const size_t SIZE = 256; 1180 char buffer[SIZE]; 1181 String8 result; 1182 1183 bool locked = tryLock(mLock); 1184 if (!locked) { 1185 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1186 write(fd, buffer, strlen(buffer)); 1187 } 1188 1189 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1190 result.append(buffer); 1191 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1192 result.append(buffer); 1193 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1194 result.append(buffer); 1195 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1196 result.append(buffer); 1197 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1202 result.append(buffer); 1203 1204 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1205 result.append(buffer); 1206 result.append(" Index Command"); 1207 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1208 snprintf(buffer, SIZE, "\n %02d ", i); 1209 result.append(buffer); 1210 result.append(mNewParameters[i]); 1211 } 1212 1213 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1214 result.append(buffer); 1215 snprintf(buffer, SIZE, " Index event param\n"); 1216 result.append(buffer); 1217 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1218 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1219 result.append(buffer); 1220 } 1221 result.append("\n"); 1222 1223 write(fd, result.string(), result.size()); 1224 1225 if (locked) { 1226 mLock.unlock(); 1227 } 1228 return NO_ERROR; 1229} 1230 1231status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1232{ 1233 const size_t SIZE = 256; 1234 char buffer[SIZE]; 1235 String8 result; 1236 1237 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1238 write(fd, buffer, strlen(buffer)); 1239 1240 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1241 sp<EffectChain> chain = mEffectChains[i]; 1242 if (chain != 0) { 1243 chain->dump(fd, args); 1244 } 1245 } 1246 return NO_ERROR; 1247} 1248 1249void AudioFlinger::ThreadBase::acquireWakeLock() 1250{ 1251 Mutex::Autolock _l(mLock); 1252 acquireWakeLock_l(); 1253} 1254 1255void AudioFlinger::ThreadBase::acquireWakeLock_l() 1256{ 1257 if (mPowerManager == 0) { 1258 // use checkService() to avoid blocking if power service is not up yet 1259 sp<IBinder> binder = 1260 defaultServiceManager()->checkService(String16("power")); 1261 if (binder == 0) { 1262 ALOGW("Thread %s cannot connect to the power manager service", mName); 1263 } else { 1264 mPowerManager = interface_cast<IPowerManager>(binder); 1265 binder->linkToDeath(mDeathRecipient); 1266 } 1267 } 1268 if (mPowerManager != 0) { 1269 sp<IBinder> binder = new BBinder(); 1270 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1271 binder, 1272 String16(mName)); 1273 if (status == NO_ERROR) { 1274 mWakeLockToken = binder; 1275 } 1276 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1277 } 1278} 1279 1280void AudioFlinger::ThreadBase::releaseWakeLock() 1281{ 1282 Mutex::Autolock _l(mLock); 1283 releaseWakeLock_l(); 1284} 1285 1286void AudioFlinger::ThreadBase::releaseWakeLock_l() 1287{ 1288 if (mWakeLockToken != 0) { 1289 ALOGV("releaseWakeLock_l() %s", mName); 1290 if (mPowerManager != 0) { 1291 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1292 } 1293 mWakeLockToken.clear(); 1294 } 1295} 1296 1297void AudioFlinger::ThreadBase::clearPowerManager() 1298{ 1299 Mutex::Autolock _l(mLock); 1300 releaseWakeLock_l(); 1301 mPowerManager.clear(); 1302} 1303 1304void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1305{ 1306 sp<ThreadBase> thread = mThread.promote(); 1307 if (thread != 0) { 1308 thread->clearPowerManager(); 1309 } 1310 ALOGW("power manager service died !!!"); 1311} 1312 1313void AudioFlinger::ThreadBase::setEffectSuspended( 1314 const effect_uuid_t *type, bool suspend, int sessionId) 1315{ 1316 Mutex::Autolock _l(mLock); 1317 setEffectSuspended_l(type, suspend, sessionId); 1318} 1319 1320void AudioFlinger::ThreadBase::setEffectSuspended_l( 1321 const effect_uuid_t *type, bool suspend, int sessionId) 1322{ 1323 sp<EffectChain> chain = getEffectChain_l(sessionId); 1324 if (chain != 0) { 1325 if (type != NULL) { 1326 chain->setEffectSuspended_l(type, suspend); 1327 } else { 1328 chain->setEffectSuspendedAll_l(suspend); 1329 } 1330 } 1331 1332 updateSuspendedSessions_l(type, suspend, sessionId); 1333} 1334 1335void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1336{ 1337 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1338 if (index < 0) { 1339 return; 1340 } 1341 1342 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1343 mSuspendedSessions.editValueAt(index); 1344 1345 for (size_t i = 0; i < sessionEffects.size(); i++) { 1346 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1347 for (int j = 0; j < desc->mRefCount; j++) { 1348 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1349 chain->setEffectSuspendedAll_l(true); 1350 } else { 1351 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1352 desc->mType.timeLow); 1353 chain->setEffectSuspended_l(&desc->mType, true); 1354 } 1355 } 1356 } 1357} 1358 1359void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1360 bool suspend, 1361 int sessionId) 1362{ 1363 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1364 1365 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1366 1367 if (suspend) { 1368 if (index >= 0) { 1369 sessionEffects = mSuspendedSessions.editValueAt(index); 1370 } else { 1371 mSuspendedSessions.add(sessionId, sessionEffects); 1372 } 1373 } else { 1374 if (index < 0) { 1375 return; 1376 } 1377 sessionEffects = mSuspendedSessions.editValueAt(index); 1378 } 1379 1380 1381 int key = EffectChain::kKeyForSuspendAll; 1382 if (type != NULL) { 1383 key = type->timeLow; 1384 } 1385 index = sessionEffects.indexOfKey(key); 1386 1387 sp <SuspendedSessionDesc> desc; 1388 if (suspend) { 1389 if (index >= 0) { 1390 desc = sessionEffects.valueAt(index); 1391 } else { 1392 desc = new SuspendedSessionDesc(); 1393 if (type != NULL) { 1394 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1395 } 1396 sessionEffects.add(key, desc); 1397 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1398 } 1399 desc->mRefCount++; 1400 } else { 1401 if (index < 0) { 1402 return; 1403 } 1404 desc = sessionEffects.valueAt(index); 1405 if (--desc->mRefCount == 0) { 1406 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1407 sessionEffects.removeItemsAt(index); 1408 if (sessionEffects.isEmpty()) { 1409 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1410 sessionId); 1411 mSuspendedSessions.removeItem(sessionId); 1412 } 1413 } 1414 } 1415 if (!sessionEffects.isEmpty()) { 1416 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1417 } 1418} 1419 1420void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1421 bool enabled, 1422 int sessionId) 1423{ 1424 Mutex::Autolock _l(mLock); 1425 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1426} 1427 1428void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1429 bool enabled, 1430 int sessionId) 1431{ 1432 if (mType != RECORD) { 1433 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1434 // another session. This gives the priority to well behaved effect control panels 1435 // and applications not using global effects. 1436 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1437 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1438 } 1439 } 1440 1441 sp<EffectChain> chain = getEffectChain_l(sessionId); 1442 if (chain != 0) { 1443 chain->checkSuspendOnEffectEnabled(effect, enabled); 1444 } 1445} 1446 1447// ---------------------------------------------------------------------------- 1448 1449AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1450 AudioStreamOut* output, 1451 audio_io_handle_t id, 1452 uint32_t device, 1453 type_t type) 1454 : ThreadBase(audioFlinger, id, device, type), 1455 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1456 // Assumes constructor is called by AudioFlinger with it's mLock held, 1457 // but it would be safer to explicitly pass initial masterMute as parameter 1458 mMasterMute(audioFlinger->masterMute_l()), 1459 // mStreamTypes[] initialized in constructor body 1460 mOutput(output), 1461 // Assumes constructor is called by AudioFlinger with it's mLock held, 1462 // but it would be safer to explicitly pass initial masterVolume as parameter 1463 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1464 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1465{ 1466 snprintf(mName, kNameLength, "AudioOut_%d", id); 1467 1468 readOutputParameters(); 1469 1470 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1471 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1472 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1473 stream = (audio_stream_type_t) (stream + 1)) { 1474 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1475 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1476 // initialized by stream_type_t default constructor 1477 // mStreamTypes[stream].valid = true; 1478 } 1479 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1480 // because mAudioFlinger doesn't have one to copy from 1481} 1482 1483AudioFlinger::PlaybackThread::~PlaybackThread() 1484{ 1485 delete [] mMixBuffer; 1486} 1487 1488status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1489{ 1490 dumpInternals(fd, args); 1491 dumpTracks(fd, args); 1492 dumpEffectChains(fd, args); 1493 return NO_ERROR; 1494} 1495 1496status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1497{ 1498 const size_t SIZE = 256; 1499 char buffer[SIZE]; 1500 String8 result; 1501 1502 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1503 result.append(buffer); 1504 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1505 for (size_t i = 0; i < mTracks.size(); ++i) { 1506 sp<Track> track = mTracks[i]; 1507 if (track != 0) { 1508 track->dump(buffer, SIZE); 1509 result.append(buffer); 1510 } 1511 } 1512 1513 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1514 result.append(buffer); 1515 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1516 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1517 sp<Track> track = mActiveTracks[i].promote(); 1518 if (track != 0) { 1519 track->dump(buffer, SIZE); 1520 result.append(buffer); 1521 } 1522 } 1523 write(fd, result.string(), result.size()); 1524 return NO_ERROR; 1525} 1526 1527status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1528{ 1529 const size_t SIZE = 256; 1530 char buffer[SIZE]; 1531 String8 result; 1532 1533 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1534 result.append(buffer); 1535 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1536 result.append(buffer); 1537 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1538 result.append(buffer); 1539 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1540 result.append(buffer); 1541 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1542 result.append(buffer); 1543 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1544 result.append(buffer); 1545 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1546 result.append(buffer); 1547 write(fd, result.string(), result.size()); 1548 1549 dumpBase(fd, args); 1550 1551 return NO_ERROR; 1552} 1553 1554// Thread virtuals 1555status_t AudioFlinger::PlaybackThread::readyToRun() 1556{ 1557 status_t status = initCheck(); 1558 if (status == NO_ERROR) { 1559 ALOGI("AudioFlinger's thread %p ready to run", this); 1560 } else { 1561 ALOGE("No working audio driver found."); 1562 } 1563 return status; 1564} 1565 1566void AudioFlinger::PlaybackThread::onFirstRef() 1567{ 1568 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1569} 1570 1571// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1572sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1573 const sp<AudioFlinger::Client>& client, 1574 audio_stream_type_t streamType, 1575 uint32_t sampleRate, 1576 audio_format_t format, 1577 uint32_t channelMask, 1578 int frameCount, 1579 const sp<IMemory>& sharedBuffer, 1580 int sessionId, 1581 bool isTimed, 1582 status_t *status) 1583{ 1584 sp<Track> track; 1585 status_t lStatus; 1586 1587 if (mType == DIRECT) { 1588 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1589 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1590 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1591 "for output %p with format %d", 1592 sampleRate, format, channelMask, mOutput, mFormat); 1593 lStatus = BAD_VALUE; 1594 goto Exit; 1595 } 1596 } 1597 } else { 1598 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1599 if (sampleRate > mSampleRate*2) { 1600 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1601 lStatus = BAD_VALUE; 1602 goto Exit; 1603 } 1604 } 1605 1606 lStatus = initCheck(); 1607 if (lStatus != NO_ERROR) { 1608 ALOGE("Audio driver not initialized."); 1609 goto Exit; 1610 } 1611 1612 { // scope for mLock 1613 Mutex::Autolock _l(mLock); 1614 1615 // all tracks in same audio session must share the same routing strategy otherwise 1616 // conflicts will happen when tracks are moved from one output to another by audio policy 1617 // manager 1618 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1619 for (size_t i = 0; i < mTracks.size(); ++i) { 1620 sp<Track> t = mTracks[i]; 1621 if (t != 0) { 1622 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1623 if (sessionId == t->sessionId() && strategy != actual) { 1624 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1625 strategy, actual); 1626 lStatus = BAD_VALUE; 1627 goto Exit; 1628 } 1629 } 1630 } 1631 1632 if (!isTimed) { 1633 track = new Track(this, client, streamType, sampleRate, format, 1634 channelMask, frameCount, sharedBuffer, sessionId); 1635 } else { 1636 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1637 channelMask, frameCount, sharedBuffer, sessionId); 1638 } 1639 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1640 lStatus = NO_MEMORY; 1641 goto Exit; 1642 } 1643 mTracks.add(track); 1644 1645 sp<EffectChain> chain = getEffectChain_l(sessionId); 1646 if (chain != 0) { 1647 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1648 track->setMainBuffer(chain->inBuffer()); 1649 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1650 chain->incTrackCnt(); 1651 } 1652 1653 // invalidate track immediately if the stream type was moved to another thread since 1654 // createTrack() was called by the client process. 1655 if (!mStreamTypes[streamType].valid) { 1656 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1657 this, streamType); 1658 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1659 } 1660 } 1661 lStatus = NO_ERROR; 1662 1663Exit: 1664 if(status) { 1665 *status = lStatus; 1666 } 1667 return track; 1668} 1669 1670uint32_t AudioFlinger::PlaybackThread::latency() const 1671{ 1672 Mutex::Autolock _l(mLock); 1673 if (initCheck() == NO_ERROR) { 1674 return mOutput->stream->get_latency(mOutput->stream); 1675 } else { 1676 return 0; 1677 } 1678} 1679 1680void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1681{ 1682 Mutex::Autolock _l(mLock); 1683 mMasterVolume = value; 1684} 1685 1686void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1687{ 1688 Mutex::Autolock _l(mLock); 1689 setMasterMute_l(muted); 1690} 1691 1692void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1693{ 1694 Mutex::Autolock _l(mLock); 1695 mStreamTypes[stream].volume = value; 1696} 1697 1698void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1699{ 1700 Mutex::Autolock _l(mLock); 1701 mStreamTypes[stream].mute = muted; 1702} 1703 1704float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1705{ 1706 Mutex::Autolock _l(mLock); 1707 return mStreamTypes[stream].volume; 1708} 1709 1710// addTrack_l() must be called with ThreadBase::mLock held 1711status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1712{ 1713 status_t status = ALREADY_EXISTS; 1714 1715 // set retry count for buffer fill 1716 track->mRetryCount = kMaxTrackStartupRetries; 1717 if (mActiveTracks.indexOf(track) < 0) { 1718 // the track is newly added, make sure it fills up all its 1719 // buffers before playing. This is to ensure the client will 1720 // effectively get the latency it requested. 1721 track->mFillingUpStatus = Track::FS_FILLING; 1722 track->mResetDone = false; 1723 mActiveTracks.add(track); 1724 if (track->mainBuffer() != mMixBuffer) { 1725 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1726 if (chain != 0) { 1727 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1728 chain->incActiveTrackCnt(); 1729 } 1730 } 1731 1732 status = NO_ERROR; 1733 } 1734 1735 ALOGV("mWaitWorkCV.broadcast"); 1736 mWaitWorkCV.broadcast(); 1737 1738 return status; 1739} 1740 1741// destroyTrack_l() must be called with ThreadBase::mLock held 1742void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1743{ 1744 track->mState = TrackBase::TERMINATED; 1745 if (mActiveTracks.indexOf(track) < 0) { 1746 removeTrack_l(track); 1747 } 1748} 1749 1750void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1751{ 1752 mTracks.remove(track); 1753 deleteTrackName_l(track->name()); 1754 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1755 if (chain != 0) { 1756 chain->decTrackCnt(); 1757 } 1758} 1759 1760String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1761{ 1762 String8 out_s8 = String8(""); 1763 char *s; 1764 1765 Mutex::Autolock _l(mLock); 1766 if (initCheck() != NO_ERROR) { 1767 return out_s8; 1768 } 1769 1770 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1771 out_s8 = String8(s); 1772 free(s); 1773 return out_s8; 1774} 1775 1776// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1777void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1778 AudioSystem::OutputDescriptor desc; 1779 void *param2 = NULL; 1780 1781 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1782 1783 switch (event) { 1784 case AudioSystem::OUTPUT_OPENED: 1785 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1786 desc.channels = mChannelMask; 1787 desc.samplingRate = mSampleRate; 1788 desc.format = mFormat; 1789 desc.frameCount = mFrameCount; 1790 desc.latency = latency(); 1791 param2 = &desc; 1792 break; 1793 1794 case AudioSystem::STREAM_CONFIG_CHANGED: 1795 param2 = ¶m; 1796 case AudioSystem::OUTPUT_CLOSED: 1797 default: 1798 break; 1799 } 1800 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1801} 1802 1803void AudioFlinger::PlaybackThread::readOutputParameters() 1804{ 1805 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1806 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1807 mChannelCount = (uint16_t)popcount(mChannelMask); 1808 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1809 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1810 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1811 1812 // FIXME - Current mixer implementation only supports stereo output: Always 1813 // Allocate a stereo buffer even if HW output is mono. 1814 delete[] mMixBuffer; 1815 mMixBuffer = new int16_t[mFrameCount * 2]; 1816 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1817 1818 // force reconfiguration of effect chains and engines to take new buffer size and audio 1819 // parameters into account 1820 // Note that mLock is not held when readOutputParameters() is called from the constructor 1821 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1822 // matter. 1823 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1824 Vector< sp<EffectChain> > effectChains = mEffectChains; 1825 for (size_t i = 0; i < effectChains.size(); i ++) { 1826 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1827 } 1828} 1829 1830status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1831{ 1832 if (halFrames == NULL || dspFrames == NULL) { 1833 return BAD_VALUE; 1834 } 1835 Mutex::Autolock _l(mLock); 1836 if (initCheck() != NO_ERROR) { 1837 return INVALID_OPERATION; 1838 } 1839 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1840 1841 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1842} 1843 1844uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1845{ 1846 Mutex::Autolock _l(mLock); 1847 uint32_t result = 0; 1848 if (getEffectChain_l(sessionId) != 0) { 1849 result = EFFECT_SESSION; 1850 } 1851 1852 for (size_t i = 0; i < mTracks.size(); ++i) { 1853 sp<Track> track = mTracks[i]; 1854 if (sessionId == track->sessionId() && 1855 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1856 result |= TRACK_SESSION; 1857 break; 1858 } 1859 } 1860 1861 return result; 1862} 1863 1864uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1865{ 1866 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1867 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1868 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1869 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1870 } 1871 for (size_t i = 0; i < mTracks.size(); i++) { 1872 sp<Track> track = mTracks[i]; 1873 if (sessionId == track->sessionId() && 1874 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1875 return AudioSystem::getStrategyForStream(track->streamType()); 1876 } 1877 } 1878 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1879} 1880 1881 1882AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1883{ 1884 Mutex::Autolock _l(mLock); 1885 return mOutput; 1886} 1887 1888AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1889{ 1890 Mutex::Autolock _l(mLock); 1891 AudioStreamOut *output = mOutput; 1892 mOutput = NULL; 1893 return output; 1894} 1895 1896// this method must always be called either with ThreadBase mLock held or inside the thread loop 1897audio_stream_t* AudioFlinger::PlaybackThread::stream() 1898{ 1899 if (mOutput == NULL) { 1900 return NULL; 1901 } 1902 return &mOutput->stream->common; 1903} 1904 1905uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1906{ 1907 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1908 // decoding and transfer time. So sleeping for half of the latency would likely cause 1909 // underruns 1910 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1911 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1912 } else { 1913 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1914 } 1915} 1916 1917// ---------------------------------------------------------------------------- 1918 1919AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1920 audio_io_handle_t id, uint32_t device, type_t type) 1921 : PlaybackThread(audioFlinger, output, id, device, type), 1922 mAudioMixer(new AudioMixer(mFrameCount, mSampleRate)), 1923 mPrevMixerStatus(MIXER_IDLE) 1924{ 1925 // FIXME - Current mixer implementation only supports stereo output 1926 if (mChannelCount == 1) { 1927 ALOGE("Invalid audio hardware channel count"); 1928 } 1929} 1930 1931AudioFlinger::MixerThread::~MixerThread() 1932{ 1933 delete mAudioMixer; 1934} 1935 1936class CpuStats { 1937public: 1938 void sample(); 1939#ifdef DEBUG_CPU_USAGE 1940private: 1941 ThreadCpuUsage mCpu; 1942#endif 1943}; 1944 1945void CpuStats::sample() { 1946#ifdef DEBUG_CPU_USAGE 1947 const CentralTendencyStatistics& stats = mCpu.statistics(); 1948 mCpu.sampleAndEnable(); 1949 unsigned n = stats.n(); 1950 // mCpu.elapsed() is expensive, so don't call it every loop 1951 if ((n & 127) == 1) { 1952 long long elapsed = mCpu.elapsed(); 1953 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1954 double perLoop = elapsed / (double) n; 1955 double perLoop100 = perLoop * 0.01; 1956 double mean = stats.mean(); 1957 double stddev = stats.stddev(); 1958 double minimum = stats.minimum(); 1959 double maximum = stats.maximum(); 1960 mCpu.resetStatistics(); 1961 ALOGI("CPU usage over past %.1f secs (%u mixer loops at %.1f mean ms per loop):\n us per mix loop: mean=%.0f stddev=%.0f min=%.0f max=%.0f\n %% of wall: mean=%.1f stddev=%.1f min=%.1f max=%.1f", 1962 elapsed * .000000001, n, perLoop * .000001, 1963 mean * .001, 1964 stddev * .001, 1965 minimum * .001, 1966 maximum * .001, 1967 mean / perLoop100, 1968 stddev / perLoop100, 1969 minimum / perLoop100, 1970 maximum / perLoop100); 1971 } 1972 } 1973#endif 1974}; 1975 1976void AudioFlinger::PlaybackThread::checkSilentMode_l() 1977{ 1978 if (!mMasterMute) { 1979 char value[PROPERTY_VALUE_MAX]; 1980 if (property_get("ro.audio.silent", value, "0") > 0) { 1981 char *endptr; 1982 unsigned long ul = strtoul(value, &endptr, 0); 1983 if (*endptr == '\0' && ul != 0) { 1984 ALOGD("Silence is golden"); 1985 // The setprop command will not allow a property to be changed after 1986 // the first time it is set, so we don't have to worry about un-muting. 1987 setMasterMute_l(true); 1988 } 1989 } 1990 } 1991} 1992 1993bool AudioFlinger::MixerThread::threadLoop() 1994{ 1995 Vector< sp<Track> > tracksToRemove; 1996 mixer_state mixerStatus = MIXER_IDLE; 1997 nsecs_t standbyTime = systemTime(); 1998 size_t mixBufferSize = mFrameCount * mFrameSize; 1999 // FIXME: Relaxed timing because of a certain device that can't meet latency 2000 // Should be reduced to 2x after the vendor fixes the driver issue 2001 // increase threshold again due to low power audio mode. The way this warning threshold is 2002 // calculated and its usefulness should be reconsidered anyway. 2003 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2004 nsecs_t lastWarning = 0; 2005 bool longStandbyExit = false; 2006 uint32_t activeSleepTime = activeSleepTimeUs(); 2007 uint32_t idleSleepTime = idleSleepTimeUs(); 2008 uint32_t sleepTime = idleSleepTime; 2009 uint32_t sleepTimeShift = 0; 2010 Vector< sp<EffectChain> > effectChains; 2011 CpuStats cpuStats; 2012 2013 acquireWakeLock(); 2014 2015 while (!exitPending()) 2016 { 2017 cpuStats.sample(); 2018 processConfigEvents(); 2019 2020 mixerStatus = MIXER_IDLE; 2021 { // scope for mLock 2022 2023 Mutex::Autolock _l(mLock); 2024 2025 if (checkForNewParameters_l()) { 2026 mixBufferSize = mFrameCount * mFrameSize; 2027 // FIXME: Relaxed timing because of a certain device that can't meet latency 2028 // Should be reduced to 2x after the vendor fixes the driver issue 2029 // increase threshold again due to low power audio mode. The way this warning 2030 // threshold is calculated and its usefulness should be reconsidered anyway. 2031 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2032 activeSleepTime = activeSleepTimeUs(); 2033 idleSleepTime = idleSleepTimeUs(); 2034 } 2035 2036 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 2037 2038 // put audio hardware into standby after short delay 2039 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 2040 mSuspended)) { 2041 if (!mStandby) { 2042 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2043 mOutput->stream->common.standby(&mOutput->stream->common); 2044 mStandby = true; 2045 mBytesWritten = 0; 2046 } 2047 2048 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 2049 // we're about to wait, flush the binder command buffer 2050 IPCThreadState::self()->flushCommands(); 2051 2052 if (exitPending()) break; 2053 2054 releaseWakeLock_l(); 2055 // wait until we have something to do... 2056 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2057 mWaitWorkCV.wait(mLock); 2058 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2059 acquireWakeLock_l(); 2060 2061 mPrevMixerStatus = MIXER_IDLE; 2062 checkSilentMode_l(); 2063 2064 standbyTime = systemTime() + mStandbyTimeInNsecs; 2065 sleepTime = idleSleepTime; 2066 sleepTimeShift = 0; 2067 continue; 2068 } 2069 } 2070 2071 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 2072 2073 // prevent any changes in effect chain list and in each effect chain 2074 // during mixing and effect process as the audio buffers could be deleted 2075 // or modified if an effect is created or deleted 2076 lockEffectChains_l(effectChains); 2077 } 2078 2079 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2080 // obtain the presentation timestamp of the next output buffer 2081 int64_t pts; 2082 status_t status = INVALID_OPERATION; 2083 2084 if (NULL != mOutput->stream->get_next_write_timestamp) { 2085 status = mOutput->stream->get_next_write_timestamp( 2086 mOutput->stream, &pts); 2087 } 2088 2089 if (status != NO_ERROR) { 2090 pts = AudioBufferProvider::kInvalidPTS; 2091 } 2092 2093 // mix buffers... 2094 mAudioMixer->process(pts); 2095 // increase sleep time progressively when application underrun condition clears. 2096 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2097 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2098 // such that we would underrun the audio HAL. 2099 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2100 sleepTimeShift--; 2101 } 2102 sleepTime = 0; 2103 standbyTime = systemTime() + mStandbyTimeInNsecs; 2104 //TODO: delay standby when effects have a tail 2105 } else { 2106 // If no tracks are ready, sleep once for the duration of an output 2107 // buffer size, then write 0s to the output 2108 if (sleepTime == 0) { 2109 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2110 sleepTime = activeSleepTime >> sleepTimeShift; 2111 if (sleepTime < kMinThreadSleepTimeUs) { 2112 sleepTime = kMinThreadSleepTimeUs; 2113 } 2114 // reduce sleep time in case of consecutive application underruns to avoid 2115 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2116 // duration we would end up writing less data than needed by the audio HAL if 2117 // the condition persists. 2118 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2119 sleepTimeShift++; 2120 } 2121 } else { 2122 sleepTime = idleSleepTime; 2123 } 2124 } else if (mBytesWritten != 0 || 2125 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2126 memset (mMixBuffer, 0, mixBufferSize); 2127 sleepTime = 0; 2128 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2129 } 2130 // TODO add standby time extension fct of effect tail 2131 } 2132 2133 if (mSuspended) { 2134 sleepTime = suspendSleepTimeUs(); 2135 } 2136 // sleepTime == 0 means we must write to audio hardware 2137 if (sleepTime == 0) { 2138 for (size_t i = 0; i < effectChains.size(); i ++) { 2139 effectChains[i]->process_l(); 2140 } 2141 // enable changes in effect chain 2142 unlockEffectChains(effectChains); 2143 mLastWriteTime = systemTime(); 2144 mInWrite = true; 2145 mBytesWritten += mixBufferSize; 2146 2147 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2148 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2149 mNumWrites++; 2150 mInWrite = false; 2151 nsecs_t now = systemTime(); 2152 nsecs_t delta = now - mLastWriteTime; 2153 if (!mStandby && delta > maxPeriod) { 2154 mNumDelayedWrites++; 2155 if ((now - lastWarning) > kWarningThrottleNs) { 2156 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2157 ns2ms(delta), mNumDelayedWrites, this); 2158 lastWarning = now; 2159 } 2160 if (mStandby) { 2161 longStandbyExit = true; 2162 } 2163 } 2164 mStandby = false; 2165 } else { 2166 // enable changes in effect chain 2167 unlockEffectChains(effectChains); 2168 usleep(sleepTime); 2169 } 2170 2171 // finally let go of all our tracks, without the lock held 2172 // since we can't guarantee the destructors won't acquire that 2173 // same lock. 2174 tracksToRemove.clear(); 2175 2176 // Effect chains will be actually deleted here if they were removed from 2177 // mEffectChains list during mixing or effects processing 2178 effectChains.clear(); 2179 } 2180 2181 if (!mStandby) { 2182 mOutput->stream->common.standby(&mOutput->stream->common); 2183 } 2184 2185 releaseWakeLock(); 2186 2187 ALOGV("Thread %p type %d exiting", this, mType); 2188 return false; 2189} 2190 2191// prepareTracks_l() must be called with ThreadBase::mLock held 2192AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2193 const SortedVector< wp<Track> >& activeTracks, Vector< sp<Track> > *tracksToRemove) 2194{ 2195 2196 mixer_state mixerStatus = MIXER_IDLE; 2197 // find out which tracks need to be processed 2198 size_t count = activeTracks.size(); 2199 size_t mixedTracks = 0; 2200 size_t tracksWithEffect = 0; 2201 2202 float masterVolume = mMasterVolume; 2203 bool masterMute = mMasterMute; 2204 2205 if (masterMute) { 2206 masterVolume = 0; 2207 } 2208 // Delegate master volume control to effect in output mix effect chain if needed 2209 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2210 if (chain != 0) { 2211 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2212 chain->setVolume_l(&v, &v); 2213 masterVolume = (float)((v + (1 << 23)) >> 24); 2214 chain.clear(); 2215 } 2216 2217 for (size_t i=0 ; i<count ; i++) { 2218 sp<Track> t = activeTracks[i].promote(); 2219 if (t == 0) continue; 2220 2221 // this const just means the local variable doesn't change 2222 Track* const track = t.get(); 2223 audio_track_cblk_t* cblk = track->cblk(); 2224 2225 // The first time a track is added we wait 2226 // for all its buffers to be filled before processing it 2227 int name = track->name(); 2228 // make sure that we have enough frames to mix one full buffer. 2229 // enforce this condition only once to enable draining the buffer in case the client 2230 // app does not call stop() and relies on underrun to stop: 2231 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2232 // during last round 2233 uint32_t minFrames = 1; 2234 if (!track->isStopped() && !track->isPausing() && 2235 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2236 if (t->sampleRate() == (int)mSampleRate) { 2237 minFrames = mFrameCount; 2238 } else { 2239 // +1 for rounding and +1 for additional sample needed for interpolation 2240 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2241 // add frames already consumed but not yet released by the resampler 2242 // because cblk->framesReady() will include these frames 2243 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2244 // the minimum track buffer size is normally twice the number of frames necessary 2245 // to fill one buffer and the resampler should not leave more than one buffer worth 2246 // of unreleased frames after each pass, but just in case... 2247 ALOG_ASSERT(minFrames <= cblk->frameCount); 2248 } 2249 } 2250 if ((track->framesReady() >= minFrames) && track->isReady() && 2251 !track->isPaused() && !track->isTerminated()) 2252 { 2253 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2254 2255 mixedTracks++; 2256 2257 // track->mainBuffer() != mMixBuffer means there is an effect chain 2258 // connected to the track 2259 chain.clear(); 2260 if (track->mainBuffer() != mMixBuffer) { 2261 chain = getEffectChain_l(track->sessionId()); 2262 // Delegate volume control to effect in track effect chain if needed 2263 if (chain != 0) { 2264 tracksWithEffect++; 2265 } else { 2266 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2267 name, track->sessionId()); 2268 } 2269 } 2270 2271 2272 int param = AudioMixer::VOLUME; 2273 if (track->mFillingUpStatus == Track::FS_FILLED) { 2274 // no ramp for the first volume setting 2275 track->mFillingUpStatus = Track::FS_ACTIVE; 2276 if (track->mState == TrackBase::RESUMING) { 2277 track->mState = TrackBase::ACTIVE; 2278 param = AudioMixer::RAMP_VOLUME; 2279 } 2280 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2281 } else if (cblk->server != 0) { 2282 // If the track is stopped before the first frame was mixed, 2283 // do not apply ramp 2284 param = AudioMixer::RAMP_VOLUME; 2285 } 2286 2287 // compute volume for this track 2288 uint32_t vl, vr, va; 2289 if (track->isMuted() || track->isPausing() || 2290 mStreamTypes[track->streamType()].mute) { 2291 vl = vr = va = 0; 2292 if (track->isPausing()) { 2293 track->setPaused(); 2294 } 2295 } else { 2296 2297 // read original volumes with volume control 2298 float typeVolume = mStreamTypes[track->streamType()].volume; 2299 float v = masterVolume * typeVolume; 2300 uint32_t vlr = cblk->getVolumeLR(); 2301 vl = vlr & 0xFFFF; 2302 vr = vlr >> 16; 2303 // track volumes come from shared memory, so can't be trusted and must be clamped 2304 if (vl > MAX_GAIN_INT) { 2305 ALOGV("Track left volume out of range: %04X", vl); 2306 vl = MAX_GAIN_INT; 2307 } 2308 if (vr > MAX_GAIN_INT) { 2309 ALOGV("Track right volume out of range: %04X", vr); 2310 vr = MAX_GAIN_INT; 2311 } 2312 // now apply the master volume and stream type volume 2313 vl = (uint32_t)(v * vl) << 12; 2314 vr = (uint32_t)(v * vr) << 12; 2315 // assuming master volume and stream type volume each go up to 1.0, 2316 // vl and vr are now in 8.24 format 2317 2318 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2319 // send level comes from shared memory and so may be corrupt 2320 if (sendLevel > MAX_GAIN_INT) { 2321 ALOGV("Track send level out of range: %04X", sendLevel); 2322 sendLevel = MAX_GAIN_INT; 2323 } 2324 va = (uint32_t)(v * sendLevel); 2325 } 2326 // Delegate volume control to effect in track effect chain if needed 2327 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2328 // Do not ramp volume if volume is controlled by effect 2329 param = AudioMixer::VOLUME; 2330 track->mHasVolumeController = true; 2331 } else { 2332 // force no volume ramp when volume controller was just disabled or removed 2333 // from effect chain to avoid volume spike 2334 if (track->mHasVolumeController) { 2335 param = AudioMixer::VOLUME; 2336 } 2337 track->mHasVolumeController = false; 2338 } 2339 2340 // Convert volumes from 8.24 to 4.12 format 2341 // This additional clamping is needed in case chain->setVolume_l() overshot 2342 vl = (vl + (1 << 11)) >> 12; 2343 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2344 vr = (vr + (1 << 11)) >> 12; 2345 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2346 2347 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2348 2349 // XXX: these things DON'T need to be done each time 2350 mAudioMixer->setBufferProvider(name, track); 2351 mAudioMixer->enable(name); 2352 2353 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2354 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2355 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2356 mAudioMixer->setParameter( 2357 name, 2358 AudioMixer::TRACK, 2359 AudioMixer::FORMAT, (void *)track->format()); 2360 mAudioMixer->setParameter( 2361 name, 2362 AudioMixer::TRACK, 2363 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2364 mAudioMixer->setParameter( 2365 name, 2366 AudioMixer::RESAMPLE, 2367 AudioMixer::SAMPLE_RATE, 2368 (void *)(cblk->sampleRate)); 2369 mAudioMixer->setParameter( 2370 name, 2371 AudioMixer::TRACK, 2372 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2373 mAudioMixer->setParameter( 2374 name, 2375 AudioMixer::TRACK, 2376 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2377 2378 // reset retry count 2379 track->mRetryCount = kMaxTrackRetries; 2380 // If one track is ready, set the mixer ready if: 2381 // - the mixer was not ready during previous round OR 2382 // - no other track is not ready 2383 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2384 mixerStatus != MIXER_TRACKS_ENABLED) { 2385 mixerStatus = MIXER_TRACKS_READY; 2386 } 2387 } else { 2388 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2389 if (track->isStopped()) { 2390 track->reset(); 2391 } 2392 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2393 // We have consumed all the buffers of this track. 2394 // Remove it from the list of active tracks. 2395 tracksToRemove->add(track); 2396 } else { 2397 // No buffers for this track. Give it a few chances to 2398 // fill a buffer, then remove it from active list. 2399 if (--(track->mRetryCount) <= 0) { 2400 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2401 tracksToRemove->add(track); 2402 // indicate to client process that the track was disabled because of underrun 2403 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2404 // If one track is not ready, mark the mixer also not ready if: 2405 // - the mixer was ready during previous round OR 2406 // - no other track is ready 2407 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2408 mixerStatus != MIXER_TRACKS_READY) { 2409 mixerStatus = MIXER_TRACKS_ENABLED; 2410 } 2411 } 2412 mAudioMixer->disable(name); 2413 } 2414 } 2415 2416 // remove all the tracks that need to be... 2417 count = tracksToRemove->size(); 2418 if (CC_UNLIKELY(count)) { 2419 for (size_t i=0 ; i<count ; i++) { 2420 const sp<Track>& track = tracksToRemove->itemAt(i); 2421 mActiveTracks.remove(track); 2422 if (track->mainBuffer() != mMixBuffer) { 2423 chain = getEffectChain_l(track->sessionId()); 2424 if (chain != 0) { 2425 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2426 chain->decActiveTrackCnt(); 2427 } 2428 } 2429 if (track->isTerminated()) { 2430 removeTrack_l(track); 2431 } 2432 } 2433 } 2434 2435 // mix buffer must be cleared if all tracks are connected to an 2436 // effect chain as in this case the mixer will not write to 2437 // mix buffer and track effects will accumulate into it 2438 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2439 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2440 } 2441 2442 mPrevMixerStatus = mixerStatus; 2443 return mixerStatus; 2444} 2445 2446void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2447{ 2448 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2449 this, streamType, mTracks.size()); 2450 Mutex::Autolock _l(mLock); 2451 2452 size_t size = mTracks.size(); 2453 for (size_t i = 0; i < size; i++) { 2454 sp<Track> t = mTracks[i]; 2455 if (t->streamType() == streamType) { 2456 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2457 t->mCblk->cv.signal(); 2458 } 2459 } 2460} 2461 2462void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2463{ 2464 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2465 this, streamType, valid); 2466 Mutex::Autolock _l(mLock); 2467 2468 mStreamTypes[streamType].valid = valid; 2469} 2470 2471// getTrackName_l() must be called with ThreadBase::mLock held 2472int AudioFlinger::MixerThread::getTrackName_l() 2473{ 2474 return mAudioMixer->getTrackName(); 2475} 2476 2477// deleteTrackName_l() must be called with ThreadBase::mLock held 2478void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2479{ 2480 ALOGV("remove track (%d) and delete from mixer", name); 2481 mAudioMixer->deleteTrackName(name); 2482} 2483 2484// checkForNewParameters_l() must be called with ThreadBase::mLock held 2485bool AudioFlinger::MixerThread::checkForNewParameters_l() 2486{ 2487 bool reconfig = false; 2488 2489 while (!mNewParameters.isEmpty()) { 2490 status_t status = NO_ERROR; 2491 String8 keyValuePair = mNewParameters[0]; 2492 AudioParameter param = AudioParameter(keyValuePair); 2493 int value; 2494 2495 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2496 reconfig = true; 2497 } 2498 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2499 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2500 status = BAD_VALUE; 2501 } else { 2502 reconfig = true; 2503 } 2504 } 2505 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2506 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2507 status = BAD_VALUE; 2508 } else { 2509 reconfig = true; 2510 } 2511 } 2512 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2513 // do not accept frame count changes if tracks are open as the track buffer 2514 // size depends on frame count and correct behavior would not be guaranteed 2515 // if frame count is changed after track creation 2516 if (!mTracks.isEmpty()) { 2517 status = INVALID_OPERATION; 2518 } else { 2519 reconfig = true; 2520 } 2521 } 2522 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2523 // when changing the audio output device, call addBatteryData to notify 2524 // the change 2525 if ((int)mDevice != value) { 2526 uint32_t params = 0; 2527 // check whether speaker is on 2528 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2529 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2530 } 2531 2532 int deviceWithoutSpeaker 2533 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2534 // check if any other device (except speaker) is on 2535 if (value & deviceWithoutSpeaker ) { 2536 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2537 } 2538 2539 if (params != 0) { 2540 addBatteryData(params); 2541 } 2542 } 2543 2544 // forward device change to effects that have requested to be 2545 // aware of attached audio device. 2546 mDevice = (uint32_t)value; 2547 for (size_t i = 0; i < mEffectChains.size(); i++) { 2548 mEffectChains[i]->setDevice_l(mDevice); 2549 } 2550 } 2551 2552 if (status == NO_ERROR) { 2553 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2554 keyValuePair.string()); 2555 if (!mStandby && status == INVALID_OPERATION) { 2556 mOutput->stream->common.standby(&mOutput->stream->common); 2557 mStandby = true; 2558 mBytesWritten = 0; 2559 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2560 keyValuePair.string()); 2561 } 2562 if (status == NO_ERROR && reconfig) { 2563 delete mAudioMixer; 2564 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2565 mAudioMixer = NULL; 2566 readOutputParameters(); 2567 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2568 for (size_t i = 0; i < mTracks.size() ; i++) { 2569 int name = getTrackName_l(); 2570 if (name < 0) break; 2571 mTracks[i]->mName = name; 2572 // limit track sample rate to 2 x new output sample rate 2573 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2574 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2575 } 2576 } 2577 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2578 } 2579 } 2580 2581 mNewParameters.removeAt(0); 2582 2583 mParamStatus = status; 2584 mParamCond.signal(); 2585 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2586 // already timed out waiting for the status and will never signal the condition. 2587 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2588 } 2589 return reconfig; 2590} 2591 2592status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2593{ 2594 const size_t SIZE = 256; 2595 char buffer[SIZE]; 2596 String8 result; 2597 2598 PlaybackThread::dumpInternals(fd, args); 2599 2600 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2601 result.append(buffer); 2602 write(fd, result.string(), result.size()); 2603 return NO_ERROR; 2604} 2605 2606uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2607{ 2608 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2609} 2610 2611uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2612{ 2613 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2614} 2615 2616// ---------------------------------------------------------------------------- 2617AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2618 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2619 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2620 // mLeftVolFloat, mRightVolFloat 2621 // mLeftVolShort, mRightVolShort 2622{ 2623} 2624 2625AudioFlinger::DirectOutputThread::~DirectOutputThread() 2626{ 2627} 2628 2629void AudioFlinger::DirectOutputThread::applyVolume(uint16_t leftVol, uint16_t rightVol, bool ramp) 2630{ 2631 // Do not apply volume on compressed audio 2632 if (!audio_is_linear_pcm(mFormat)) { 2633 return; 2634 } 2635 2636 // convert to signed 16 bit before volume calculation 2637 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2638 size_t count = mFrameCount * mChannelCount; 2639 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2640 int16_t *dst = mMixBuffer + count-1; 2641 while(count--) { 2642 *dst-- = (int16_t)(*src--^0x80) << 8; 2643 } 2644 } 2645 2646 size_t frameCount = mFrameCount; 2647 int16_t *out = mMixBuffer; 2648 if (ramp) { 2649 if (mChannelCount == 1) { 2650 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2651 int32_t vlInc = d / (int32_t)frameCount; 2652 int32_t vl = ((int32_t)mLeftVolShort << 16); 2653 do { 2654 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2655 out++; 2656 vl += vlInc; 2657 } while (--frameCount); 2658 2659 } else { 2660 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2661 int32_t vlInc = d / (int32_t)frameCount; 2662 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2663 int32_t vrInc = d / (int32_t)frameCount; 2664 int32_t vl = ((int32_t)mLeftVolShort << 16); 2665 int32_t vr = ((int32_t)mRightVolShort << 16); 2666 do { 2667 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2668 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2669 out += 2; 2670 vl += vlInc; 2671 vr += vrInc; 2672 } while (--frameCount); 2673 } 2674 } else { 2675 if (mChannelCount == 1) { 2676 do { 2677 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2678 out++; 2679 } while (--frameCount); 2680 } else { 2681 do { 2682 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2683 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2684 out += 2; 2685 } while (--frameCount); 2686 } 2687 } 2688 2689 // convert back to unsigned 8 bit after volume calculation 2690 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2691 size_t count = mFrameCount * mChannelCount; 2692 int16_t *src = mMixBuffer; 2693 uint8_t *dst = (uint8_t *)mMixBuffer; 2694 while(count--) { 2695 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2696 } 2697 } 2698 2699 mLeftVolShort = leftVol; 2700 mRightVolShort = rightVol; 2701} 2702 2703bool AudioFlinger::DirectOutputThread::threadLoop() 2704{ 2705 mixer_state mixerStatus = MIXER_IDLE; 2706 sp<Track> trackToRemove; 2707 sp<Track> activeTrack; 2708 nsecs_t standbyTime = systemTime(); 2709 size_t mixBufferSize = mFrameCount*mFrameSize; 2710 uint32_t activeSleepTime = activeSleepTimeUs(); 2711 uint32_t idleSleepTime = idleSleepTimeUs(); 2712 uint32_t sleepTime = idleSleepTime; 2713 // use shorter standby delay as on normal output to release 2714 // hardware resources as soon as possible 2715 nsecs_t standbyDelay = microseconds(activeSleepTime*2); 2716 2717 acquireWakeLock(); 2718 2719 while (!exitPending()) 2720 { 2721 bool rampVolume; 2722 uint16_t leftVol; 2723 uint16_t rightVol; 2724 Vector< sp<EffectChain> > effectChains; 2725 2726 processConfigEvents(); 2727 2728 mixerStatus = MIXER_IDLE; 2729 2730 { // scope for the mLock 2731 2732 Mutex::Autolock _l(mLock); 2733 2734 if (checkForNewParameters_l()) { 2735 mixBufferSize = mFrameCount*mFrameSize; 2736 activeSleepTime = activeSleepTimeUs(); 2737 idleSleepTime = idleSleepTimeUs(); 2738 standbyDelay = microseconds(activeSleepTime*2); 2739 } 2740 2741 // put audio hardware into standby after short delay 2742 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2743 mSuspended)) { 2744 // wait until we have something to do... 2745 if (!mStandby) { 2746 ALOGV("Audio hardware entering standby, mixer %p, mSuspended %d", this, mSuspended); 2747 mOutput->stream->common.standby(&mOutput->stream->common); 2748 mStandby = true; 2749 mBytesWritten = 0; 2750 } 2751 2752 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2753 // we're about to wait, flush the binder command buffer 2754 IPCThreadState::self()->flushCommands(); 2755 2756 if (exitPending()) break; 2757 2758 releaseWakeLock_l(); 2759 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2760 mWaitWorkCV.wait(mLock); 2761 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2762 acquireWakeLock_l(); 2763 2764 checkSilentMode_l(); 2765 2766 standbyTime = systemTime() + standbyDelay; 2767 sleepTime = idleSleepTime; 2768 continue; 2769 } 2770 } 2771 2772 effectChains = mEffectChains; 2773 2774 // find out which tracks need to be processed 2775 if (mActiveTracks.size() != 0) { 2776 sp<Track> t = mActiveTracks[0].promote(); 2777 if (t == 0) continue; 2778 2779 Track* const track = t.get(); 2780 audio_track_cblk_t* cblk = track->cblk(); 2781 2782 // The first time a track is added we wait 2783 // for all its buffers to be filled before processing it 2784 if (cblk->framesReady() && track->isReady() && 2785 !track->isPaused() && !track->isTerminated()) 2786 { 2787 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2788 2789 if (track->mFillingUpStatus == Track::FS_FILLED) { 2790 track->mFillingUpStatus = Track::FS_ACTIVE; 2791 mLeftVolFloat = mRightVolFloat = 0; 2792 mLeftVolShort = mRightVolShort = 0; 2793 if (track->mState == TrackBase::RESUMING) { 2794 track->mState = TrackBase::ACTIVE; 2795 rampVolume = true; 2796 } 2797 } else if (cblk->server != 0) { 2798 // If the track is stopped before the first frame was mixed, 2799 // do not apply ramp 2800 rampVolume = true; 2801 } 2802 // compute volume for this track 2803 float left, right; 2804 if (track->isMuted() || mMasterMute || track->isPausing() || 2805 mStreamTypes[track->streamType()].mute) { 2806 left = right = 0; 2807 if (track->isPausing()) { 2808 track->setPaused(); 2809 } 2810 } else { 2811 float typeVolume = mStreamTypes[track->streamType()].volume; 2812 float v = mMasterVolume * typeVolume; 2813 uint32_t vlr = cblk->getVolumeLR(); 2814 float v_clamped = v * (vlr & 0xFFFF); 2815 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2816 left = v_clamped/MAX_GAIN; 2817 v_clamped = v * (vlr >> 16); 2818 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2819 right = v_clamped/MAX_GAIN; 2820 } 2821 2822 if (left != mLeftVolFloat || right != mRightVolFloat) { 2823 mLeftVolFloat = left; 2824 mRightVolFloat = right; 2825 2826 // If audio HAL implements volume control, 2827 // force software volume to nominal value 2828 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2829 left = 1.0f; 2830 right = 1.0f; 2831 } 2832 2833 // Convert volumes from float to 8.24 2834 uint32_t vl = (uint32_t)(left * (1 << 24)); 2835 uint32_t vr = (uint32_t)(right * (1 << 24)); 2836 2837 // Delegate volume control to effect in track effect chain if needed 2838 // only one effect chain can be present on DirectOutputThread, so if 2839 // there is one, the track is connected to it 2840 if (!effectChains.isEmpty()) { 2841 // Do not ramp volume if volume is controlled by effect 2842 if(effectChains[0]->setVolume_l(&vl, &vr)) { 2843 rampVolume = false; 2844 } 2845 } 2846 2847 // Convert volumes from 8.24 to 4.12 format 2848 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2849 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2850 leftVol = (uint16_t)v_clamped; 2851 v_clamped = (vr + (1 << 11)) >> 12; 2852 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2853 rightVol = (uint16_t)v_clamped; 2854 } else { 2855 leftVol = mLeftVolShort; 2856 rightVol = mRightVolShort; 2857 rampVolume = false; 2858 } 2859 2860 // reset retry count 2861 track->mRetryCount = kMaxTrackRetriesDirect; 2862 activeTrack = t; 2863 mixerStatus = MIXER_TRACKS_READY; 2864 } else { 2865 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2866 if (track->isStopped()) { 2867 track->reset(); 2868 } 2869 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2870 // We have consumed all the buffers of this track. 2871 // Remove it from the list of active tracks. 2872 trackToRemove = track; 2873 } else { 2874 // No buffers for this track. Give it a few chances to 2875 // fill a buffer, then remove it from active list. 2876 if (--(track->mRetryCount) <= 0) { 2877 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2878 trackToRemove = track; 2879 } else { 2880 mixerStatus = MIXER_TRACKS_ENABLED; 2881 } 2882 } 2883 } 2884 } 2885 2886 // remove all the tracks that need to be... 2887 if (CC_UNLIKELY(trackToRemove != 0)) { 2888 mActiveTracks.remove(trackToRemove); 2889 if (!effectChains.isEmpty()) { 2890 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2891 trackToRemove->sessionId()); 2892 effectChains[0]->decActiveTrackCnt(); 2893 } 2894 if (trackToRemove->isTerminated()) { 2895 removeTrack_l(trackToRemove); 2896 } 2897 } 2898 2899 lockEffectChains_l(effectChains); 2900 } 2901 2902 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2903 AudioBufferProvider::Buffer buffer; 2904 size_t frameCount = mFrameCount; 2905 int8_t *curBuf = (int8_t *)mMixBuffer; 2906 // output audio to hardware 2907 while (frameCount) { 2908 buffer.frameCount = frameCount; 2909 activeTrack->getNextBuffer(&buffer, 2910 AudioBufferProvider::kInvalidPTS); 2911 if (CC_UNLIKELY(buffer.raw == NULL)) { 2912 memset(curBuf, 0, frameCount * mFrameSize); 2913 break; 2914 } 2915 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2916 frameCount -= buffer.frameCount; 2917 curBuf += buffer.frameCount * mFrameSize; 2918 activeTrack->releaseBuffer(&buffer); 2919 } 2920 sleepTime = 0; 2921 standbyTime = systemTime() + standbyDelay; 2922 } else { 2923 if (sleepTime == 0) { 2924 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2925 sleepTime = activeSleepTime; 2926 } else { 2927 sleepTime = idleSleepTime; 2928 } 2929 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2930 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2931 sleepTime = 0; 2932 } 2933 } 2934 2935 if (mSuspended) { 2936 sleepTime = suspendSleepTimeUs(); 2937 } 2938 // sleepTime == 0 means we must write to audio hardware 2939 if (sleepTime == 0) { 2940 if (mixerStatus == MIXER_TRACKS_READY) { 2941 applyVolume(leftVol, rightVol, rampVolume); 2942 } 2943 for (size_t i = 0; i < effectChains.size(); i ++) { 2944 effectChains[i]->process_l(); 2945 } 2946 unlockEffectChains(effectChains); 2947 2948 mLastWriteTime = systemTime(); 2949 mInWrite = true; 2950 mBytesWritten += mixBufferSize; 2951 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2952 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2953 mNumWrites++; 2954 mInWrite = false; 2955 mStandby = false; 2956 } else { 2957 unlockEffectChains(effectChains); 2958 usleep(sleepTime); 2959 } 2960 2961 // finally let go of removed track, without the lock held 2962 // since we can't guarantee the destructors won't acquire that 2963 // same lock. 2964 trackToRemove.clear(); 2965 activeTrack.clear(); 2966 2967 // Effect chains will be actually deleted here if they were removed from 2968 // mEffectChains list during mixing or effects processing 2969 effectChains.clear(); 2970 } 2971 2972 if (!mStandby) { 2973 mOutput->stream->common.standby(&mOutput->stream->common); 2974 } 2975 2976 releaseWakeLock(); 2977 2978 ALOGV("Thread %p type %d exiting", this, mType); 2979 return false; 2980} 2981 2982// getTrackName_l() must be called with ThreadBase::mLock held 2983int AudioFlinger::DirectOutputThread::getTrackName_l() 2984{ 2985 return 0; 2986} 2987 2988// deleteTrackName_l() must be called with ThreadBase::mLock held 2989void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2990{ 2991} 2992 2993// checkForNewParameters_l() must be called with ThreadBase::mLock held 2994bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2995{ 2996 bool reconfig = false; 2997 2998 while (!mNewParameters.isEmpty()) { 2999 status_t status = NO_ERROR; 3000 String8 keyValuePair = mNewParameters[0]; 3001 AudioParameter param = AudioParameter(keyValuePair); 3002 int value; 3003 3004 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3005 // do not accept frame count changes if tracks are open as the track buffer 3006 // size depends on frame count and correct behavior would not be garantied 3007 // if frame count is changed after track creation 3008 if (!mTracks.isEmpty()) { 3009 status = INVALID_OPERATION; 3010 } else { 3011 reconfig = true; 3012 } 3013 } 3014 if (status == NO_ERROR) { 3015 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3016 keyValuePair.string()); 3017 if (!mStandby && status == INVALID_OPERATION) { 3018 mOutput->stream->common.standby(&mOutput->stream->common); 3019 mStandby = true; 3020 mBytesWritten = 0; 3021 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3022 keyValuePair.string()); 3023 } 3024 if (status == NO_ERROR && reconfig) { 3025 readOutputParameters(); 3026 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3027 } 3028 } 3029 3030 mNewParameters.removeAt(0); 3031 3032 mParamStatus = status; 3033 mParamCond.signal(); 3034 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3035 // already timed out waiting for the status and will never signal the condition. 3036 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3037 } 3038 return reconfig; 3039} 3040 3041uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3042{ 3043 uint32_t time; 3044 if (audio_is_linear_pcm(mFormat)) { 3045 time = PlaybackThread::activeSleepTimeUs(); 3046 } else { 3047 time = 10000; 3048 } 3049 return time; 3050} 3051 3052uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3053{ 3054 uint32_t time; 3055 if (audio_is_linear_pcm(mFormat)) { 3056 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3057 } else { 3058 time = 10000; 3059 } 3060 return time; 3061} 3062 3063uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3064{ 3065 uint32_t time; 3066 if (audio_is_linear_pcm(mFormat)) { 3067 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3068 } else { 3069 time = 10000; 3070 } 3071 return time; 3072} 3073 3074 3075// ---------------------------------------------------------------------------- 3076 3077AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3078 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3079 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3080 mWaitTimeMs(UINT_MAX) 3081{ 3082 addOutputTrack(mainThread); 3083} 3084 3085AudioFlinger::DuplicatingThread::~DuplicatingThread() 3086{ 3087 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3088 mOutputTracks[i]->destroy(); 3089 } 3090} 3091 3092bool AudioFlinger::DuplicatingThread::threadLoop() 3093{ 3094 Vector< sp<Track> > tracksToRemove; 3095 mixer_state mixerStatus = MIXER_IDLE; 3096 nsecs_t standbyTime = systemTime(); 3097 size_t mixBufferSize = mFrameCount*mFrameSize; 3098 SortedVector< sp<OutputTrack> > outputTracks; 3099 uint32_t writeFrames = 0; 3100 uint32_t activeSleepTime = activeSleepTimeUs(); 3101 uint32_t idleSleepTime = idleSleepTimeUs(); 3102 uint32_t sleepTime = idleSleepTime; 3103 Vector< sp<EffectChain> > effectChains; 3104 3105 acquireWakeLock(); 3106 3107 while (!exitPending()) 3108 { 3109 processConfigEvents(); 3110 3111 mixerStatus = MIXER_IDLE; 3112 { // scope for the mLock 3113 3114 Mutex::Autolock _l(mLock); 3115 3116 if (checkForNewParameters_l()) { 3117 mixBufferSize = mFrameCount*mFrameSize; 3118 updateWaitTime(); 3119 activeSleepTime = activeSleepTimeUs(); 3120 idleSleepTime = idleSleepTimeUs(); 3121 } 3122 3123 const SortedVector< wp<Track> >& activeTracks = mActiveTracks; 3124 3125 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3126 outputTracks.add(mOutputTracks[i]); 3127 } 3128 3129 // put audio hardware into standby after short delay 3130 if (CC_UNLIKELY((!activeTracks.size() && systemTime() > standbyTime) || 3131 mSuspended)) { 3132 if (!mStandby) { 3133 for (size_t i = 0; i < outputTracks.size(); i++) { 3134 outputTracks[i]->stop(); 3135 } 3136 mStandby = true; 3137 mBytesWritten = 0; 3138 } 3139 3140 if (!activeTracks.size() && mConfigEvents.isEmpty()) { 3141 // we're about to wait, flush the binder command buffer 3142 IPCThreadState::self()->flushCommands(); 3143 outputTracks.clear(); 3144 3145 if (exitPending()) break; 3146 3147 releaseWakeLock_l(); 3148 // wait until we have something to do... 3149 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 3150 mWaitWorkCV.wait(mLock); 3151 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 3152 acquireWakeLock_l(); 3153 3154 checkSilentMode_l(); 3155 3156 standbyTime = systemTime() + mStandbyTimeInNsecs; 3157 sleepTime = idleSleepTime; 3158 continue; 3159 } 3160 } 3161 3162 mixerStatus = prepareTracks_l(activeTracks, &tracksToRemove); 3163 3164 // prevent any changes in effect chain list and in each effect chain 3165 // during mixing and effect process as the audio buffers could be deleted 3166 // or modified if an effect is created or deleted 3167 lockEffectChains_l(effectChains); 3168 } 3169 3170 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 3171 // mix buffers... 3172 if (outputsReady(outputTracks)) { 3173 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3174 } else { 3175 memset(mMixBuffer, 0, mixBufferSize); 3176 } 3177 sleepTime = 0; 3178 writeFrames = mFrameCount; 3179 } else { 3180 if (sleepTime == 0) { 3181 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3182 sleepTime = activeSleepTime; 3183 } else { 3184 sleepTime = idleSleepTime; 3185 } 3186 } else if (mBytesWritten != 0) { 3187 // flush remaining overflow buffers in output tracks 3188 for (size_t i = 0; i < outputTracks.size(); i++) { 3189 if (outputTracks[i]->isActive()) { 3190 sleepTime = 0; 3191 writeFrames = 0; 3192 memset(mMixBuffer, 0, mixBufferSize); 3193 break; 3194 } 3195 } 3196 } 3197 } 3198 3199 if (mSuspended) { 3200 sleepTime = suspendSleepTimeUs(); 3201 } 3202 // sleepTime == 0 means we must write to audio hardware 3203 if (sleepTime == 0) { 3204 for (size_t i = 0; i < effectChains.size(); i ++) { 3205 effectChains[i]->process_l(); 3206 } 3207 // enable changes in effect chain 3208 unlockEffectChains(effectChains); 3209 3210 standbyTime = systemTime() + mStandbyTimeInNsecs; 3211 for (size_t i = 0; i < outputTracks.size(); i++) { 3212 outputTracks[i]->write(mMixBuffer, writeFrames); 3213 } 3214 mStandby = false; 3215 mBytesWritten += mixBufferSize; 3216 } else { 3217 // enable changes in effect chain 3218 unlockEffectChains(effectChains); 3219 usleep(sleepTime); 3220 } 3221 3222 // finally let go of all our tracks, without the lock held 3223 // since we can't guarantee the destructors won't acquire that 3224 // same lock. 3225 tracksToRemove.clear(); 3226 outputTracks.clear(); 3227 3228 // Effect chains will be actually deleted here if they were removed from 3229 // mEffectChains list during mixing or effects processing 3230 effectChains.clear(); 3231 } 3232 3233 releaseWakeLock(); 3234 3235 ALOGV("Thread %p type %d exiting", this, mType); 3236 return false; 3237} 3238 3239void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3240{ 3241 // FIXME explain this formula 3242 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3243 OutputTrack *outputTrack = new OutputTrack(thread, 3244 this, 3245 mSampleRate, 3246 mFormat, 3247 mChannelMask, 3248 frameCount); 3249 if (outputTrack->cblk() != NULL) { 3250 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3251 mOutputTracks.add(outputTrack); 3252 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3253 updateWaitTime(); 3254 } 3255} 3256 3257void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3258{ 3259 Mutex::Autolock _l(mLock); 3260 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3261 if (mOutputTracks[i]->thread() == thread) { 3262 mOutputTracks[i]->destroy(); 3263 mOutputTracks.removeAt(i); 3264 updateWaitTime(); 3265 return; 3266 } 3267 } 3268 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3269} 3270 3271void AudioFlinger::DuplicatingThread::updateWaitTime() 3272{ 3273 mWaitTimeMs = UINT_MAX; 3274 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3275 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3276 if (strong != 0) { 3277 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3278 if (waitTimeMs < mWaitTimeMs) { 3279 mWaitTimeMs = waitTimeMs; 3280 } 3281 } 3282 } 3283} 3284 3285 3286bool AudioFlinger::DuplicatingThread::outputsReady(SortedVector< sp<OutputTrack> > &outputTracks) 3287{ 3288 for (size_t i = 0; i < outputTracks.size(); i++) { 3289 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3290 if (thread == 0) { 3291 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3292 return false; 3293 } 3294 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3295 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3296 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3297 return false; 3298 } 3299 } 3300 return true; 3301} 3302 3303uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3304{ 3305 return (mWaitTimeMs * 1000) / 2; 3306} 3307 3308// ---------------------------------------------------------------------------- 3309 3310// TrackBase constructor must be called with AudioFlinger::mLock held 3311AudioFlinger::ThreadBase::TrackBase::TrackBase( 3312 ThreadBase *thread, 3313 const sp<Client>& client, 3314 uint32_t sampleRate, 3315 audio_format_t format, 3316 uint32_t channelMask, 3317 int frameCount, 3318 const sp<IMemory>& sharedBuffer, 3319 int sessionId) 3320 : RefBase(), 3321 mThread(thread), 3322 mClient(client), 3323 mCblk(NULL), 3324 // mBuffer 3325 // mBufferEnd 3326 mFrameCount(0), 3327 mState(IDLE), 3328 mFormat(format), 3329 mStepServerFailed(false), 3330 mSessionId(sessionId) 3331 // mChannelCount 3332 // mChannelMask 3333{ 3334 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3335 3336 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3337 size_t size = sizeof(audio_track_cblk_t); 3338 uint8_t channelCount = popcount(channelMask); 3339 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3340 if (sharedBuffer == 0) { 3341 size += bufferSize; 3342 } 3343 3344 if (client != NULL) { 3345 mCblkMemory = client->heap()->allocate(size); 3346 if (mCblkMemory != 0) { 3347 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3348 if (mCblk != NULL) { // construct the shared structure in-place. 3349 new(mCblk) audio_track_cblk_t(); 3350 // clear all buffers 3351 mCblk->frameCount = frameCount; 3352 mCblk->sampleRate = sampleRate; 3353 mChannelCount = channelCount; 3354 mChannelMask = channelMask; 3355 if (sharedBuffer == 0) { 3356 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3357 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3358 // Force underrun condition to avoid false underrun callback until first data is 3359 // written to buffer (other flags are cleared) 3360 mCblk->flags = CBLK_UNDERRUN_ON; 3361 } else { 3362 mBuffer = sharedBuffer->pointer(); 3363 } 3364 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3365 } 3366 } else { 3367 ALOGE("not enough memory for AudioTrack size=%u", size); 3368 client->heap()->dump("AudioTrack"); 3369 return; 3370 } 3371 } else { 3372 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3373 // construct the shared structure in-place. 3374 new(mCblk) audio_track_cblk_t(); 3375 // clear all buffers 3376 mCblk->frameCount = frameCount; 3377 mCblk->sampleRate = sampleRate; 3378 mChannelCount = channelCount; 3379 mChannelMask = channelMask; 3380 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3381 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3382 // Force underrun condition to avoid false underrun callback until first data is 3383 // written to buffer (other flags are cleared) 3384 mCblk->flags = CBLK_UNDERRUN_ON; 3385 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3386 } 3387} 3388 3389AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3390{ 3391 if (mCblk != NULL) { 3392 if (mClient == 0) { 3393 delete mCblk; 3394 } else { 3395 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3396 } 3397 } 3398 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3399 if (mClient != 0) { 3400 // Client destructor must run with AudioFlinger mutex locked 3401 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3402 // If the client's reference count drops to zero, the associated destructor 3403 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3404 // relying on the automatic clear() at end of scope. 3405 mClient.clear(); 3406 } 3407} 3408 3409void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3410{ 3411 buffer->raw = NULL; 3412 mFrameCount = buffer->frameCount; 3413 step(); 3414 buffer->frameCount = 0; 3415} 3416 3417bool AudioFlinger::ThreadBase::TrackBase::step() { 3418 bool result; 3419 audio_track_cblk_t* cblk = this->cblk(); 3420 3421 result = cblk->stepServer(mFrameCount); 3422 if (!result) { 3423 ALOGV("stepServer failed acquiring cblk mutex"); 3424 mStepServerFailed = true; 3425 } 3426 return result; 3427} 3428 3429void AudioFlinger::ThreadBase::TrackBase::reset() { 3430 audio_track_cblk_t* cblk = this->cblk(); 3431 3432 cblk->user = 0; 3433 cblk->server = 0; 3434 cblk->userBase = 0; 3435 cblk->serverBase = 0; 3436 mStepServerFailed = false; 3437 ALOGV("TrackBase::reset"); 3438} 3439 3440int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3441 return (int)mCblk->sampleRate; 3442} 3443 3444void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3445 audio_track_cblk_t* cblk = this->cblk(); 3446 size_t frameSize = cblk->frameSize; 3447 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3448 int8_t *bufferEnd = bufferStart + frames * frameSize; 3449 3450 // Check validity of returned pointer in case the track control block would have been corrupted. 3451 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3452 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3453 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3454 server %d, serverBase %d, user %d, userBase %d", 3455 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3456 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3457 return NULL; 3458 } 3459 3460 return bufferStart; 3461} 3462 3463// ---------------------------------------------------------------------------- 3464 3465// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3466AudioFlinger::PlaybackThread::Track::Track( 3467 PlaybackThread *thread, 3468 const sp<Client>& client, 3469 audio_stream_type_t streamType, 3470 uint32_t sampleRate, 3471 audio_format_t format, 3472 uint32_t channelMask, 3473 int frameCount, 3474 const sp<IMemory>& sharedBuffer, 3475 int sessionId) 3476 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3477 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3478 mAuxEffectId(0), mHasVolumeController(false) 3479{ 3480 if (mCblk != NULL) { 3481 if (thread != NULL) { 3482 mName = thread->getTrackName_l(); 3483 mMainBuffer = thread->mixBuffer(); 3484 } 3485 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3486 if (mName < 0) { 3487 ALOGE("no more track names available"); 3488 } 3489 mStreamType = streamType; 3490 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3491 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3492 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3493 } 3494} 3495 3496AudioFlinger::PlaybackThread::Track::~Track() 3497{ 3498 ALOGV("PlaybackThread::Track destructor"); 3499 sp<ThreadBase> thread = mThread.promote(); 3500 if (thread != 0) { 3501 Mutex::Autolock _l(thread->mLock); 3502 mState = TERMINATED; 3503 } 3504} 3505 3506void AudioFlinger::PlaybackThread::Track::destroy() 3507{ 3508 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3509 // by removing it from mTracks vector, so there is a risk that this Tracks's 3510 // destructor is called. As the destructor needs to lock mLock, 3511 // we must acquire a strong reference on this Track before locking mLock 3512 // here so that the destructor is called only when exiting this function. 3513 // On the other hand, as long as Track::destroy() is only called by 3514 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3515 // this Track with its member mTrack. 3516 sp<Track> keep(this); 3517 { // scope for mLock 3518 sp<ThreadBase> thread = mThread.promote(); 3519 if (thread != 0) { 3520 if (!isOutputTrack()) { 3521 if (mState == ACTIVE || mState == RESUMING) { 3522 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3523 3524 // to track the speaker usage 3525 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3526 } 3527 AudioSystem::releaseOutput(thread->id()); 3528 } 3529 Mutex::Autolock _l(thread->mLock); 3530 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3531 playbackThread->destroyTrack_l(this); 3532 } 3533 } 3534} 3535 3536void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3537{ 3538 uint32_t vlr = mCblk->getVolumeLR(); 3539 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", 3540 mName - AudioMixer::TRACK0, 3541 (mClient == 0) ? getpid_cached : mClient->pid(), 3542 mStreamType, 3543 mFormat, 3544 mChannelMask, 3545 mSessionId, 3546 mFrameCount, 3547 mState, 3548 mMute, 3549 mFillingUpStatus, 3550 mCblk->sampleRate, 3551 vlr & 0xFFFF, 3552 vlr >> 16, 3553 mCblk->server, 3554 mCblk->user, 3555 (int)mMainBuffer, 3556 (int)mAuxBuffer); 3557} 3558 3559status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3560 AudioBufferProvider::Buffer* buffer, int64_t pts) 3561{ 3562 audio_track_cblk_t* cblk = this->cblk(); 3563 uint32_t framesReady; 3564 uint32_t framesReq = buffer->frameCount; 3565 3566 // Check if last stepServer failed, try to step now 3567 if (mStepServerFailed) { 3568 if (!step()) goto getNextBuffer_exit; 3569 ALOGV("stepServer recovered"); 3570 mStepServerFailed = false; 3571 } 3572 3573 framesReady = cblk->framesReady(); 3574 3575 if (CC_LIKELY(framesReady)) { 3576 uint32_t s = cblk->server; 3577 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3578 3579 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3580 if (framesReq > framesReady) { 3581 framesReq = framesReady; 3582 } 3583 if (s + framesReq > bufferEnd) { 3584 framesReq = bufferEnd - s; 3585 } 3586 3587 buffer->raw = getBuffer(s, framesReq); 3588 if (buffer->raw == NULL) goto getNextBuffer_exit; 3589 3590 buffer->frameCount = framesReq; 3591 return NO_ERROR; 3592 } 3593 3594getNextBuffer_exit: 3595 buffer->raw = NULL; 3596 buffer->frameCount = 0; 3597 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3598 return NOT_ENOUGH_DATA; 3599} 3600 3601uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3602 return mCblk->framesReady(); 3603} 3604 3605bool AudioFlinger::PlaybackThread::Track::isReady() const { 3606 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3607 3608 if (framesReady() >= mCblk->frameCount || 3609 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3610 mFillingUpStatus = FS_FILLED; 3611 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3612 return true; 3613 } 3614 return false; 3615} 3616 3617status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3618{ 3619 status_t status = NO_ERROR; 3620 ALOGV("start(%d), calling pid %d session %d tid %d", 3621 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3622 sp<ThreadBase> thread = mThread.promote(); 3623 if (thread != 0) { 3624 Mutex::Autolock _l(thread->mLock); 3625 track_state state = mState; 3626 // here the track could be either new, or restarted 3627 // in both cases "unstop" the track 3628 if (mState == PAUSED) { 3629 mState = TrackBase::RESUMING; 3630 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3631 } else { 3632 mState = TrackBase::ACTIVE; 3633 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3634 } 3635 3636 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3637 thread->mLock.unlock(); 3638 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3639 thread->mLock.lock(); 3640 3641 // to track the speaker usage 3642 if (status == NO_ERROR) { 3643 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3644 } 3645 } 3646 if (status == NO_ERROR) { 3647 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3648 playbackThread->addTrack_l(this); 3649 } else { 3650 mState = state; 3651 } 3652 } else { 3653 status = BAD_VALUE; 3654 } 3655 return status; 3656} 3657 3658void AudioFlinger::PlaybackThread::Track::stop() 3659{ 3660 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3661 sp<ThreadBase> thread = mThread.promote(); 3662 if (thread != 0) { 3663 Mutex::Autolock _l(thread->mLock); 3664 track_state state = mState; 3665 if (mState > STOPPED) { 3666 mState = STOPPED; 3667 // If the track is not active (PAUSED and buffers full), flush buffers 3668 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3669 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3670 reset(); 3671 } 3672 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3673 } 3674 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3675 thread->mLock.unlock(); 3676 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3677 thread->mLock.lock(); 3678 3679 // to track the speaker usage 3680 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3681 } 3682 } 3683} 3684 3685void AudioFlinger::PlaybackThread::Track::pause() 3686{ 3687 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3688 sp<ThreadBase> thread = mThread.promote(); 3689 if (thread != 0) { 3690 Mutex::Autolock _l(thread->mLock); 3691 if (mState == ACTIVE || mState == RESUMING) { 3692 mState = PAUSING; 3693 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3694 if (!isOutputTrack()) { 3695 thread->mLock.unlock(); 3696 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3697 thread->mLock.lock(); 3698 3699 // to track the speaker usage 3700 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3701 } 3702 } 3703 } 3704} 3705 3706void AudioFlinger::PlaybackThread::Track::flush() 3707{ 3708 ALOGV("flush(%d)", mName); 3709 sp<ThreadBase> thread = mThread.promote(); 3710 if (thread != 0) { 3711 Mutex::Autolock _l(thread->mLock); 3712 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3713 return; 3714 } 3715 // No point remaining in PAUSED state after a flush => go to 3716 // STOPPED state 3717 mState = STOPPED; 3718 3719 // do not reset the track if it is still in the process of being stopped or paused. 3720 // this will be done by prepareTracks_l() when the track is stopped. 3721 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3722 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3723 reset(); 3724 } 3725 } 3726} 3727 3728void AudioFlinger::PlaybackThread::Track::reset() 3729{ 3730 // Do not reset twice to avoid discarding data written just after a flush and before 3731 // the audioflinger thread detects the track is stopped. 3732 if (!mResetDone) { 3733 TrackBase::reset(); 3734 // Force underrun condition to avoid false underrun callback until first data is 3735 // written to buffer 3736 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3737 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3738 mFillingUpStatus = FS_FILLING; 3739 mResetDone = true; 3740 } 3741} 3742 3743void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3744{ 3745 mMute = muted; 3746} 3747 3748status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3749{ 3750 status_t status = DEAD_OBJECT; 3751 sp<ThreadBase> thread = mThread.promote(); 3752 if (thread != 0) { 3753 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3754 status = playbackThread->attachAuxEffect(this, EffectId); 3755 } 3756 return status; 3757} 3758 3759void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3760{ 3761 mAuxEffectId = EffectId; 3762 mAuxBuffer = buffer; 3763} 3764 3765// timed audio tracks 3766 3767sp<AudioFlinger::PlaybackThread::TimedTrack> 3768AudioFlinger::PlaybackThread::TimedTrack::create( 3769 PlaybackThread *thread, 3770 const sp<Client>& client, 3771 audio_stream_type_t streamType, 3772 uint32_t sampleRate, 3773 audio_format_t format, 3774 uint32_t channelMask, 3775 int frameCount, 3776 const sp<IMemory>& sharedBuffer, 3777 int sessionId) { 3778 if (!client->reserveTimedTrack()) 3779 return NULL; 3780 3781 sp<TimedTrack> track = new TimedTrack( 3782 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3783 sharedBuffer, sessionId); 3784 3785 if (track == NULL) { 3786 client->releaseTimedTrack(); 3787 return NULL; 3788 } 3789 3790 return track; 3791} 3792 3793AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3794 PlaybackThread *thread, 3795 const sp<Client>& client, 3796 audio_stream_type_t streamType, 3797 uint32_t sampleRate, 3798 audio_format_t format, 3799 uint32_t channelMask, 3800 int frameCount, 3801 const sp<IMemory>& sharedBuffer, 3802 int sessionId) 3803 : Track(thread, client, streamType, sampleRate, format, channelMask, 3804 frameCount, sharedBuffer, sessionId), 3805 mTimedSilenceBuffer(NULL), 3806 mTimedSilenceBufferSize(0), 3807 mTimedAudioOutputOnTime(false), 3808 mMediaTimeTransformValid(false) 3809{ 3810 LocalClock lc; 3811 mLocalTimeFreq = lc.getLocalFreq(); 3812 3813 mLocalTimeToSampleTransform.a_zero = 0; 3814 mLocalTimeToSampleTransform.b_zero = 0; 3815 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3816 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3817 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3818 &mLocalTimeToSampleTransform.a_to_b_denom); 3819} 3820 3821AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3822 mClient->releaseTimedTrack(); 3823 delete [] mTimedSilenceBuffer; 3824} 3825 3826status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3827 size_t size, sp<IMemory>* buffer) { 3828 3829 Mutex::Autolock _l(mTimedBufferQueueLock); 3830 3831 trimTimedBufferQueue_l(); 3832 3833 // lazily initialize the shared memory heap for timed buffers 3834 if (mTimedMemoryDealer == NULL) { 3835 const int kTimedBufferHeapSize = 512 << 10; 3836 3837 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3838 "AudioFlingerTimed"); 3839 if (mTimedMemoryDealer == NULL) 3840 return NO_MEMORY; 3841 } 3842 3843 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3844 if (newBuffer == NULL) { 3845 newBuffer = mTimedMemoryDealer->allocate(size); 3846 if (newBuffer == NULL) 3847 return NO_MEMORY; 3848 } 3849 3850 *buffer = newBuffer; 3851 return NO_ERROR; 3852} 3853 3854// caller must hold mTimedBufferQueueLock 3855void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3856 int64_t mediaTimeNow; 3857 { 3858 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3859 if (!mMediaTimeTransformValid) 3860 return; 3861 3862 int64_t targetTimeNow; 3863 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3864 ? mCCHelper.getCommonTime(&targetTimeNow) 3865 : mCCHelper.getLocalTime(&targetTimeNow); 3866 3867 if (OK != res) 3868 return; 3869 3870 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3871 &mediaTimeNow)) { 3872 return; 3873 } 3874 } 3875 3876 size_t trimIndex; 3877 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3878 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3879 break; 3880 } 3881 3882 if (trimIndex) { 3883 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3884 } 3885} 3886 3887status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3888 const sp<IMemory>& buffer, int64_t pts) { 3889 3890 { 3891 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3892 if (!mMediaTimeTransformValid) 3893 return INVALID_OPERATION; 3894 } 3895 3896 Mutex::Autolock _l(mTimedBufferQueueLock); 3897 3898 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3899 3900 return NO_ERROR; 3901} 3902 3903status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3904 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3905 3906 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3907 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3908 target); 3909 3910 if (!(target == TimedAudioTrack::LOCAL_TIME || 3911 target == TimedAudioTrack::COMMON_TIME)) { 3912 return BAD_VALUE; 3913 } 3914 3915 Mutex::Autolock lock(mMediaTimeTransformLock); 3916 mMediaTimeTransform = xform; 3917 mMediaTimeTransformTarget = target; 3918 mMediaTimeTransformValid = true; 3919 3920 return NO_ERROR; 3921} 3922 3923#define min(a, b) ((a) < (b) ? (a) : (b)) 3924 3925// implementation of getNextBuffer for tracks whose buffers have timestamps 3926status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3927 AudioBufferProvider::Buffer* buffer, int64_t pts) 3928{ 3929 if (pts == AudioBufferProvider::kInvalidPTS) { 3930 buffer->raw = 0; 3931 buffer->frameCount = 0; 3932 return INVALID_OPERATION; 3933 } 3934 3935 Mutex::Autolock _l(mTimedBufferQueueLock); 3936 3937 while (true) { 3938 3939 // if we have no timed buffers, then fail 3940 if (mTimedBufferQueue.isEmpty()) { 3941 buffer->raw = 0; 3942 buffer->frameCount = 0; 3943 return NOT_ENOUGH_DATA; 3944 } 3945 3946 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3947 3948 // calculate the PTS of the head of the timed buffer queue expressed in 3949 // local time 3950 int64_t headLocalPTS; 3951 { 3952 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3953 3954 assert(mMediaTimeTransformValid); 3955 3956 if (mMediaTimeTransform.a_to_b_denom == 0) { 3957 // the transform represents a pause, so yield silence 3958 timedYieldSilence(buffer->frameCount, buffer); 3959 return NO_ERROR; 3960 } 3961 3962 int64_t transformedPTS; 3963 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3964 &transformedPTS)) { 3965 // the transform failed. this shouldn't happen, but if it does 3966 // then just drop this buffer 3967 ALOGW("timedGetNextBuffer transform failed"); 3968 buffer->raw = 0; 3969 buffer->frameCount = 0; 3970 mTimedBufferQueue.removeAt(0); 3971 return NO_ERROR; 3972 } 3973 3974 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3975 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3976 &headLocalPTS)) { 3977 buffer->raw = 0; 3978 buffer->frameCount = 0; 3979 return INVALID_OPERATION; 3980 } 3981 } else { 3982 headLocalPTS = transformedPTS; 3983 } 3984 } 3985 3986 // adjust the head buffer's PTS to reflect the portion of the head buffer 3987 // that has already been consumed 3988 int64_t effectivePTS = headLocalPTS + 3989 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3990 3991 // Calculate the delta in samples between the head of the input buffer 3992 // queue and the start of the next output buffer that will be written. 3993 // If the transformation fails because of over or underflow, it means 3994 // that the sample's position in the output stream is so far out of 3995 // whack that it should just be dropped. 3996 int64_t sampleDelta; 3997 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3998 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3999 mTimedBufferQueue.removeAt(0); 4000 continue; 4001 } 4002 if (!mLocalTimeToSampleTransform.doForwardTransform( 4003 (effectivePTS - pts) << 32, &sampleDelta)) { 4004 ALOGV("*** too late during sample rate transform: dropped buffer"); 4005 mTimedBufferQueue.removeAt(0); 4006 continue; 4007 } 4008 4009 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 4010 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 4011 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 4012 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 4013 4014 // if the delta between the ideal placement for the next input sample and 4015 // the current output position is within this threshold, then we will 4016 // concatenate the next input samples to the previous output 4017 const int64_t kSampleContinuityThreshold = 4018 (static_cast<int64_t>(sampleRate()) << 32) / 10; 4019 4020 // if this is the first buffer of audio that we're emitting from this track 4021 // then it should be almost exactly on time. 4022 const int64_t kSampleStartupThreshold = 1LL << 32; 4023 4024 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 4025 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 4026 // the next input is close enough to being on time, so concatenate it 4027 // with the last output 4028 timedYieldSamples(buffer); 4029 4030 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4031 return NO_ERROR; 4032 } else if (sampleDelta > 0) { 4033 // the gap between the current output position and the proper start of 4034 // the next input sample is too big, so fill it with silence 4035 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 4036 4037 timedYieldSilence(framesUntilNextInput, buffer); 4038 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 4039 return NO_ERROR; 4040 } else { 4041 // the next input sample is late 4042 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 4043 size_t onTimeSamplePosition = 4044 head.position() + lateFrames * mCblk->frameSize; 4045 4046 if (onTimeSamplePosition > head.buffer()->size()) { 4047 // all the remaining samples in the head are too late, so 4048 // drop it and move on 4049 ALOGV("*** too late: dropped buffer"); 4050 mTimedBufferQueue.removeAt(0); 4051 continue; 4052 } else { 4053 // skip over the late samples 4054 head.setPosition(onTimeSamplePosition); 4055 4056 // yield the available samples 4057 timedYieldSamples(buffer); 4058 4059 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 4060 return NO_ERROR; 4061 } 4062 } 4063 } 4064} 4065 4066// Yield samples from the timed buffer queue head up to the given output 4067// buffer's capacity. 4068// 4069// Caller must hold mTimedBufferQueueLock 4070void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4071 AudioBufferProvider::Buffer* buffer) { 4072 4073 const TimedBuffer& head = mTimedBufferQueue[0]; 4074 4075 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4076 head.position()); 4077 4078 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4079 mCblk->frameSize); 4080 size_t framesRequested = buffer->frameCount; 4081 buffer->frameCount = min(framesLeftInHead, framesRequested); 4082 4083 mTimedAudioOutputOnTime = true; 4084} 4085 4086// Yield samples of silence up to the given output buffer's capacity 4087// 4088// Caller must hold mTimedBufferQueueLock 4089void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4090 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4091 4092 // lazily allocate a buffer filled with silence 4093 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4094 delete [] mTimedSilenceBuffer; 4095 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4096 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4097 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4098 } 4099 4100 buffer->raw = mTimedSilenceBuffer; 4101 size_t framesRequested = buffer->frameCount; 4102 buffer->frameCount = min(numFrames, framesRequested); 4103 4104 mTimedAudioOutputOnTime = false; 4105} 4106 4107void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4108 AudioBufferProvider::Buffer* buffer) { 4109 4110 Mutex::Autolock _l(mTimedBufferQueueLock); 4111 4112 // If the buffer which was just released is part of the buffer at the head 4113 // of the queue, be sure to update the amt of the buffer which has been 4114 // consumed. If the buffer being returned is not part of the head of the 4115 // queue, its either because the buffer is part of the silence buffer, or 4116 // because the head of the timed queue was trimmed after the mixer called 4117 // getNextBuffer but before the mixer called releaseBuffer. 4118 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4119 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4120 4121 void* start = head.buffer()->pointer(); 4122 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4123 4124 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4125 head.setPosition(head.position() + 4126 (buffer->frameCount * mCblk->frameSize)); 4127 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4128 mTimedBufferQueue.removeAt(0); 4129 } 4130 } 4131 } 4132 4133 buffer->raw = 0; 4134 buffer->frameCount = 0; 4135} 4136 4137uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4138 Mutex::Autolock _l(mTimedBufferQueueLock); 4139 4140 uint32_t frames = 0; 4141 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4142 const TimedBuffer& tb = mTimedBufferQueue[i]; 4143 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4144 } 4145 4146 return frames; 4147} 4148 4149AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4150 : mPTS(0), mPosition(0) {} 4151 4152AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4153 const sp<IMemory>& buffer, int64_t pts) 4154 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4155 4156// ---------------------------------------------------------------------------- 4157 4158// RecordTrack constructor must be called with AudioFlinger::mLock held 4159AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4160 RecordThread *thread, 4161 const sp<Client>& client, 4162 uint32_t sampleRate, 4163 audio_format_t format, 4164 uint32_t channelMask, 4165 int frameCount, 4166 int sessionId) 4167 : TrackBase(thread, client, sampleRate, format, 4168 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4169 mOverflow(false) 4170{ 4171 if (mCblk != NULL) { 4172 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4173 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4174 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4175 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4176 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4177 } else { 4178 mCblk->frameSize = sizeof(int8_t); 4179 } 4180 } 4181} 4182 4183AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4184{ 4185 sp<ThreadBase> thread = mThread.promote(); 4186 if (thread != 0) { 4187 AudioSystem::releaseInput(thread->id()); 4188 } 4189} 4190 4191status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4192{ 4193 audio_track_cblk_t* cblk = this->cblk(); 4194 uint32_t framesAvail; 4195 uint32_t framesReq = buffer->frameCount; 4196 4197 // Check if last stepServer failed, try to step now 4198 if (mStepServerFailed) { 4199 if (!step()) goto getNextBuffer_exit; 4200 ALOGV("stepServer recovered"); 4201 mStepServerFailed = false; 4202 } 4203 4204 framesAvail = cblk->framesAvailable_l(); 4205 4206 if (CC_LIKELY(framesAvail)) { 4207 uint32_t s = cblk->server; 4208 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4209 4210 if (framesReq > framesAvail) { 4211 framesReq = framesAvail; 4212 } 4213 if (s + framesReq > bufferEnd) { 4214 framesReq = bufferEnd - s; 4215 } 4216 4217 buffer->raw = getBuffer(s, framesReq); 4218 if (buffer->raw == NULL) goto getNextBuffer_exit; 4219 4220 buffer->frameCount = framesReq; 4221 return NO_ERROR; 4222 } 4223 4224getNextBuffer_exit: 4225 buffer->raw = NULL; 4226 buffer->frameCount = 0; 4227 return NOT_ENOUGH_DATA; 4228} 4229 4230status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4231{ 4232 sp<ThreadBase> thread = mThread.promote(); 4233 if (thread != 0) { 4234 RecordThread *recordThread = (RecordThread *)thread.get(); 4235 return recordThread->start(this, tid); 4236 } else { 4237 return BAD_VALUE; 4238 } 4239} 4240 4241void AudioFlinger::RecordThread::RecordTrack::stop() 4242{ 4243 sp<ThreadBase> thread = mThread.promote(); 4244 if (thread != 0) { 4245 RecordThread *recordThread = (RecordThread *)thread.get(); 4246 recordThread->stop(this); 4247 TrackBase::reset(); 4248 // Force overerrun condition to avoid false overrun callback until first data is 4249 // read from buffer 4250 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4251 } 4252} 4253 4254void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4255{ 4256 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4257 (mClient == 0) ? getpid_cached : mClient->pid(), 4258 mFormat, 4259 mChannelMask, 4260 mSessionId, 4261 mFrameCount, 4262 mState, 4263 mCblk->sampleRate, 4264 mCblk->server, 4265 mCblk->user); 4266} 4267 4268 4269// ---------------------------------------------------------------------------- 4270 4271AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4272 PlaybackThread *playbackThread, 4273 DuplicatingThread *sourceThread, 4274 uint32_t sampleRate, 4275 audio_format_t format, 4276 uint32_t channelMask, 4277 int frameCount) 4278 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4279 mActive(false), mSourceThread(sourceThread) 4280{ 4281 4282 if (mCblk != NULL) { 4283 mCblk->flags |= CBLK_DIRECTION_OUT; 4284 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4285 mOutBuffer.frameCount = 0; 4286 playbackThread->mTracks.add(this); 4287 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4288 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4289 mCblk, mBuffer, mCblk->buffers, 4290 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4291 } else { 4292 ALOGW("Error creating output track on thread %p", playbackThread); 4293 } 4294} 4295 4296AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4297{ 4298 clearBufferQueue(); 4299} 4300 4301status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4302{ 4303 status_t status = Track::start(tid); 4304 if (status != NO_ERROR) { 4305 return status; 4306 } 4307 4308 mActive = true; 4309 mRetryCount = 127; 4310 return status; 4311} 4312 4313void AudioFlinger::PlaybackThread::OutputTrack::stop() 4314{ 4315 Track::stop(); 4316 clearBufferQueue(); 4317 mOutBuffer.frameCount = 0; 4318 mActive = false; 4319} 4320 4321bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4322{ 4323 Buffer *pInBuffer; 4324 Buffer inBuffer; 4325 uint32_t channelCount = mChannelCount; 4326 bool outputBufferFull = false; 4327 inBuffer.frameCount = frames; 4328 inBuffer.i16 = data; 4329 4330 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4331 4332 if (!mActive && frames != 0) { 4333 start(0); 4334 sp<ThreadBase> thread = mThread.promote(); 4335 if (thread != 0) { 4336 MixerThread *mixerThread = (MixerThread *)thread.get(); 4337 if (mCblk->frameCount > frames){ 4338 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4339 uint32_t startFrames = (mCblk->frameCount - frames); 4340 pInBuffer = new Buffer; 4341 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4342 pInBuffer->frameCount = startFrames; 4343 pInBuffer->i16 = pInBuffer->mBuffer; 4344 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4345 mBufferQueue.add(pInBuffer); 4346 } else { 4347 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4348 } 4349 } 4350 } 4351 } 4352 4353 while (waitTimeLeftMs) { 4354 // First write pending buffers, then new data 4355 if (mBufferQueue.size()) { 4356 pInBuffer = mBufferQueue.itemAt(0); 4357 } else { 4358 pInBuffer = &inBuffer; 4359 } 4360 4361 if (pInBuffer->frameCount == 0) { 4362 break; 4363 } 4364 4365 if (mOutBuffer.frameCount == 0) { 4366 mOutBuffer.frameCount = pInBuffer->frameCount; 4367 nsecs_t startTime = systemTime(); 4368 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4369 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4370 outputBufferFull = true; 4371 break; 4372 } 4373 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4374 if (waitTimeLeftMs >= waitTimeMs) { 4375 waitTimeLeftMs -= waitTimeMs; 4376 } else { 4377 waitTimeLeftMs = 0; 4378 } 4379 } 4380 4381 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4382 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4383 mCblk->stepUser(outFrames); 4384 pInBuffer->frameCount -= outFrames; 4385 pInBuffer->i16 += outFrames * channelCount; 4386 mOutBuffer.frameCount -= outFrames; 4387 mOutBuffer.i16 += outFrames * channelCount; 4388 4389 if (pInBuffer->frameCount == 0) { 4390 if (mBufferQueue.size()) { 4391 mBufferQueue.removeAt(0); 4392 delete [] pInBuffer->mBuffer; 4393 delete pInBuffer; 4394 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4395 } else { 4396 break; 4397 } 4398 } 4399 } 4400 4401 // If we could not write all frames, allocate a buffer and queue it for next time. 4402 if (inBuffer.frameCount) { 4403 sp<ThreadBase> thread = mThread.promote(); 4404 if (thread != 0 && !thread->standby()) { 4405 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4406 pInBuffer = new Buffer; 4407 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4408 pInBuffer->frameCount = inBuffer.frameCount; 4409 pInBuffer->i16 = pInBuffer->mBuffer; 4410 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4411 mBufferQueue.add(pInBuffer); 4412 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4413 } else { 4414 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4415 } 4416 } 4417 } 4418 4419 // Calling write() with a 0 length buffer, means that no more data will be written: 4420 // If no more buffers are pending, fill output track buffer to make sure it is started 4421 // by output mixer. 4422 if (frames == 0 && mBufferQueue.size() == 0) { 4423 if (mCblk->user < mCblk->frameCount) { 4424 frames = mCblk->frameCount - mCblk->user; 4425 pInBuffer = new Buffer; 4426 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4427 pInBuffer->frameCount = frames; 4428 pInBuffer->i16 = pInBuffer->mBuffer; 4429 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4430 mBufferQueue.add(pInBuffer); 4431 } else if (mActive) { 4432 stop(); 4433 } 4434 } 4435 4436 return outputBufferFull; 4437} 4438 4439status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4440{ 4441 int active; 4442 status_t result; 4443 audio_track_cblk_t* cblk = mCblk; 4444 uint32_t framesReq = buffer->frameCount; 4445 4446// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4447 buffer->frameCount = 0; 4448 4449 uint32_t framesAvail = cblk->framesAvailable(); 4450 4451 4452 if (framesAvail == 0) { 4453 Mutex::Autolock _l(cblk->lock); 4454 goto start_loop_here; 4455 while (framesAvail == 0) { 4456 active = mActive; 4457 if (CC_UNLIKELY(!active)) { 4458 ALOGV("Not active and NO_MORE_BUFFERS"); 4459 return NO_MORE_BUFFERS; 4460 } 4461 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4462 if (result != NO_ERROR) { 4463 return NO_MORE_BUFFERS; 4464 } 4465 // read the server count again 4466 start_loop_here: 4467 framesAvail = cblk->framesAvailable_l(); 4468 } 4469 } 4470 4471// if (framesAvail < framesReq) { 4472// return NO_MORE_BUFFERS; 4473// } 4474 4475 if (framesReq > framesAvail) { 4476 framesReq = framesAvail; 4477 } 4478 4479 uint32_t u = cblk->user; 4480 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4481 4482 if (u + framesReq > bufferEnd) { 4483 framesReq = bufferEnd - u; 4484 } 4485 4486 buffer->frameCount = framesReq; 4487 buffer->raw = (void *)cblk->buffer(u); 4488 return NO_ERROR; 4489} 4490 4491 4492void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4493{ 4494 size_t size = mBufferQueue.size(); 4495 4496 for (size_t i = 0; i < size; i++) { 4497 Buffer *pBuffer = mBufferQueue.itemAt(i); 4498 delete [] pBuffer->mBuffer; 4499 delete pBuffer; 4500 } 4501 mBufferQueue.clear(); 4502} 4503 4504// ---------------------------------------------------------------------------- 4505 4506AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4507 : RefBase(), 4508 mAudioFlinger(audioFlinger), 4509 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4510 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4511 mPid(pid), 4512 mTimedTrackCount(0) 4513{ 4514 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4515} 4516 4517// Client destructor must be called with AudioFlinger::mLock held 4518AudioFlinger::Client::~Client() 4519{ 4520 mAudioFlinger->removeClient_l(mPid); 4521} 4522 4523sp<MemoryDealer> AudioFlinger::Client::heap() const 4524{ 4525 return mMemoryDealer; 4526} 4527 4528// Reserve one of the limited slots for a timed audio track associated 4529// with this client 4530bool AudioFlinger::Client::reserveTimedTrack() 4531{ 4532 const int kMaxTimedTracksPerClient = 4; 4533 4534 Mutex::Autolock _l(mTimedTrackLock); 4535 4536 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4537 ALOGW("can not create timed track - pid %d has exceeded the limit", 4538 mPid); 4539 return false; 4540 } 4541 4542 mTimedTrackCount++; 4543 return true; 4544} 4545 4546// Release a slot for a timed audio track 4547void AudioFlinger::Client::releaseTimedTrack() 4548{ 4549 Mutex::Autolock _l(mTimedTrackLock); 4550 mTimedTrackCount--; 4551} 4552 4553// ---------------------------------------------------------------------------- 4554 4555AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4556 const sp<IAudioFlingerClient>& client, 4557 pid_t pid) 4558 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4559{ 4560} 4561 4562AudioFlinger::NotificationClient::~NotificationClient() 4563{ 4564} 4565 4566void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4567{ 4568 sp<NotificationClient> keep(this); 4569 mAudioFlinger->removeNotificationClient(mPid); 4570} 4571 4572// ---------------------------------------------------------------------------- 4573 4574AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4575 : BnAudioTrack(), 4576 mTrack(track) 4577{ 4578} 4579 4580AudioFlinger::TrackHandle::~TrackHandle() { 4581 // just stop the track on deletion, associated resources 4582 // will be freed from the main thread once all pending buffers have 4583 // been played. Unless it's not in the active track list, in which 4584 // case we free everything now... 4585 mTrack->destroy(); 4586} 4587 4588sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4589 return mTrack->getCblk(); 4590} 4591 4592status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4593 return mTrack->start(tid); 4594} 4595 4596void AudioFlinger::TrackHandle::stop() { 4597 mTrack->stop(); 4598} 4599 4600void AudioFlinger::TrackHandle::flush() { 4601 mTrack->flush(); 4602} 4603 4604void AudioFlinger::TrackHandle::mute(bool e) { 4605 mTrack->mute(e); 4606} 4607 4608void AudioFlinger::TrackHandle::pause() { 4609 mTrack->pause(); 4610} 4611 4612status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4613{ 4614 return mTrack->attachAuxEffect(EffectId); 4615} 4616 4617status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4618 sp<IMemory>* buffer) { 4619 if (!mTrack->isTimedTrack()) 4620 return INVALID_OPERATION; 4621 4622 PlaybackThread::TimedTrack* tt = 4623 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4624 return tt->allocateTimedBuffer(size, buffer); 4625} 4626 4627status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4628 int64_t pts) { 4629 if (!mTrack->isTimedTrack()) 4630 return INVALID_OPERATION; 4631 4632 PlaybackThread::TimedTrack* tt = 4633 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4634 return tt->queueTimedBuffer(buffer, pts); 4635} 4636 4637status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4638 const LinearTransform& xform, int target) { 4639 4640 if (!mTrack->isTimedTrack()) 4641 return INVALID_OPERATION; 4642 4643 PlaybackThread::TimedTrack* tt = 4644 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4645 return tt->setMediaTimeTransform( 4646 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4647} 4648 4649status_t AudioFlinger::TrackHandle::onTransact( 4650 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4651{ 4652 return BnAudioTrack::onTransact(code, data, reply, flags); 4653} 4654 4655// ---------------------------------------------------------------------------- 4656 4657sp<IAudioRecord> AudioFlinger::openRecord( 4658 pid_t pid, 4659 audio_io_handle_t input, 4660 uint32_t sampleRate, 4661 audio_format_t format, 4662 uint32_t channelMask, 4663 int frameCount, 4664 // FIXME dead, remove from IAudioFlinger 4665 uint32_t flags, 4666 int *sessionId, 4667 status_t *status) 4668{ 4669 sp<RecordThread::RecordTrack> recordTrack; 4670 sp<RecordHandle> recordHandle; 4671 sp<Client> client; 4672 status_t lStatus; 4673 RecordThread *thread; 4674 size_t inFrameCount; 4675 int lSessionId; 4676 4677 // check calling permissions 4678 if (!recordingAllowed()) { 4679 lStatus = PERMISSION_DENIED; 4680 goto Exit; 4681 } 4682 4683 // add client to list 4684 { // scope for mLock 4685 Mutex::Autolock _l(mLock); 4686 thread = checkRecordThread_l(input); 4687 if (thread == NULL) { 4688 lStatus = BAD_VALUE; 4689 goto Exit; 4690 } 4691 4692 client = registerPid_l(pid); 4693 4694 // If no audio session id is provided, create one here 4695 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4696 lSessionId = *sessionId; 4697 } else { 4698 lSessionId = nextUniqueId(); 4699 if (sessionId != NULL) { 4700 *sessionId = lSessionId; 4701 } 4702 } 4703 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4704 recordTrack = thread->createRecordTrack_l(client, 4705 sampleRate, 4706 format, 4707 channelMask, 4708 frameCount, 4709 lSessionId, 4710 &lStatus); 4711 } 4712 if (lStatus != NO_ERROR) { 4713 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4714 // destructor is called by the TrackBase destructor with mLock held 4715 client.clear(); 4716 recordTrack.clear(); 4717 goto Exit; 4718 } 4719 4720 // return to handle to client 4721 recordHandle = new RecordHandle(recordTrack); 4722 lStatus = NO_ERROR; 4723 4724Exit: 4725 if (status) { 4726 *status = lStatus; 4727 } 4728 return recordHandle; 4729} 4730 4731// ---------------------------------------------------------------------------- 4732 4733AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4734 : BnAudioRecord(), 4735 mRecordTrack(recordTrack) 4736{ 4737} 4738 4739AudioFlinger::RecordHandle::~RecordHandle() { 4740 stop(); 4741} 4742 4743sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4744 return mRecordTrack->getCblk(); 4745} 4746 4747status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4748 ALOGV("RecordHandle::start()"); 4749 return mRecordTrack->start(tid); 4750} 4751 4752void AudioFlinger::RecordHandle::stop() { 4753 ALOGV("RecordHandle::stop()"); 4754 mRecordTrack->stop(); 4755} 4756 4757status_t AudioFlinger::RecordHandle::onTransact( 4758 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4759{ 4760 return BnAudioRecord::onTransact(code, data, reply, flags); 4761} 4762 4763// ---------------------------------------------------------------------------- 4764 4765AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4766 AudioStreamIn *input, 4767 uint32_t sampleRate, 4768 uint32_t channels, 4769 audio_io_handle_t id, 4770 uint32_t device) : 4771 ThreadBase(audioFlinger, id, device, RECORD), 4772 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4773 // mRsmpInIndex and mInputBytes set by readInputParameters() 4774 mReqChannelCount(popcount(channels)), 4775 mReqSampleRate(sampleRate) 4776 // mBytesRead is only meaningful while active, and so is cleared in start() 4777 // (but might be better to also clear here for dump?) 4778{ 4779 snprintf(mName, kNameLength, "AudioIn_%d", id); 4780 4781 readInputParameters(); 4782} 4783 4784 4785AudioFlinger::RecordThread::~RecordThread() 4786{ 4787 delete[] mRsmpInBuffer; 4788 delete mResampler; 4789 delete[] mRsmpOutBuffer; 4790} 4791 4792void AudioFlinger::RecordThread::onFirstRef() 4793{ 4794 run(mName, PRIORITY_URGENT_AUDIO); 4795} 4796 4797status_t AudioFlinger::RecordThread::readyToRun() 4798{ 4799 status_t status = initCheck(); 4800 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4801 return status; 4802} 4803 4804bool AudioFlinger::RecordThread::threadLoop() 4805{ 4806 AudioBufferProvider::Buffer buffer; 4807 sp<RecordTrack> activeTrack; 4808 Vector< sp<EffectChain> > effectChains; 4809 4810 nsecs_t lastWarning = 0; 4811 4812 acquireWakeLock(); 4813 4814 // start recording 4815 while (!exitPending()) { 4816 4817 processConfigEvents(); 4818 4819 { // scope for mLock 4820 Mutex::Autolock _l(mLock); 4821 checkForNewParameters_l(); 4822 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4823 if (!mStandby) { 4824 mInput->stream->common.standby(&mInput->stream->common); 4825 mStandby = true; 4826 } 4827 4828 if (exitPending()) break; 4829 4830 releaseWakeLock_l(); 4831 ALOGV("RecordThread: loop stopping"); 4832 // go to sleep 4833 mWaitWorkCV.wait(mLock); 4834 ALOGV("RecordThread: loop starting"); 4835 acquireWakeLock_l(); 4836 continue; 4837 } 4838 if (mActiveTrack != 0) { 4839 if (mActiveTrack->mState == TrackBase::PAUSING) { 4840 if (!mStandby) { 4841 mInput->stream->common.standby(&mInput->stream->common); 4842 mStandby = true; 4843 } 4844 mActiveTrack.clear(); 4845 mStartStopCond.broadcast(); 4846 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4847 if (mReqChannelCount != mActiveTrack->channelCount()) { 4848 mActiveTrack.clear(); 4849 mStartStopCond.broadcast(); 4850 } else if (mBytesRead != 0) { 4851 // record start succeeds only if first read from audio input 4852 // succeeds 4853 if (mBytesRead > 0) { 4854 mActiveTrack->mState = TrackBase::ACTIVE; 4855 } else { 4856 mActiveTrack.clear(); 4857 } 4858 mStartStopCond.broadcast(); 4859 } 4860 mStandby = false; 4861 } 4862 } 4863 lockEffectChains_l(effectChains); 4864 } 4865 4866 if (mActiveTrack != 0) { 4867 if (mActiveTrack->mState != TrackBase::ACTIVE && 4868 mActiveTrack->mState != TrackBase::RESUMING) { 4869 unlockEffectChains(effectChains); 4870 usleep(kRecordThreadSleepUs); 4871 continue; 4872 } 4873 for (size_t i = 0; i < effectChains.size(); i ++) { 4874 effectChains[i]->process_l(); 4875 } 4876 4877 buffer.frameCount = mFrameCount; 4878 if (CC_LIKELY(mActiveTrack->getNextBuffer( 4879 &buffer, AudioBufferProvider::kInvalidPTS) == NO_ERROR)) { 4880 size_t framesOut = buffer.frameCount; 4881 if (mResampler == NULL) { 4882 // no resampling 4883 while (framesOut) { 4884 size_t framesIn = mFrameCount - mRsmpInIndex; 4885 if (framesIn) { 4886 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4887 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4888 if (framesIn > framesOut) 4889 framesIn = framesOut; 4890 mRsmpInIndex += framesIn; 4891 framesOut -= framesIn; 4892 if ((int)mChannelCount == mReqChannelCount || 4893 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4894 memcpy(dst, src, framesIn * mFrameSize); 4895 } else { 4896 int16_t *src16 = (int16_t *)src; 4897 int16_t *dst16 = (int16_t *)dst; 4898 if (mChannelCount == 1) { 4899 while (framesIn--) { 4900 *dst16++ = *src16; 4901 *dst16++ = *src16++; 4902 } 4903 } else { 4904 while (framesIn--) { 4905 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4906 src16 += 2; 4907 } 4908 } 4909 } 4910 } 4911 if (framesOut && mFrameCount == mRsmpInIndex) { 4912 if (framesOut == mFrameCount && 4913 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4914 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4915 framesOut = 0; 4916 } else { 4917 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4918 mRsmpInIndex = 0; 4919 } 4920 if (mBytesRead < 0) { 4921 ALOGE("Error reading audio input"); 4922 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4923 // Force input into standby so that it tries to 4924 // recover at next read attempt 4925 mInput->stream->common.standby(&mInput->stream->common); 4926 usleep(kRecordThreadSleepUs); 4927 } 4928 mRsmpInIndex = mFrameCount; 4929 framesOut = 0; 4930 buffer.frameCount = 0; 4931 } 4932 } 4933 } 4934 } else { 4935 // resampling 4936 4937 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4938 // alter output frame count as if we were expecting stereo samples 4939 if (mChannelCount == 1 && mReqChannelCount == 1) { 4940 framesOut >>= 1; 4941 } 4942 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4943 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4944 // are 32 bit aligned which should be always true. 4945 if (mChannelCount == 2 && mReqChannelCount == 1) { 4946 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4947 // the resampler always outputs stereo samples: do post stereo to mono conversion 4948 int16_t *src = (int16_t *)mRsmpOutBuffer; 4949 int16_t *dst = buffer.i16; 4950 while (framesOut--) { 4951 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4952 src += 2; 4953 } 4954 } else { 4955 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4956 } 4957 4958 } 4959 mActiveTrack->releaseBuffer(&buffer); 4960 mActiveTrack->overflow(); 4961 } 4962 // client isn't retrieving buffers fast enough 4963 else { 4964 if (!mActiveTrack->setOverflow()) { 4965 nsecs_t now = systemTime(); 4966 if ((now - lastWarning) > kWarningThrottleNs) { 4967 ALOGW("RecordThread: buffer overflow"); 4968 lastWarning = now; 4969 } 4970 } 4971 // Release the processor for a while before asking for a new buffer. 4972 // This will give the application more chance to read from the buffer and 4973 // clear the overflow. 4974 usleep(kRecordThreadSleepUs); 4975 } 4976 } 4977 // enable changes in effect chain 4978 unlockEffectChains(effectChains); 4979 effectChains.clear(); 4980 } 4981 4982 if (!mStandby) { 4983 mInput->stream->common.standby(&mInput->stream->common); 4984 } 4985 mActiveTrack.clear(); 4986 4987 mStartStopCond.broadcast(); 4988 4989 releaseWakeLock(); 4990 4991 ALOGV("RecordThread %p exiting", this); 4992 return false; 4993} 4994 4995 4996sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4997 const sp<AudioFlinger::Client>& client, 4998 uint32_t sampleRate, 4999 audio_format_t format, 5000 int channelMask, 5001 int frameCount, 5002 int sessionId, 5003 status_t *status) 5004{ 5005 sp<RecordTrack> track; 5006 status_t lStatus; 5007 5008 lStatus = initCheck(); 5009 if (lStatus != NO_ERROR) { 5010 ALOGE("Audio driver not initialized."); 5011 goto Exit; 5012 } 5013 5014 { // scope for mLock 5015 Mutex::Autolock _l(mLock); 5016 5017 track = new RecordTrack(this, client, sampleRate, 5018 format, channelMask, frameCount, sessionId); 5019 5020 if (track->getCblk() == 0) { 5021 lStatus = NO_MEMORY; 5022 goto Exit; 5023 } 5024 5025 mTrack = track.get(); 5026 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5027 bool suspend = audio_is_bluetooth_sco_device( 5028 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 5029 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 5030 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 5031 } 5032 lStatus = NO_ERROR; 5033 5034Exit: 5035 if (status) { 5036 *status = lStatus; 5037 } 5038 return track; 5039} 5040 5041status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 5042{ 5043 ALOGV("RecordThread::start tid=%d", tid); 5044 sp <ThreadBase> strongMe = this; 5045 status_t status = NO_ERROR; 5046 { 5047 AutoMutex lock(mLock); 5048 if (mActiveTrack != 0) { 5049 if (recordTrack != mActiveTrack.get()) { 5050 status = -EBUSY; 5051 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 5052 mActiveTrack->mState = TrackBase::ACTIVE; 5053 } 5054 return status; 5055 } 5056 5057 recordTrack->mState = TrackBase::IDLE; 5058 mActiveTrack = recordTrack; 5059 mLock.unlock(); 5060 status_t status = AudioSystem::startInput(mId); 5061 mLock.lock(); 5062 if (status != NO_ERROR) { 5063 mActiveTrack.clear(); 5064 return status; 5065 } 5066 mRsmpInIndex = mFrameCount; 5067 mBytesRead = 0; 5068 if (mResampler != NULL) { 5069 mResampler->reset(); 5070 } 5071 mActiveTrack->mState = TrackBase::RESUMING; 5072 // signal thread to start 5073 ALOGV("Signal record thread"); 5074 mWaitWorkCV.signal(); 5075 // do not wait for mStartStopCond if exiting 5076 if (exitPending()) { 5077 mActiveTrack.clear(); 5078 status = INVALID_OPERATION; 5079 goto startError; 5080 } 5081 mStartStopCond.wait(mLock); 5082 if (mActiveTrack == 0) { 5083 ALOGV("Record failed to start"); 5084 status = BAD_VALUE; 5085 goto startError; 5086 } 5087 ALOGV("Record started OK"); 5088 return status; 5089 } 5090startError: 5091 AudioSystem::stopInput(mId); 5092 return status; 5093} 5094 5095void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5096 ALOGV("RecordThread::stop"); 5097 sp <ThreadBase> strongMe = this; 5098 { 5099 AutoMutex lock(mLock); 5100 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5101 mActiveTrack->mState = TrackBase::PAUSING; 5102 // do not wait for mStartStopCond if exiting 5103 if (exitPending()) { 5104 return; 5105 } 5106 mStartStopCond.wait(mLock); 5107 // if we have been restarted, recordTrack == mActiveTrack.get() here 5108 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5109 mLock.unlock(); 5110 AudioSystem::stopInput(mId); 5111 mLock.lock(); 5112 ALOGV("Record stopped OK"); 5113 } 5114 } 5115 } 5116} 5117 5118status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5119{ 5120 const size_t SIZE = 256; 5121 char buffer[SIZE]; 5122 String8 result; 5123 5124 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5125 result.append(buffer); 5126 5127 if (mActiveTrack != 0) { 5128 result.append("Active Track:\n"); 5129 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5130 mActiveTrack->dump(buffer, SIZE); 5131 result.append(buffer); 5132 5133 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5134 result.append(buffer); 5135 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5136 result.append(buffer); 5137 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5138 result.append(buffer); 5139 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5140 result.append(buffer); 5141 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5142 result.append(buffer); 5143 5144 5145 } else { 5146 result.append("No record client\n"); 5147 } 5148 write(fd, result.string(), result.size()); 5149 5150 dumpBase(fd, args); 5151 dumpEffectChains(fd, args); 5152 5153 return NO_ERROR; 5154} 5155 5156status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5157{ 5158 size_t framesReq = buffer->frameCount; 5159 size_t framesReady = mFrameCount - mRsmpInIndex; 5160 int channelCount; 5161 5162 if (framesReady == 0) { 5163 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5164 if (mBytesRead < 0) { 5165 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5166 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5167 // Force input into standby so that it tries to 5168 // recover at next read attempt 5169 mInput->stream->common.standby(&mInput->stream->common); 5170 usleep(kRecordThreadSleepUs); 5171 } 5172 buffer->raw = NULL; 5173 buffer->frameCount = 0; 5174 return NOT_ENOUGH_DATA; 5175 } 5176 mRsmpInIndex = 0; 5177 framesReady = mFrameCount; 5178 } 5179 5180 if (framesReq > framesReady) { 5181 framesReq = framesReady; 5182 } 5183 5184 if (mChannelCount == 1 && mReqChannelCount == 2) { 5185 channelCount = 1; 5186 } else { 5187 channelCount = 2; 5188 } 5189 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5190 buffer->frameCount = framesReq; 5191 return NO_ERROR; 5192} 5193 5194void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5195{ 5196 mRsmpInIndex += buffer->frameCount; 5197 buffer->frameCount = 0; 5198} 5199 5200bool AudioFlinger::RecordThread::checkForNewParameters_l() 5201{ 5202 bool reconfig = false; 5203 5204 while (!mNewParameters.isEmpty()) { 5205 status_t status = NO_ERROR; 5206 String8 keyValuePair = mNewParameters[0]; 5207 AudioParameter param = AudioParameter(keyValuePair); 5208 int value; 5209 audio_format_t reqFormat = mFormat; 5210 int reqSamplingRate = mReqSampleRate; 5211 int reqChannelCount = mReqChannelCount; 5212 5213 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5214 reqSamplingRate = value; 5215 reconfig = true; 5216 } 5217 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5218 reqFormat = (audio_format_t) value; 5219 reconfig = true; 5220 } 5221 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5222 reqChannelCount = popcount(value); 5223 reconfig = true; 5224 } 5225 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5226 // do not accept frame count changes if tracks are open as the track buffer 5227 // size depends on frame count and correct behavior would not be guaranteed 5228 // if frame count is changed after track creation 5229 if (mActiveTrack != 0) { 5230 status = INVALID_OPERATION; 5231 } else { 5232 reconfig = true; 5233 } 5234 } 5235 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5236 // forward device change to effects that have requested to be 5237 // aware of attached audio device. 5238 for (size_t i = 0; i < mEffectChains.size(); i++) { 5239 mEffectChains[i]->setDevice_l(value); 5240 } 5241 // store input device and output device but do not forward output device to audio HAL. 5242 // Note that status is ignored by the caller for output device 5243 // (see AudioFlinger::setParameters() 5244 if (value & AUDIO_DEVICE_OUT_ALL) { 5245 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5246 status = BAD_VALUE; 5247 } else { 5248 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5249 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5250 if (mTrack != NULL) { 5251 bool suspend = audio_is_bluetooth_sco_device( 5252 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5253 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5254 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5255 } 5256 } 5257 mDevice |= (uint32_t)value; 5258 } 5259 if (status == NO_ERROR) { 5260 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5261 if (status == INVALID_OPERATION) { 5262 mInput->stream->common.standby(&mInput->stream->common); 5263 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5264 } 5265 if (reconfig) { 5266 if (status == BAD_VALUE && 5267 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5268 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5269 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5270 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5271 (reqChannelCount < 3)) { 5272 status = NO_ERROR; 5273 } 5274 if (status == NO_ERROR) { 5275 readInputParameters(); 5276 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5277 } 5278 } 5279 } 5280 5281 mNewParameters.removeAt(0); 5282 5283 mParamStatus = status; 5284 mParamCond.signal(); 5285 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5286 // already timed out waiting for the status and will never signal the condition. 5287 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5288 } 5289 return reconfig; 5290} 5291 5292String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5293{ 5294 char *s; 5295 String8 out_s8 = String8(); 5296 5297 Mutex::Autolock _l(mLock); 5298 if (initCheck() != NO_ERROR) { 5299 return out_s8; 5300 } 5301 5302 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5303 out_s8 = String8(s); 5304 free(s); 5305 return out_s8; 5306} 5307 5308void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5309 AudioSystem::OutputDescriptor desc; 5310 void *param2 = NULL; 5311 5312 switch (event) { 5313 case AudioSystem::INPUT_OPENED: 5314 case AudioSystem::INPUT_CONFIG_CHANGED: 5315 desc.channels = mChannelMask; 5316 desc.samplingRate = mSampleRate; 5317 desc.format = mFormat; 5318 desc.frameCount = mFrameCount; 5319 desc.latency = 0; 5320 param2 = &desc; 5321 break; 5322 5323 case AudioSystem::INPUT_CLOSED: 5324 default: 5325 break; 5326 } 5327 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5328} 5329 5330void AudioFlinger::RecordThread::readInputParameters() 5331{ 5332 delete mRsmpInBuffer; 5333 // mRsmpInBuffer is always assigned a new[] below 5334 delete mRsmpOutBuffer; 5335 mRsmpOutBuffer = NULL; 5336 delete mResampler; 5337 mResampler = NULL; 5338 5339 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5340 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5341 mChannelCount = (uint16_t)popcount(mChannelMask); 5342 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5343 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5344 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5345 mFrameCount = mInputBytes / mFrameSize; 5346 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5347 5348 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5349 { 5350 int channelCount; 5351 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5352 // stereo to mono post process as the resampler always outputs stereo. 5353 if (mChannelCount == 1 && mReqChannelCount == 2) { 5354 channelCount = 1; 5355 } else { 5356 channelCount = 2; 5357 } 5358 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5359 mResampler->setSampleRate(mSampleRate); 5360 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5361 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5362 5363 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5364 if (mChannelCount == 1 && mReqChannelCount == 1) { 5365 mFrameCount >>= 1; 5366 } 5367 5368 } 5369 mRsmpInIndex = mFrameCount; 5370} 5371 5372unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5373{ 5374 Mutex::Autolock _l(mLock); 5375 if (initCheck() != NO_ERROR) { 5376 return 0; 5377 } 5378 5379 return mInput->stream->get_input_frames_lost(mInput->stream); 5380} 5381 5382uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5383{ 5384 Mutex::Autolock _l(mLock); 5385 uint32_t result = 0; 5386 if (getEffectChain_l(sessionId) != 0) { 5387 result = EFFECT_SESSION; 5388 } 5389 5390 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5391 result |= TRACK_SESSION; 5392 } 5393 5394 return result; 5395} 5396 5397AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5398{ 5399 Mutex::Autolock _l(mLock); 5400 return mTrack; 5401} 5402 5403AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5404{ 5405 Mutex::Autolock _l(mLock); 5406 return mInput; 5407} 5408 5409AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5410{ 5411 Mutex::Autolock _l(mLock); 5412 AudioStreamIn *input = mInput; 5413 mInput = NULL; 5414 return input; 5415} 5416 5417// this method must always be called either with ThreadBase mLock held or inside the thread loop 5418audio_stream_t* AudioFlinger::RecordThread::stream() 5419{ 5420 if (mInput == NULL) { 5421 return NULL; 5422 } 5423 return &mInput->stream->common; 5424} 5425 5426 5427// ---------------------------------------------------------------------------- 5428 5429audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5430 uint32_t *pSamplingRate, 5431 audio_format_t *pFormat, 5432 uint32_t *pChannels, 5433 uint32_t *pLatencyMs, 5434 uint32_t flags) 5435{ 5436 status_t status; 5437 PlaybackThread *thread = NULL; 5438 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5439 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5440 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5441 uint32_t channels = pChannels ? *pChannels : 0; 5442 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5443 audio_stream_out_t *outStream; 5444 audio_hw_device_t *outHwDev; 5445 5446 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5447 pDevices ? *pDevices : 0, 5448 samplingRate, 5449 format, 5450 channels, 5451 flags); 5452 5453 if (pDevices == NULL || *pDevices == 0) { 5454 return 0; 5455 } 5456 5457 Mutex::Autolock _l(mLock); 5458 5459 outHwDev = findSuitableHwDev_l(*pDevices); 5460 if (outHwDev == NULL) 5461 return 0; 5462 5463 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5464 &channels, &samplingRate, &outStream); 5465 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5466 outStream, 5467 samplingRate, 5468 format, 5469 channels, 5470 status); 5471 5472 mHardwareStatus = AUDIO_HW_IDLE; 5473 if (outStream != NULL) { 5474 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5475 audio_io_handle_t id = nextUniqueId(); 5476 5477 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5478 (format != AUDIO_FORMAT_PCM_16_BIT) || 5479 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5480 thread = new DirectOutputThread(this, output, id, *pDevices); 5481 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5482 } else { 5483 thread = new MixerThread(this, output, id, *pDevices); 5484 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5485 } 5486 mPlaybackThreads.add(id, thread); 5487 5488 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5489 if (pFormat != NULL) *pFormat = format; 5490 if (pChannels != NULL) *pChannels = channels; 5491 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5492 5493 // notify client processes of the new output creation 5494 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5495 return id; 5496 } 5497 5498 return 0; 5499} 5500 5501audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5502 audio_io_handle_t output2) 5503{ 5504 Mutex::Autolock _l(mLock); 5505 MixerThread *thread1 = checkMixerThread_l(output1); 5506 MixerThread *thread2 = checkMixerThread_l(output2); 5507 5508 if (thread1 == NULL || thread2 == NULL) { 5509 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5510 return 0; 5511 } 5512 5513 audio_io_handle_t id = nextUniqueId(); 5514 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5515 thread->addOutputTrack(thread2); 5516 mPlaybackThreads.add(id, thread); 5517 // notify client processes of the new output creation 5518 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5519 return id; 5520} 5521 5522status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5523{ 5524 // keep strong reference on the playback thread so that 5525 // it is not destroyed while exit() is executed 5526 sp <PlaybackThread> thread; 5527 { 5528 Mutex::Autolock _l(mLock); 5529 thread = checkPlaybackThread_l(output); 5530 if (thread == NULL) { 5531 return BAD_VALUE; 5532 } 5533 5534 ALOGV("closeOutput() %d", output); 5535 5536 if (thread->type() == ThreadBase::MIXER) { 5537 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5538 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5539 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5540 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5541 } 5542 } 5543 } 5544 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5545 mPlaybackThreads.removeItem(output); 5546 } 5547 thread->exit(); 5548 // The thread entity (active unit of execution) is no longer running here, 5549 // but the ThreadBase container still exists. 5550 5551 if (thread->type() != ThreadBase::DUPLICATING) { 5552 AudioStreamOut *out = thread->clearOutput(); 5553 assert(out != NULL); 5554 // from now on thread->mOutput is NULL 5555 out->hwDev->close_output_stream(out->hwDev, out->stream); 5556 delete out; 5557 } 5558 return NO_ERROR; 5559} 5560 5561status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5562{ 5563 Mutex::Autolock _l(mLock); 5564 PlaybackThread *thread = checkPlaybackThread_l(output); 5565 5566 if (thread == NULL) { 5567 return BAD_VALUE; 5568 } 5569 5570 ALOGV("suspendOutput() %d", output); 5571 thread->suspend(); 5572 5573 return NO_ERROR; 5574} 5575 5576status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5577{ 5578 Mutex::Autolock _l(mLock); 5579 PlaybackThread *thread = checkPlaybackThread_l(output); 5580 5581 if (thread == NULL) { 5582 return BAD_VALUE; 5583 } 5584 5585 ALOGV("restoreOutput() %d", output); 5586 5587 thread->restore(); 5588 5589 return NO_ERROR; 5590} 5591 5592audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5593 uint32_t *pSamplingRate, 5594 audio_format_t *pFormat, 5595 uint32_t *pChannels, 5596 audio_in_acoustics_t acoustics) 5597{ 5598 status_t status; 5599 RecordThread *thread = NULL; 5600 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5601 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5602 uint32_t channels = pChannels ? *pChannels : 0; 5603 uint32_t reqSamplingRate = samplingRate; 5604 audio_format_t reqFormat = format; 5605 uint32_t reqChannels = channels; 5606 audio_stream_in_t *inStream; 5607 audio_hw_device_t *inHwDev; 5608 5609 if (pDevices == NULL || *pDevices == 0) { 5610 return 0; 5611 } 5612 5613 Mutex::Autolock _l(mLock); 5614 5615 inHwDev = findSuitableHwDev_l(*pDevices); 5616 if (inHwDev == NULL) 5617 return 0; 5618 5619 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5620 &channels, &samplingRate, 5621 acoustics, 5622 &inStream); 5623 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5624 inStream, 5625 samplingRate, 5626 format, 5627 channels, 5628 acoustics, 5629 status); 5630 5631 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5632 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5633 // or stereo to mono conversions on 16 bit PCM inputs. 5634 if (inStream == NULL && status == BAD_VALUE && 5635 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5636 (samplingRate <= 2 * reqSamplingRate) && 5637 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5638 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5639 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5640 &channels, &samplingRate, 5641 acoustics, 5642 &inStream); 5643 } 5644 5645 if (inStream != NULL) { 5646 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5647 5648 audio_io_handle_t id = nextUniqueId(); 5649 // Start record thread 5650 // RecorThread require both input and output device indication to forward to audio 5651 // pre processing modules 5652 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5653 thread = new RecordThread(this, 5654 input, 5655 reqSamplingRate, 5656 reqChannels, 5657 id, 5658 device); 5659 mRecordThreads.add(id, thread); 5660 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5661 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5662 if (pFormat != NULL) *pFormat = format; 5663 if (pChannels != NULL) *pChannels = reqChannels; 5664 5665 input->stream->common.standby(&input->stream->common); 5666 5667 // notify client processes of the new input creation 5668 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5669 return id; 5670 } 5671 5672 return 0; 5673} 5674 5675status_t AudioFlinger::closeInput(audio_io_handle_t input) 5676{ 5677 // keep strong reference on the record thread so that 5678 // it is not destroyed while exit() is executed 5679 sp <RecordThread> thread; 5680 { 5681 Mutex::Autolock _l(mLock); 5682 thread = checkRecordThread_l(input); 5683 if (thread == NULL) { 5684 return BAD_VALUE; 5685 } 5686 5687 ALOGV("closeInput() %d", input); 5688 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5689 mRecordThreads.removeItem(input); 5690 } 5691 thread->exit(); 5692 // The thread entity (active unit of execution) is no longer running here, 5693 // but the ThreadBase container still exists. 5694 5695 AudioStreamIn *in = thread->clearInput(); 5696 assert(in != NULL); 5697 // from now on thread->mInput is NULL 5698 in->hwDev->close_input_stream(in->hwDev, in->stream); 5699 delete in; 5700 5701 return NO_ERROR; 5702} 5703 5704status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5705{ 5706 Mutex::Autolock _l(mLock); 5707 MixerThread *dstThread = checkMixerThread_l(output); 5708 if (dstThread == NULL) { 5709 ALOGW("setStreamOutput() bad output id %d", output); 5710 return BAD_VALUE; 5711 } 5712 5713 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5714 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5715 5716 dstThread->setStreamValid(stream, true); 5717 5718 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5719 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5720 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5721 MixerThread *srcThread = (MixerThread *)thread; 5722 srcThread->setStreamValid(stream, false); 5723 srcThread->invalidateTracks(stream); 5724 } 5725 } 5726 5727 return NO_ERROR; 5728} 5729 5730 5731int AudioFlinger::newAudioSessionId() 5732{ 5733 return nextUniqueId(); 5734} 5735 5736void AudioFlinger::acquireAudioSessionId(int audioSession) 5737{ 5738 Mutex::Autolock _l(mLock); 5739 pid_t caller = IPCThreadState::self()->getCallingPid(); 5740 ALOGV("acquiring %d from %d", audioSession, caller); 5741 size_t num = mAudioSessionRefs.size(); 5742 for (size_t i = 0; i< num; i++) { 5743 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5744 if (ref->sessionid == audioSession && ref->pid == caller) { 5745 ref->cnt++; 5746 ALOGV(" incremented refcount to %d", ref->cnt); 5747 return; 5748 } 5749 } 5750 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5751 ALOGV(" added new entry for %d", audioSession); 5752} 5753 5754void AudioFlinger::releaseAudioSessionId(int audioSession) 5755{ 5756 Mutex::Autolock _l(mLock); 5757 pid_t caller = IPCThreadState::self()->getCallingPid(); 5758 ALOGV("releasing %d from %d", audioSession, caller); 5759 size_t num = mAudioSessionRefs.size(); 5760 for (size_t i = 0; i< num; i++) { 5761 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5762 if (ref->sessionid == audioSession && ref->pid == caller) { 5763 ref->cnt--; 5764 ALOGV(" decremented refcount to %d", ref->cnt); 5765 if (ref->cnt == 0) { 5766 mAudioSessionRefs.removeAt(i); 5767 delete ref; 5768 purgeStaleEffects_l(); 5769 } 5770 return; 5771 } 5772 } 5773 ALOGW("session id %d not found for pid %d", audioSession, caller); 5774} 5775 5776void AudioFlinger::purgeStaleEffects_l() { 5777 5778 ALOGV("purging stale effects"); 5779 5780 Vector< sp<EffectChain> > chains; 5781 5782 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5783 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5784 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5785 sp<EffectChain> ec = t->mEffectChains[j]; 5786 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5787 chains.push(ec); 5788 } 5789 } 5790 } 5791 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5792 sp<RecordThread> t = mRecordThreads.valueAt(i); 5793 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5794 sp<EffectChain> ec = t->mEffectChains[j]; 5795 chains.push(ec); 5796 } 5797 } 5798 5799 for (size_t i = 0; i < chains.size(); i++) { 5800 sp<EffectChain> ec = chains[i]; 5801 int sessionid = ec->sessionId(); 5802 sp<ThreadBase> t = ec->mThread.promote(); 5803 if (t == 0) { 5804 continue; 5805 } 5806 size_t numsessionrefs = mAudioSessionRefs.size(); 5807 bool found = false; 5808 for (size_t k = 0; k < numsessionrefs; k++) { 5809 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5810 if (ref->sessionid == sessionid) { 5811 ALOGV(" session %d still exists for %d with %d refs", 5812 sessionid, ref->pid, ref->cnt); 5813 found = true; 5814 break; 5815 } 5816 } 5817 if (!found) { 5818 // remove all effects from the chain 5819 while (ec->mEffects.size()) { 5820 sp<EffectModule> effect = ec->mEffects[0]; 5821 effect->unPin(); 5822 Mutex::Autolock _l (t->mLock); 5823 t->removeEffect_l(effect); 5824 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5825 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5826 if (handle != 0) { 5827 handle->mEffect.clear(); 5828 if (handle->mHasControl && handle->mEnabled) { 5829 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5830 } 5831 } 5832 } 5833 AudioSystem::unregisterEffect(effect->id()); 5834 } 5835 } 5836 } 5837 return; 5838} 5839 5840// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5841AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5842{ 5843 return mPlaybackThreads.valueFor(output).get(); 5844} 5845 5846// checkMixerThread_l() must be called with AudioFlinger::mLock held 5847AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5848{ 5849 PlaybackThread *thread = checkPlaybackThread_l(output); 5850 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5851} 5852 5853// checkRecordThread_l() must be called with AudioFlinger::mLock held 5854AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5855{ 5856 return mRecordThreads.valueFor(input).get(); 5857} 5858 5859uint32_t AudioFlinger::nextUniqueId() 5860{ 5861 return android_atomic_inc(&mNextUniqueId); 5862} 5863 5864AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() 5865{ 5866 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5867 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5868 AudioStreamOut *output = thread->getOutput(); 5869 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5870 return thread; 5871 } 5872 } 5873 return NULL; 5874} 5875 5876uint32_t AudioFlinger::primaryOutputDevice_l() 5877{ 5878 PlaybackThread *thread = primaryPlaybackThread_l(); 5879 5880 if (thread == NULL) { 5881 return 0; 5882 } 5883 5884 return thread->device(); 5885} 5886 5887 5888// ---------------------------------------------------------------------------- 5889// Effect management 5890// ---------------------------------------------------------------------------- 5891 5892 5893status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5894{ 5895 Mutex::Autolock _l(mLock); 5896 return EffectQueryNumberEffects(numEffects); 5897} 5898 5899status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5900{ 5901 Mutex::Autolock _l(mLock); 5902 return EffectQueryEffect(index, descriptor); 5903} 5904 5905status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5906 effect_descriptor_t *descriptor) const 5907{ 5908 Mutex::Autolock _l(mLock); 5909 return EffectGetDescriptor(pUuid, descriptor); 5910} 5911 5912 5913sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5914 effect_descriptor_t *pDesc, 5915 const sp<IEffectClient>& effectClient, 5916 int32_t priority, 5917 audio_io_handle_t io, 5918 int sessionId, 5919 status_t *status, 5920 int *id, 5921 int *enabled) 5922{ 5923 status_t lStatus = NO_ERROR; 5924 sp<EffectHandle> handle; 5925 effect_descriptor_t desc; 5926 5927 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5928 pid, effectClient.get(), priority, sessionId, io); 5929 5930 if (pDesc == NULL) { 5931 lStatus = BAD_VALUE; 5932 goto Exit; 5933 } 5934 5935 // check audio settings permission for global effects 5936 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5937 lStatus = PERMISSION_DENIED; 5938 goto Exit; 5939 } 5940 5941 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5942 // that can only be created by audio policy manager (running in same process) 5943 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5944 lStatus = PERMISSION_DENIED; 5945 goto Exit; 5946 } 5947 5948 if (io == 0) { 5949 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5950 // output must be specified by AudioPolicyManager when using session 5951 // AUDIO_SESSION_OUTPUT_STAGE 5952 lStatus = BAD_VALUE; 5953 goto Exit; 5954 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5955 // if the output returned by getOutputForEffect() is removed before we lock the 5956 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5957 // and we will exit safely 5958 io = AudioSystem::getOutputForEffect(&desc); 5959 } 5960 } 5961 5962 { 5963 Mutex::Autolock _l(mLock); 5964 5965 5966 if (!EffectIsNullUuid(&pDesc->uuid)) { 5967 // if uuid is specified, request effect descriptor 5968 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5969 if (lStatus < 0) { 5970 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5971 goto Exit; 5972 } 5973 } else { 5974 // if uuid is not specified, look for an available implementation 5975 // of the required type in effect factory 5976 if (EffectIsNullUuid(&pDesc->type)) { 5977 ALOGW("createEffect() no effect type"); 5978 lStatus = BAD_VALUE; 5979 goto Exit; 5980 } 5981 uint32_t numEffects = 0; 5982 effect_descriptor_t d; 5983 d.flags = 0; // prevent compiler warning 5984 bool found = false; 5985 5986 lStatus = EffectQueryNumberEffects(&numEffects); 5987 if (lStatus < 0) { 5988 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5989 goto Exit; 5990 } 5991 for (uint32_t i = 0; i < numEffects; i++) { 5992 lStatus = EffectQueryEffect(i, &desc); 5993 if (lStatus < 0) { 5994 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5995 continue; 5996 } 5997 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5998 // If matching type found save effect descriptor. If the session is 5999 // 0 and the effect is not auxiliary, continue enumeration in case 6000 // an auxiliary version of this effect type is available 6001 found = true; 6002 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 6003 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 6004 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6005 break; 6006 } 6007 } 6008 } 6009 if (!found) { 6010 lStatus = BAD_VALUE; 6011 ALOGW("createEffect() effect not found"); 6012 goto Exit; 6013 } 6014 // For same effect type, chose auxiliary version over insert version if 6015 // connect to output mix (Compliance to OpenSL ES) 6016 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 6017 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 6018 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 6019 } 6020 } 6021 6022 // Do not allow auxiliary effects on a session different from 0 (output mix) 6023 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 6024 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6025 lStatus = INVALID_OPERATION; 6026 goto Exit; 6027 } 6028 6029 // check recording permission for visualizer 6030 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 6031 !recordingAllowed()) { 6032 lStatus = PERMISSION_DENIED; 6033 goto Exit; 6034 } 6035 6036 // return effect descriptor 6037 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 6038 6039 // If output is not specified try to find a matching audio session ID in one of the 6040 // output threads. 6041 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 6042 // because of code checking output when entering the function. 6043 // Note: io is never 0 when creating an effect on an input 6044 if (io == 0) { 6045 // look for the thread where the specified audio session is present 6046 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 6047 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6048 io = mPlaybackThreads.keyAt(i); 6049 break; 6050 } 6051 } 6052 if (io == 0) { 6053 for (size_t i = 0; i < mRecordThreads.size(); i++) { 6054 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 6055 io = mRecordThreads.keyAt(i); 6056 break; 6057 } 6058 } 6059 } 6060 // If no output thread contains the requested session ID, default to 6061 // first output. The effect chain will be moved to the correct output 6062 // thread when a track with the same session ID is created 6063 if (io == 0 && mPlaybackThreads.size()) { 6064 io = mPlaybackThreads.keyAt(0); 6065 } 6066 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6067 } 6068 ThreadBase *thread = checkRecordThread_l(io); 6069 if (thread == NULL) { 6070 thread = checkPlaybackThread_l(io); 6071 if (thread == NULL) { 6072 ALOGE("createEffect() unknown output thread"); 6073 lStatus = BAD_VALUE; 6074 goto Exit; 6075 } 6076 } 6077 6078 sp<Client> client = registerPid_l(pid); 6079 6080 // create effect on selected output thread 6081 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6082 &desc, enabled, &lStatus); 6083 if (handle != 0 && id != NULL) { 6084 *id = handle->id(); 6085 } 6086 } 6087 6088Exit: 6089 if(status) { 6090 *status = lStatus; 6091 } 6092 return handle; 6093} 6094 6095status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6096 audio_io_handle_t dstOutput) 6097{ 6098 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6099 sessionId, srcOutput, dstOutput); 6100 Mutex::Autolock _l(mLock); 6101 if (srcOutput == dstOutput) { 6102 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6103 return NO_ERROR; 6104 } 6105 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6106 if (srcThread == NULL) { 6107 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6108 return BAD_VALUE; 6109 } 6110 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6111 if (dstThread == NULL) { 6112 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6113 return BAD_VALUE; 6114 } 6115 6116 Mutex::Autolock _dl(dstThread->mLock); 6117 Mutex::Autolock _sl(srcThread->mLock); 6118 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6119 6120 return NO_ERROR; 6121} 6122 6123// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6124status_t AudioFlinger::moveEffectChain_l(int sessionId, 6125 AudioFlinger::PlaybackThread *srcThread, 6126 AudioFlinger::PlaybackThread *dstThread, 6127 bool reRegister) 6128{ 6129 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6130 sessionId, srcThread, dstThread); 6131 6132 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6133 if (chain == 0) { 6134 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6135 sessionId, srcThread); 6136 return INVALID_OPERATION; 6137 } 6138 6139 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6140 // so that a new chain is created with correct parameters when first effect is added. This is 6141 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6142 // removed. 6143 srcThread->removeEffectChain_l(chain); 6144 6145 // transfer all effects one by one so that new effect chain is created on new thread with 6146 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6147 audio_io_handle_t dstOutput = dstThread->id(); 6148 sp<EffectChain> dstChain; 6149 uint32_t strategy = 0; // prevent compiler warning 6150 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6151 while (effect != 0) { 6152 srcThread->removeEffect_l(effect); 6153 dstThread->addEffect_l(effect); 6154 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6155 if (effect->state() == EffectModule::ACTIVE || 6156 effect->state() == EffectModule::STOPPING) { 6157 effect->start(); 6158 } 6159 // if the move request is not received from audio policy manager, the effect must be 6160 // re-registered with the new strategy and output 6161 if (dstChain == 0) { 6162 dstChain = effect->chain().promote(); 6163 if (dstChain == 0) { 6164 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6165 srcThread->addEffect_l(effect); 6166 return NO_INIT; 6167 } 6168 strategy = dstChain->strategy(); 6169 } 6170 if (reRegister) { 6171 AudioSystem::unregisterEffect(effect->id()); 6172 AudioSystem::registerEffect(&effect->desc(), 6173 dstOutput, 6174 strategy, 6175 sessionId, 6176 effect->id()); 6177 } 6178 effect = chain->getEffectFromId_l(0); 6179 } 6180 6181 return NO_ERROR; 6182} 6183 6184 6185// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6186sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6187 const sp<AudioFlinger::Client>& client, 6188 const sp<IEffectClient>& effectClient, 6189 int32_t priority, 6190 int sessionId, 6191 effect_descriptor_t *desc, 6192 int *enabled, 6193 status_t *status 6194 ) 6195{ 6196 sp<EffectModule> effect; 6197 sp<EffectHandle> handle; 6198 status_t lStatus; 6199 sp<EffectChain> chain; 6200 bool chainCreated = false; 6201 bool effectCreated = false; 6202 bool effectRegistered = false; 6203 6204 lStatus = initCheck(); 6205 if (lStatus != NO_ERROR) { 6206 ALOGW("createEffect_l() Audio driver not initialized."); 6207 goto Exit; 6208 } 6209 6210 // Do not allow effects with session ID 0 on direct output or duplicating threads 6211 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6212 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6213 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6214 desc->name, sessionId); 6215 lStatus = BAD_VALUE; 6216 goto Exit; 6217 } 6218 // Only Pre processor effects are allowed on input threads and only on input threads 6219 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6220 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6221 desc->name, desc->flags, mType); 6222 lStatus = BAD_VALUE; 6223 goto Exit; 6224 } 6225 6226 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6227 6228 { // scope for mLock 6229 Mutex::Autolock _l(mLock); 6230 6231 // check for existing effect chain with the requested audio session 6232 chain = getEffectChain_l(sessionId); 6233 if (chain == 0) { 6234 // create a new chain for this session 6235 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6236 chain = new EffectChain(this, sessionId); 6237 addEffectChain_l(chain); 6238 chain->setStrategy(getStrategyForSession_l(sessionId)); 6239 chainCreated = true; 6240 } else { 6241 effect = chain->getEffectFromDesc_l(desc); 6242 } 6243 6244 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6245 6246 if (effect == 0) { 6247 int id = mAudioFlinger->nextUniqueId(); 6248 // Check CPU and memory usage 6249 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6250 if (lStatus != NO_ERROR) { 6251 goto Exit; 6252 } 6253 effectRegistered = true; 6254 // create a new effect module if none present in the chain 6255 effect = new EffectModule(this, chain, desc, id, sessionId); 6256 lStatus = effect->status(); 6257 if (lStatus != NO_ERROR) { 6258 goto Exit; 6259 } 6260 lStatus = chain->addEffect_l(effect); 6261 if (lStatus != NO_ERROR) { 6262 goto Exit; 6263 } 6264 effectCreated = true; 6265 6266 effect->setDevice(mDevice); 6267 effect->setMode(mAudioFlinger->getMode()); 6268 } 6269 // create effect handle and connect it to effect module 6270 handle = new EffectHandle(effect, client, effectClient, priority); 6271 lStatus = effect->addHandle(handle); 6272 if (enabled != NULL) { 6273 *enabled = (int)effect->isEnabled(); 6274 } 6275 } 6276 6277Exit: 6278 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6279 Mutex::Autolock _l(mLock); 6280 if (effectCreated) { 6281 chain->removeEffect_l(effect); 6282 } 6283 if (effectRegistered) { 6284 AudioSystem::unregisterEffect(effect->id()); 6285 } 6286 if (chainCreated) { 6287 removeEffectChain_l(chain); 6288 } 6289 handle.clear(); 6290 } 6291 6292 if(status) { 6293 *status = lStatus; 6294 } 6295 return handle; 6296} 6297 6298sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6299{ 6300 sp<EffectChain> chain = getEffectChain_l(sessionId); 6301 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6302} 6303 6304// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6305// PlaybackThread::mLock held 6306status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6307{ 6308 // check for existing effect chain with the requested audio session 6309 int sessionId = effect->sessionId(); 6310 sp<EffectChain> chain = getEffectChain_l(sessionId); 6311 bool chainCreated = false; 6312 6313 if (chain == 0) { 6314 // create a new chain for this session 6315 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6316 chain = new EffectChain(this, sessionId); 6317 addEffectChain_l(chain); 6318 chain->setStrategy(getStrategyForSession_l(sessionId)); 6319 chainCreated = true; 6320 } 6321 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6322 6323 if (chain->getEffectFromId_l(effect->id()) != 0) { 6324 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6325 this, effect->desc().name, chain.get()); 6326 return BAD_VALUE; 6327 } 6328 6329 status_t status = chain->addEffect_l(effect); 6330 if (status != NO_ERROR) { 6331 if (chainCreated) { 6332 removeEffectChain_l(chain); 6333 } 6334 return status; 6335 } 6336 6337 effect->setDevice(mDevice); 6338 effect->setMode(mAudioFlinger->getMode()); 6339 return NO_ERROR; 6340} 6341 6342void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6343 6344 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6345 effect_descriptor_t desc = effect->desc(); 6346 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6347 detachAuxEffect_l(effect->id()); 6348 } 6349 6350 sp<EffectChain> chain = effect->chain().promote(); 6351 if (chain != 0) { 6352 // remove effect chain if removing last effect 6353 if (chain->removeEffect_l(effect) == 0) { 6354 removeEffectChain_l(chain); 6355 } 6356 } else { 6357 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6358 } 6359} 6360 6361void AudioFlinger::ThreadBase::lockEffectChains_l( 6362 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6363{ 6364 effectChains = mEffectChains; 6365 for (size_t i = 0; i < mEffectChains.size(); i++) { 6366 mEffectChains[i]->lock(); 6367 } 6368} 6369 6370void AudioFlinger::ThreadBase::unlockEffectChains( 6371 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6372{ 6373 for (size_t i = 0; i < effectChains.size(); i++) { 6374 effectChains[i]->unlock(); 6375 } 6376} 6377 6378sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6379{ 6380 Mutex::Autolock _l(mLock); 6381 return getEffectChain_l(sessionId); 6382} 6383 6384sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6385{ 6386 size_t size = mEffectChains.size(); 6387 for (size_t i = 0; i < size; i++) { 6388 if (mEffectChains[i]->sessionId() == sessionId) { 6389 return mEffectChains[i]; 6390 } 6391 } 6392 return 0; 6393} 6394 6395void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6396{ 6397 Mutex::Autolock _l(mLock); 6398 size_t size = mEffectChains.size(); 6399 for (size_t i = 0; i < size; i++) { 6400 mEffectChains[i]->setMode_l(mode); 6401 } 6402} 6403 6404void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6405 const wp<EffectHandle>& handle, 6406 bool unpinIfLast) { 6407 6408 Mutex::Autolock _l(mLock); 6409 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6410 // delete the effect module if removing last handle on it 6411 if (effect->removeHandle(handle) == 0) { 6412 if (!effect->isPinned() || unpinIfLast) { 6413 removeEffect_l(effect); 6414 AudioSystem::unregisterEffect(effect->id()); 6415 } 6416 } 6417} 6418 6419status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6420{ 6421 int session = chain->sessionId(); 6422 int16_t *buffer = mMixBuffer; 6423 bool ownsBuffer = false; 6424 6425 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6426 if (session > 0) { 6427 // Only one effect chain can be present in direct output thread and it uses 6428 // the mix buffer as input 6429 if (mType != DIRECT) { 6430 size_t numSamples = mFrameCount * mChannelCount; 6431 buffer = new int16_t[numSamples]; 6432 memset(buffer, 0, numSamples * sizeof(int16_t)); 6433 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6434 ownsBuffer = true; 6435 } 6436 6437 // Attach all tracks with same session ID to this chain. 6438 for (size_t i = 0; i < mTracks.size(); ++i) { 6439 sp<Track> track = mTracks[i]; 6440 if (session == track->sessionId()) { 6441 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6442 track->setMainBuffer(buffer); 6443 chain->incTrackCnt(); 6444 } 6445 } 6446 6447 // indicate all active tracks in the chain 6448 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6449 sp<Track> track = mActiveTracks[i].promote(); 6450 if (track == 0) continue; 6451 if (session == track->sessionId()) { 6452 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6453 chain->incActiveTrackCnt(); 6454 } 6455 } 6456 } 6457 6458 chain->setInBuffer(buffer, ownsBuffer); 6459 chain->setOutBuffer(mMixBuffer); 6460 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6461 // chains list in order to be processed last as it contains output stage effects 6462 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6463 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6464 // after track specific effects and before output stage 6465 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6466 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6467 // Effect chain for other sessions are inserted at beginning of effect 6468 // chains list to be processed before output mix effects. Relative order between other 6469 // sessions is not important 6470 size_t size = mEffectChains.size(); 6471 size_t i = 0; 6472 for (i = 0; i < size; i++) { 6473 if (mEffectChains[i]->sessionId() < session) break; 6474 } 6475 mEffectChains.insertAt(chain, i); 6476 checkSuspendOnAddEffectChain_l(chain); 6477 6478 return NO_ERROR; 6479} 6480 6481size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6482{ 6483 int session = chain->sessionId(); 6484 6485 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6486 6487 for (size_t i = 0; i < mEffectChains.size(); i++) { 6488 if (chain == mEffectChains[i]) { 6489 mEffectChains.removeAt(i); 6490 // detach all active tracks from the chain 6491 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6492 sp<Track> track = mActiveTracks[i].promote(); 6493 if (track == 0) continue; 6494 if (session == track->sessionId()) { 6495 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6496 chain.get(), session); 6497 chain->decActiveTrackCnt(); 6498 } 6499 } 6500 6501 // detach all tracks with same session ID from this chain 6502 for (size_t i = 0; i < mTracks.size(); ++i) { 6503 sp<Track> track = mTracks[i]; 6504 if (session == track->sessionId()) { 6505 track->setMainBuffer(mMixBuffer); 6506 chain->decTrackCnt(); 6507 } 6508 } 6509 break; 6510 } 6511 } 6512 return mEffectChains.size(); 6513} 6514 6515status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6516 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6517{ 6518 Mutex::Autolock _l(mLock); 6519 return attachAuxEffect_l(track, EffectId); 6520} 6521 6522status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6523 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6524{ 6525 status_t status = NO_ERROR; 6526 6527 if (EffectId == 0) { 6528 track->setAuxBuffer(0, NULL); 6529 } else { 6530 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6531 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6532 if (effect != 0) { 6533 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6534 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6535 } else { 6536 status = INVALID_OPERATION; 6537 } 6538 } else { 6539 status = BAD_VALUE; 6540 } 6541 } 6542 return status; 6543} 6544 6545void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6546{ 6547 for (size_t i = 0; i < mTracks.size(); ++i) { 6548 sp<Track> track = mTracks[i]; 6549 if (track->auxEffectId() == effectId) { 6550 attachAuxEffect_l(track, 0); 6551 } 6552 } 6553} 6554 6555status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6556{ 6557 // only one chain per input thread 6558 if (mEffectChains.size() != 0) { 6559 return INVALID_OPERATION; 6560 } 6561 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6562 6563 chain->setInBuffer(NULL); 6564 chain->setOutBuffer(NULL); 6565 6566 checkSuspendOnAddEffectChain_l(chain); 6567 6568 mEffectChains.add(chain); 6569 6570 return NO_ERROR; 6571} 6572 6573size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6574{ 6575 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6576 ALOGW_IF(mEffectChains.size() != 1, 6577 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6578 chain.get(), mEffectChains.size(), this); 6579 if (mEffectChains.size() == 1) { 6580 mEffectChains.removeAt(0); 6581 } 6582 return 0; 6583} 6584 6585// ---------------------------------------------------------------------------- 6586// EffectModule implementation 6587// ---------------------------------------------------------------------------- 6588 6589#undef LOG_TAG 6590#define LOG_TAG "AudioFlinger::EffectModule" 6591 6592AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6593 const wp<AudioFlinger::EffectChain>& chain, 6594 effect_descriptor_t *desc, 6595 int id, 6596 int sessionId) 6597 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6598 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6599{ 6600 ALOGV("Constructor %p", this); 6601 int lStatus; 6602 if (thread == NULL) { 6603 return; 6604 } 6605 6606 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6607 6608 // create effect engine from effect factory 6609 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6610 6611 if (mStatus != NO_ERROR) { 6612 return; 6613 } 6614 lStatus = init(); 6615 if (lStatus < 0) { 6616 mStatus = lStatus; 6617 goto Error; 6618 } 6619 6620 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6621 mPinned = true; 6622 } 6623 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6624 return; 6625Error: 6626 EffectRelease(mEffectInterface); 6627 mEffectInterface = NULL; 6628 ALOGV("Constructor Error %d", mStatus); 6629} 6630 6631AudioFlinger::EffectModule::~EffectModule() 6632{ 6633 ALOGV("Destructor %p", this); 6634 if (mEffectInterface != NULL) { 6635 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6636 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6637 sp<ThreadBase> thread = mThread.promote(); 6638 if (thread != 0) { 6639 audio_stream_t *stream = thread->stream(); 6640 if (stream != NULL) { 6641 stream->remove_audio_effect(stream, mEffectInterface); 6642 } 6643 } 6644 } 6645 // release effect engine 6646 EffectRelease(mEffectInterface); 6647 } 6648} 6649 6650status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6651{ 6652 status_t status; 6653 6654 Mutex::Autolock _l(mLock); 6655 int priority = handle->priority(); 6656 size_t size = mHandles.size(); 6657 sp<EffectHandle> h; 6658 size_t i; 6659 for (i = 0; i < size; i++) { 6660 h = mHandles[i].promote(); 6661 if (h == 0) continue; 6662 if (h->priority() <= priority) break; 6663 } 6664 // if inserted in first place, move effect control from previous owner to this handle 6665 if (i == 0) { 6666 bool enabled = false; 6667 if (h != 0) { 6668 enabled = h->enabled(); 6669 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6670 } 6671 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6672 status = NO_ERROR; 6673 } else { 6674 status = ALREADY_EXISTS; 6675 } 6676 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6677 mHandles.insertAt(handle, i); 6678 return status; 6679} 6680 6681size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6682{ 6683 Mutex::Autolock _l(mLock); 6684 size_t size = mHandles.size(); 6685 size_t i; 6686 for (i = 0; i < size; i++) { 6687 if (mHandles[i] == handle) break; 6688 } 6689 if (i == size) { 6690 return size; 6691 } 6692 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6693 6694 bool enabled = false; 6695 EffectHandle *hdl = handle.unsafe_get(); 6696 if (hdl != NULL) { 6697 ALOGV("removeHandle() unsafe_get OK"); 6698 enabled = hdl->enabled(); 6699 } 6700 mHandles.removeAt(i); 6701 size = mHandles.size(); 6702 // if removed from first place, move effect control from this handle to next in line 6703 if (i == 0 && size != 0) { 6704 sp<EffectHandle> h = mHandles[0].promote(); 6705 if (h != 0) { 6706 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6707 } 6708 } 6709 6710 // Prevent calls to process() and other functions on effect interface from now on. 6711 // The effect engine will be released by the destructor when the last strong reference on 6712 // this object is released which can happen after next process is called. 6713 if (size == 0 && !mPinned) { 6714 mState = DESTROYED; 6715 } 6716 6717 return size; 6718} 6719 6720sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6721{ 6722 Mutex::Autolock _l(mLock); 6723 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6724} 6725 6726void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6727{ 6728 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6729 // keep a strong reference on this EffectModule to avoid calling the 6730 // destructor before we exit 6731 sp<EffectModule> keep(this); 6732 { 6733 sp<ThreadBase> thread = mThread.promote(); 6734 if (thread != 0) { 6735 thread->disconnectEffect(keep, handle, unpinIfLast); 6736 } 6737 } 6738} 6739 6740void AudioFlinger::EffectModule::updateState() { 6741 Mutex::Autolock _l(mLock); 6742 6743 switch (mState) { 6744 case RESTART: 6745 reset_l(); 6746 // FALL THROUGH 6747 6748 case STARTING: 6749 // clear auxiliary effect input buffer for next accumulation 6750 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6751 memset(mConfig.inputCfg.buffer.raw, 6752 0, 6753 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6754 } 6755 start_l(); 6756 mState = ACTIVE; 6757 break; 6758 case STOPPING: 6759 stop_l(); 6760 mDisableWaitCnt = mMaxDisableWaitCnt; 6761 mState = STOPPED; 6762 break; 6763 case STOPPED: 6764 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6765 // turn off sequence. 6766 if (--mDisableWaitCnt == 0) { 6767 reset_l(); 6768 mState = IDLE; 6769 } 6770 break; 6771 default: //IDLE , ACTIVE, DESTROYED 6772 break; 6773 } 6774} 6775 6776void AudioFlinger::EffectModule::process() 6777{ 6778 Mutex::Autolock _l(mLock); 6779 6780 if (mState == DESTROYED || mEffectInterface == NULL || 6781 mConfig.inputCfg.buffer.raw == NULL || 6782 mConfig.outputCfg.buffer.raw == NULL) { 6783 return; 6784 } 6785 6786 if (isProcessEnabled()) { 6787 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6788 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6789 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6790 mConfig.inputCfg.buffer.s32, 6791 mConfig.inputCfg.buffer.frameCount/2); 6792 } 6793 6794 // do the actual processing in the effect engine 6795 int ret = (*mEffectInterface)->process(mEffectInterface, 6796 &mConfig.inputCfg.buffer, 6797 &mConfig.outputCfg.buffer); 6798 6799 // force transition to IDLE state when engine is ready 6800 if (mState == STOPPED && ret == -ENODATA) { 6801 mDisableWaitCnt = 1; 6802 } 6803 6804 // clear auxiliary effect input buffer for next accumulation 6805 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6806 memset(mConfig.inputCfg.buffer.raw, 0, 6807 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6808 } 6809 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6810 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6811 // If an insert effect is idle and input buffer is different from output buffer, 6812 // accumulate input onto output 6813 sp<EffectChain> chain = mChain.promote(); 6814 if (chain != 0 && chain->activeTrackCnt() != 0) { 6815 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6816 int16_t *in = mConfig.inputCfg.buffer.s16; 6817 int16_t *out = mConfig.outputCfg.buffer.s16; 6818 for (size_t i = 0; i < frameCnt; i++) { 6819 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6820 } 6821 } 6822 } 6823} 6824 6825void AudioFlinger::EffectModule::reset_l() 6826{ 6827 if (mEffectInterface == NULL) { 6828 return; 6829 } 6830 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6831} 6832 6833status_t AudioFlinger::EffectModule::configure() 6834{ 6835 uint32_t channels; 6836 if (mEffectInterface == NULL) { 6837 return NO_INIT; 6838 } 6839 6840 sp<ThreadBase> thread = mThread.promote(); 6841 if (thread == 0) { 6842 return DEAD_OBJECT; 6843 } 6844 6845 // TODO: handle configuration of effects replacing track process 6846 if (thread->channelCount() == 1) { 6847 channels = AUDIO_CHANNEL_OUT_MONO; 6848 } else { 6849 channels = AUDIO_CHANNEL_OUT_STEREO; 6850 } 6851 6852 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6853 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6854 } else { 6855 mConfig.inputCfg.channels = channels; 6856 } 6857 mConfig.outputCfg.channels = channels; 6858 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6859 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6860 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6861 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6862 mConfig.inputCfg.bufferProvider.cookie = NULL; 6863 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6864 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6865 mConfig.outputCfg.bufferProvider.cookie = NULL; 6866 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6867 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6868 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6869 // Insert effect: 6870 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6871 // always overwrites output buffer: input buffer == output buffer 6872 // - in other sessions: 6873 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6874 // other effect: overwrites output buffer: input buffer == output buffer 6875 // Auxiliary effect: 6876 // accumulates in output buffer: input buffer != output buffer 6877 // Therefore: accumulate <=> input buffer != output buffer 6878 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6879 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6880 } else { 6881 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6882 } 6883 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6884 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6885 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6886 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6887 6888 ALOGV("configure() %p thread %p buffer %p framecount %d", 6889 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6890 6891 status_t cmdStatus; 6892 uint32_t size = sizeof(int); 6893 status_t status = (*mEffectInterface)->command(mEffectInterface, 6894 EFFECT_CMD_SET_CONFIG, 6895 sizeof(effect_config_t), 6896 &mConfig, 6897 &size, 6898 &cmdStatus); 6899 if (status == 0) { 6900 status = cmdStatus; 6901 } 6902 6903 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6904 (1000 * mConfig.outputCfg.buffer.frameCount); 6905 6906 return status; 6907} 6908 6909status_t AudioFlinger::EffectModule::init() 6910{ 6911 Mutex::Autolock _l(mLock); 6912 if (mEffectInterface == NULL) { 6913 return NO_INIT; 6914 } 6915 status_t cmdStatus; 6916 uint32_t size = sizeof(status_t); 6917 status_t status = (*mEffectInterface)->command(mEffectInterface, 6918 EFFECT_CMD_INIT, 6919 0, 6920 NULL, 6921 &size, 6922 &cmdStatus); 6923 if (status == 0) { 6924 status = cmdStatus; 6925 } 6926 return status; 6927} 6928 6929status_t AudioFlinger::EffectModule::start() 6930{ 6931 Mutex::Autolock _l(mLock); 6932 return start_l(); 6933} 6934 6935status_t AudioFlinger::EffectModule::start_l() 6936{ 6937 if (mEffectInterface == NULL) { 6938 return NO_INIT; 6939 } 6940 status_t cmdStatus; 6941 uint32_t size = sizeof(status_t); 6942 status_t status = (*mEffectInterface)->command(mEffectInterface, 6943 EFFECT_CMD_ENABLE, 6944 0, 6945 NULL, 6946 &size, 6947 &cmdStatus); 6948 if (status == 0) { 6949 status = cmdStatus; 6950 } 6951 if (status == 0 && 6952 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6953 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6954 sp<ThreadBase> thread = mThread.promote(); 6955 if (thread != 0) { 6956 audio_stream_t *stream = thread->stream(); 6957 if (stream != NULL) { 6958 stream->add_audio_effect(stream, mEffectInterface); 6959 } 6960 } 6961 } 6962 return status; 6963} 6964 6965status_t AudioFlinger::EffectModule::stop() 6966{ 6967 Mutex::Autolock _l(mLock); 6968 return stop_l(); 6969} 6970 6971status_t AudioFlinger::EffectModule::stop_l() 6972{ 6973 if (mEffectInterface == NULL) { 6974 return NO_INIT; 6975 } 6976 status_t cmdStatus; 6977 uint32_t size = sizeof(status_t); 6978 status_t status = (*mEffectInterface)->command(mEffectInterface, 6979 EFFECT_CMD_DISABLE, 6980 0, 6981 NULL, 6982 &size, 6983 &cmdStatus); 6984 if (status == 0) { 6985 status = cmdStatus; 6986 } 6987 if (status == 0 && 6988 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6989 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6990 sp<ThreadBase> thread = mThread.promote(); 6991 if (thread != 0) { 6992 audio_stream_t *stream = thread->stream(); 6993 if (stream != NULL) { 6994 stream->remove_audio_effect(stream, mEffectInterface); 6995 } 6996 } 6997 } 6998 return status; 6999} 7000 7001status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 7002 uint32_t cmdSize, 7003 void *pCmdData, 7004 uint32_t *replySize, 7005 void *pReplyData) 7006{ 7007 Mutex::Autolock _l(mLock); 7008// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 7009 7010 if (mState == DESTROYED || mEffectInterface == NULL) { 7011 return NO_INIT; 7012 } 7013 status_t status = (*mEffectInterface)->command(mEffectInterface, 7014 cmdCode, 7015 cmdSize, 7016 pCmdData, 7017 replySize, 7018 pReplyData); 7019 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 7020 uint32_t size = (replySize == NULL) ? 0 : *replySize; 7021 for (size_t i = 1; i < mHandles.size(); i++) { 7022 sp<EffectHandle> h = mHandles[i].promote(); 7023 if (h != 0) { 7024 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 7025 } 7026 } 7027 } 7028 return status; 7029} 7030 7031status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 7032{ 7033 7034 Mutex::Autolock _l(mLock); 7035 ALOGV("setEnabled %p enabled %d", this, enabled); 7036 7037 if (enabled != isEnabled()) { 7038 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 7039 if (enabled && status != NO_ERROR) { 7040 return status; 7041 } 7042 7043 switch (mState) { 7044 // going from disabled to enabled 7045 case IDLE: 7046 mState = STARTING; 7047 break; 7048 case STOPPED: 7049 mState = RESTART; 7050 break; 7051 case STOPPING: 7052 mState = ACTIVE; 7053 break; 7054 7055 // going from enabled to disabled 7056 case RESTART: 7057 mState = STOPPED; 7058 break; 7059 case STARTING: 7060 mState = IDLE; 7061 break; 7062 case ACTIVE: 7063 mState = STOPPING; 7064 break; 7065 case DESTROYED: 7066 return NO_ERROR; // simply ignore as we are being destroyed 7067 } 7068 for (size_t i = 1; i < mHandles.size(); i++) { 7069 sp<EffectHandle> h = mHandles[i].promote(); 7070 if (h != 0) { 7071 h->setEnabled(enabled); 7072 } 7073 } 7074 } 7075 return NO_ERROR; 7076} 7077 7078bool AudioFlinger::EffectModule::isEnabled() const 7079{ 7080 switch (mState) { 7081 case RESTART: 7082 case STARTING: 7083 case ACTIVE: 7084 return true; 7085 case IDLE: 7086 case STOPPING: 7087 case STOPPED: 7088 case DESTROYED: 7089 default: 7090 return false; 7091 } 7092} 7093 7094bool AudioFlinger::EffectModule::isProcessEnabled() const 7095{ 7096 switch (mState) { 7097 case RESTART: 7098 case ACTIVE: 7099 case STOPPING: 7100 case STOPPED: 7101 return true; 7102 case IDLE: 7103 case STARTING: 7104 case DESTROYED: 7105 default: 7106 return false; 7107 } 7108} 7109 7110status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7111{ 7112 Mutex::Autolock _l(mLock); 7113 status_t status = NO_ERROR; 7114 7115 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7116 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7117 if (isProcessEnabled() && 7118 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7119 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7120 status_t cmdStatus; 7121 uint32_t volume[2]; 7122 uint32_t *pVolume = NULL; 7123 uint32_t size = sizeof(volume); 7124 volume[0] = *left; 7125 volume[1] = *right; 7126 if (controller) { 7127 pVolume = volume; 7128 } 7129 status = (*mEffectInterface)->command(mEffectInterface, 7130 EFFECT_CMD_SET_VOLUME, 7131 size, 7132 volume, 7133 &size, 7134 pVolume); 7135 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7136 *left = volume[0]; 7137 *right = volume[1]; 7138 } 7139 } 7140 return status; 7141} 7142 7143status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7144{ 7145 Mutex::Autolock _l(mLock); 7146 status_t status = NO_ERROR; 7147 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7148 // audio pre processing modules on RecordThread can receive both output and 7149 // input device indication in the same call 7150 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7151 if (dev) { 7152 status_t cmdStatus; 7153 uint32_t size = sizeof(status_t); 7154 7155 status = (*mEffectInterface)->command(mEffectInterface, 7156 EFFECT_CMD_SET_DEVICE, 7157 sizeof(uint32_t), 7158 &dev, 7159 &size, 7160 &cmdStatus); 7161 if (status == NO_ERROR) { 7162 status = cmdStatus; 7163 } 7164 } 7165 dev = device & AUDIO_DEVICE_IN_ALL; 7166 if (dev) { 7167 status_t cmdStatus; 7168 uint32_t size = sizeof(status_t); 7169 7170 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7171 EFFECT_CMD_SET_INPUT_DEVICE, 7172 sizeof(uint32_t), 7173 &dev, 7174 &size, 7175 &cmdStatus); 7176 if (status2 == NO_ERROR) { 7177 status2 = cmdStatus; 7178 } 7179 if (status == NO_ERROR) { 7180 status = status2; 7181 } 7182 } 7183 } 7184 return status; 7185} 7186 7187status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7188{ 7189 Mutex::Autolock _l(mLock); 7190 status_t status = NO_ERROR; 7191 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7192 status_t cmdStatus; 7193 uint32_t size = sizeof(status_t); 7194 status = (*mEffectInterface)->command(mEffectInterface, 7195 EFFECT_CMD_SET_AUDIO_MODE, 7196 sizeof(audio_mode_t), 7197 &mode, 7198 &size, 7199 &cmdStatus); 7200 if (status == NO_ERROR) { 7201 status = cmdStatus; 7202 } 7203 } 7204 return status; 7205} 7206 7207void AudioFlinger::EffectModule::setSuspended(bool suspended) 7208{ 7209 Mutex::Autolock _l(mLock); 7210 mSuspended = suspended; 7211} 7212 7213bool AudioFlinger::EffectModule::suspended() const 7214{ 7215 Mutex::Autolock _l(mLock); 7216 return mSuspended; 7217} 7218 7219status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7220{ 7221 const size_t SIZE = 256; 7222 char buffer[SIZE]; 7223 String8 result; 7224 7225 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7226 result.append(buffer); 7227 7228 bool locked = tryLock(mLock); 7229 // failed to lock - AudioFlinger is probably deadlocked 7230 if (!locked) { 7231 result.append("\t\tCould not lock Fx mutex:\n"); 7232 } 7233 7234 result.append("\t\tSession Status State Engine:\n"); 7235 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7236 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7237 result.append(buffer); 7238 7239 result.append("\t\tDescriptor:\n"); 7240 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7241 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7242 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7243 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7244 result.append(buffer); 7245 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7246 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7247 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7248 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7249 result.append(buffer); 7250 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7251 mDescriptor.apiVersion, 7252 mDescriptor.flags); 7253 result.append(buffer); 7254 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7255 mDescriptor.name); 7256 result.append(buffer); 7257 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7258 mDescriptor.implementor); 7259 result.append(buffer); 7260 7261 result.append("\t\t- Input configuration:\n"); 7262 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7263 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7264 (uint32_t)mConfig.inputCfg.buffer.raw, 7265 mConfig.inputCfg.buffer.frameCount, 7266 mConfig.inputCfg.samplingRate, 7267 mConfig.inputCfg.channels, 7268 mConfig.inputCfg.format); 7269 result.append(buffer); 7270 7271 result.append("\t\t- Output configuration:\n"); 7272 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7273 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7274 (uint32_t)mConfig.outputCfg.buffer.raw, 7275 mConfig.outputCfg.buffer.frameCount, 7276 mConfig.outputCfg.samplingRate, 7277 mConfig.outputCfg.channels, 7278 mConfig.outputCfg.format); 7279 result.append(buffer); 7280 7281 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7282 result.append(buffer); 7283 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7284 for (size_t i = 0; i < mHandles.size(); ++i) { 7285 sp<EffectHandle> handle = mHandles[i].promote(); 7286 if (handle != 0) { 7287 handle->dump(buffer, SIZE); 7288 result.append(buffer); 7289 } 7290 } 7291 7292 result.append("\n"); 7293 7294 write(fd, result.string(), result.length()); 7295 7296 if (locked) { 7297 mLock.unlock(); 7298 } 7299 7300 return NO_ERROR; 7301} 7302 7303// ---------------------------------------------------------------------------- 7304// EffectHandle implementation 7305// ---------------------------------------------------------------------------- 7306 7307#undef LOG_TAG 7308#define LOG_TAG "AudioFlinger::EffectHandle" 7309 7310AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7311 const sp<AudioFlinger::Client>& client, 7312 const sp<IEffectClient>& effectClient, 7313 int32_t priority) 7314 : BnEffect(), 7315 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7316 mPriority(priority), mHasControl(false), mEnabled(false) 7317{ 7318 ALOGV("constructor %p", this); 7319 7320 if (client == 0) { 7321 return; 7322 } 7323 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7324 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7325 if (mCblkMemory != 0) { 7326 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7327 7328 if (mCblk != NULL) { 7329 new(mCblk) effect_param_cblk_t(); 7330 mBuffer = (uint8_t *)mCblk + bufOffset; 7331 } 7332 } else { 7333 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7334 return; 7335 } 7336} 7337 7338AudioFlinger::EffectHandle::~EffectHandle() 7339{ 7340 ALOGV("Destructor %p", this); 7341 disconnect(false); 7342 ALOGV("Destructor DONE %p", this); 7343} 7344 7345status_t AudioFlinger::EffectHandle::enable() 7346{ 7347 ALOGV("enable %p", this); 7348 if (!mHasControl) return INVALID_OPERATION; 7349 if (mEffect == 0) return DEAD_OBJECT; 7350 7351 if (mEnabled) { 7352 return NO_ERROR; 7353 } 7354 7355 mEnabled = true; 7356 7357 sp<ThreadBase> thread = mEffect->thread().promote(); 7358 if (thread != 0) { 7359 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7360 } 7361 7362 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7363 if (mEffect->suspended()) { 7364 return NO_ERROR; 7365 } 7366 7367 status_t status = mEffect->setEnabled(true); 7368 if (status != NO_ERROR) { 7369 if (thread != 0) { 7370 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7371 } 7372 mEnabled = false; 7373 } 7374 return status; 7375} 7376 7377status_t AudioFlinger::EffectHandle::disable() 7378{ 7379 ALOGV("disable %p", this); 7380 if (!mHasControl) return INVALID_OPERATION; 7381 if (mEffect == 0) return DEAD_OBJECT; 7382 7383 if (!mEnabled) { 7384 return NO_ERROR; 7385 } 7386 mEnabled = false; 7387 7388 if (mEffect->suspended()) { 7389 return NO_ERROR; 7390 } 7391 7392 status_t status = mEffect->setEnabled(false); 7393 7394 sp<ThreadBase> thread = mEffect->thread().promote(); 7395 if (thread != 0) { 7396 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7397 } 7398 7399 return status; 7400} 7401 7402void AudioFlinger::EffectHandle::disconnect() 7403{ 7404 disconnect(true); 7405} 7406 7407void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7408{ 7409 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7410 if (mEffect == 0) { 7411 return; 7412 } 7413 mEffect->disconnect(this, unpinIfLast); 7414 7415 if (mHasControl && mEnabled) { 7416 sp<ThreadBase> thread = mEffect->thread().promote(); 7417 if (thread != 0) { 7418 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7419 } 7420 } 7421 7422 // release sp on module => module destructor can be called now 7423 mEffect.clear(); 7424 if (mClient != 0) { 7425 if (mCblk != NULL) { 7426 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7427 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7428 } 7429 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7430 // Client destructor must run with AudioFlinger mutex locked 7431 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7432 mClient.clear(); 7433 } 7434} 7435 7436status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7437 uint32_t cmdSize, 7438 void *pCmdData, 7439 uint32_t *replySize, 7440 void *pReplyData) 7441{ 7442// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7443// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7444 7445 // only get parameter command is permitted for applications not controlling the effect 7446 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7447 return INVALID_OPERATION; 7448 } 7449 if (mEffect == 0) return DEAD_OBJECT; 7450 if (mClient == 0) return INVALID_OPERATION; 7451 7452 // handle commands that are not forwarded transparently to effect engine 7453 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7454 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7455 // no risk to block the whole media server process or mixer threads is we are stuck here 7456 Mutex::Autolock _l(mCblk->lock); 7457 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7458 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7459 mCblk->serverIndex = 0; 7460 mCblk->clientIndex = 0; 7461 return BAD_VALUE; 7462 } 7463 status_t status = NO_ERROR; 7464 while (mCblk->serverIndex < mCblk->clientIndex) { 7465 int reply; 7466 uint32_t rsize = sizeof(int); 7467 int *p = (int *)(mBuffer + mCblk->serverIndex); 7468 int size = *p++; 7469 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7470 ALOGW("command(): invalid parameter block size"); 7471 break; 7472 } 7473 effect_param_t *param = (effect_param_t *)p; 7474 if (param->psize == 0 || param->vsize == 0) { 7475 ALOGW("command(): null parameter or value size"); 7476 mCblk->serverIndex += size; 7477 continue; 7478 } 7479 uint32_t psize = sizeof(effect_param_t) + 7480 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7481 param->vsize; 7482 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7483 psize, 7484 p, 7485 &rsize, 7486 &reply); 7487 // stop at first error encountered 7488 if (ret != NO_ERROR) { 7489 status = ret; 7490 *(int *)pReplyData = reply; 7491 break; 7492 } else if (reply != NO_ERROR) { 7493 *(int *)pReplyData = reply; 7494 break; 7495 } 7496 mCblk->serverIndex += size; 7497 } 7498 mCblk->serverIndex = 0; 7499 mCblk->clientIndex = 0; 7500 return status; 7501 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7502 *(int *)pReplyData = NO_ERROR; 7503 return enable(); 7504 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7505 *(int *)pReplyData = NO_ERROR; 7506 return disable(); 7507 } 7508 7509 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7510} 7511 7512void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7513{ 7514 ALOGV("setControl %p control %d", this, hasControl); 7515 7516 mHasControl = hasControl; 7517 mEnabled = enabled; 7518 7519 if (signal && mEffectClient != 0) { 7520 mEffectClient->controlStatusChanged(hasControl); 7521 } 7522} 7523 7524void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7525 uint32_t cmdSize, 7526 void *pCmdData, 7527 uint32_t replySize, 7528 void *pReplyData) 7529{ 7530 if (mEffectClient != 0) { 7531 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7532 } 7533} 7534 7535 7536 7537void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7538{ 7539 if (mEffectClient != 0) { 7540 mEffectClient->enableStatusChanged(enabled); 7541 } 7542} 7543 7544status_t AudioFlinger::EffectHandle::onTransact( 7545 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7546{ 7547 return BnEffect::onTransact(code, data, reply, flags); 7548} 7549 7550 7551void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7552{ 7553 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7554 7555 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7556 (mClient == 0) ? getpid_cached : mClient->pid(), 7557 mPriority, 7558 mHasControl, 7559 !locked, 7560 mCblk ? mCblk->clientIndex : 0, 7561 mCblk ? mCblk->serverIndex : 0 7562 ); 7563 7564 if (locked) { 7565 mCblk->lock.unlock(); 7566 } 7567} 7568 7569#undef LOG_TAG 7570#define LOG_TAG "AudioFlinger::EffectChain" 7571 7572AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7573 int sessionId) 7574 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7575 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7576 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7577{ 7578 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7579 if (thread == NULL) { 7580 return; 7581 } 7582 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7583 thread->frameCount(); 7584} 7585 7586AudioFlinger::EffectChain::~EffectChain() 7587{ 7588 if (mOwnInBuffer) { 7589 delete mInBuffer; 7590 } 7591 7592} 7593 7594// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7595sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7596{ 7597 size_t size = mEffects.size(); 7598 7599 for (size_t i = 0; i < size; i++) { 7600 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7601 return mEffects[i]; 7602 } 7603 } 7604 return 0; 7605} 7606 7607// getEffectFromId_l() must be called with ThreadBase::mLock held 7608sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7609{ 7610 size_t size = mEffects.size(); 7611 7612 for (size_t i = 0; i < size; i++) { 7613 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7614 if (id == 0 || mEffects[i]->id() == id) { 7615 return mEffects[i]; 7616 } 7617 } 7618 return 0; 7619} 7620 7621// getEffectFromType_l() must be called with ThreadBase::mLock held 7622sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7623 const effect_uuid_t *type) 7624{ 7625 size_t size = mEffects.size(); 7626 7627 for (size_t i = 0; i < size; i++) { 7628 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7629 return mEffects[i]; 7630 } 7631 } 7632 return 0; 7633} 7634 7635// Must be called with EffectChain::mLock locked 7636void AudioFlinger::EffectChain::process_l() 7637{ 7638 sp<ThreadBase> thread = mThread.promote(); 7639 if (thread == 0) { 7640 ALOGW("process_l(): cannot promote mixer thread"); 7641 return; 7642 } 7643 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7644 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7645 // always process effects unless no more tracks are on the session and the effect tail 7646 // has been rendered 7647 bool doProcess = true; 7648 if (!isGlobalSession) { 7649 bool tracksOnSession = (trackCnt() != 0); 7650 7651 if (!tracksOnSession && mTailBufferCount == 0) { 7652 doProcess = false; 7653 } 7654 7655 if (activeTrackCnt() == 0) { 7656 // if no track is active and the effect tail has not been rendered, 7657 // the input buffer must be cleared here as the mixer process will not do it 7658 if (tracksOnSession || mTailBufferCount > 0) { 7659 size_t numSamples = thread->frameCount() * thread->channelCount(); 7660 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7661 if (mTailBufferCount > 0) { 7662 mTailBufferCount--; 7663 } 7664 } 7665 } 7666 } 7667 7668 size_t size = mEffects.size(); 7669 if (doProcess) { 7670 for (size_t i = 0; i < size; i++) { 7671 mEffects[i]->process(); 7672 } 7673 } 7674 for (size_t i = 0; i < size; i++) { 7675 mEffects[i]->updateState(); 7676 } 7677} 7678 7679// addEffect_l() must be called with PlaybackThread::mLock held 7680status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7681{ 7682 effect_descriptor_t desc = effect->desc(); 7683 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7684 7685 Mutex::Autolock _l(mLock); 7686 effect->setChain(this); 7687 sp<ThreadBase> thread = mThread.promote(); 7688 if (thread == 0) { 7689 return NO_INIT; 7690 } 7691 effect->setThread(thread); 7692 7693 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7694 // Auxiliary effects are inserted at the beginning of mEffects vector as 7695 // they are processed first and accumulated in chain input buffer 7696 mEffects.insertAt(effect, 0); 7697 7698 // the input buffer for auxiliary effect contains mono samples in 7699 // 32 bit format. This is to avoid saturation in AudoMixer 7700 // accumulation stage. Saturation is done in EffectModule::process() before 7701 // calling the process in effect engine 7702 size_t numSamples = thread->frameCount(); 7703 int32_t *buffer = new int32_t[numSamples]; 7704 memset(buffer, 0, numSamples * sizeof(int32_t)); 7705 effect->setInBuffer((int16_t *)buffer); 7706 // auxiliary effects output samples to chain input buffer for further processing 7707 // by insert effects 7708 effect->setOutBuffer(mInBuffer); 7709 } else { 7710 // Insert effects are inserted at the end of mEffects vector as they are processed 7711 // after track and auxiliary effects. 7712 // Insert effect order as a function of indicated preference: 7713 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7714 // another effect is present 7715 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7716 // last effect claiming first position 7717 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7718 // first effect claiming last position 7719 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7720 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7721 // already present 7722 7723 size_t size = mEffects.size(); 7724 size_t idx_insert = size; 7725 ssize_t idx_insert_first = -1; 7726 ssize_t idx_insert_last = -1; 7727 7728 for (size_t i = 0; i < size; i++) { 7729 effect_descriptor_t d = mEffects[i]->desc(); 7730 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7731 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7732 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7733 // check invalid effect chaining combinations 7734 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7735 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7736 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7737 return INVALID_OPERATION; 7738 } 7739 // remember position of first insert effect and by default 7740 // select this as insert position for new effect 7741 if (idx_insert == size) { 7742 idx_insert = i; 7743 } 7744 // remember position of last insert effect claiming 7745 // first position 7746 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7747 idx_insert_first = i; 7748 } 7749 // remember position of first insert effect claiming 7750 // last position 7751 if (iPref == EFFECT_FLAG_INSERT_LAST && 7752 idx_insert_last == -1) { 7753 idx_insert_last = i; 7754 } 7755 } 7756 } 7757 7758 // modify idx_insert from first position if needed 7759 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7760 if (idx_insert_last != -1) { 7761 idx_insert = idx_insert_last; 7762 } else { 7763 idx_insert = size; 7764 } 7765 } else { 7766 if (idx_insert_first != -1) { 7767 idx_insert = idx_insert_first + 1; 7768 } 7769 } 7770 7771 // always read samples from chain input buffer 7772 effect->setInBuffer(mInBuffer); 7773 7774 // if last effect in the chain, output samples to chain 7775 // output buffer, otherwise to chain input buffer 7776 if (idx_insert == size) { 7777 if (idx_insert != 0) { 7778 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7779 mEffects[idx_insert-1]->configure(); 7780 } 7781 effect->setOutBuffer(mOutBuffer); 7782 } else { 7783 effect->setOutBuffer(mInBuffer); 7784 } 7785 mEffects.insertAt(effect, idx_insert); 7786 7787 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7788 } 7789 effect->configure(); 7790 return NO_ERROR; 7791} 7792 7793// removeEffect_l() must be called with PlaybackThread::mLock held 7794size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7795{ 7796 Mutex::Autolock _l(mLock); 7797 size_t size = mEffects.size(); 7798 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7799 7800 for (size_t i = 0; i < size; i++) { 7801 if (effect == mEffects[i]) { 7802 // calling stop here will remove pre-processing effect from the audio HAL. 7803 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7804 // the middle of a read from audio HAL 7805 if (mEffects[i]->state() == EffectModule::ACTIVE || 7806 mEffects[i]->state() == EffectModule::STOPPING) { 7807 mEffects[i]->stop(); 7808 } 7809 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7810 delete[] effect->inBuffer(); 7811 } else { 7812 if (i == size - 1 && i != 0) { 7813 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7814 mEffects[i - 1]->configure(); 7815 } 7816 } 7817 mEffects.removeAt(i); 7818 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7819 break; 7820 } 7821 } 7822 7823 return mEffects.size(); 7824} 7825 7826// setDevice_l() must be called with PlaybackThread::mLock held 7827void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7828{ 7829 size_t size = mEffects.size(); 7830 for (size_t i = 0; i < size; i++) { 7831 mEffects[i]->setDevice(device); 7832 } 7833} 7834 7835// setMode_l() must be called with PlaybackThread::mLock held 7836void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7837{ 7838 size_t size = mEffects.size(); 7839 for (size_t i = 0; i < size; i++) { 7840 mEffects[i]->setMode(mode); 7841 } 7842} 7843 7844// setVolume_l() must be called with PlaybackThread::mLock held 7845bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7846{ 7847 uint32_t newLeft = *left; 7848 uint32_t newRight = *right; 7849 bool hasControl = false; 7850 int ctrlIdx = -1; 7851 size_t size = mEffects.size(); 7852 7853 // first update volume controller 7854 for (size_t i = size; i > 0; i--) { 7855 if (mEffects[i - 1]->isProcessEnabled() && 7856 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7857 ctrlIdx = i - 1; 7858 hasControl = true; 7859 break; 7860 } 7861 } 7862 7863 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7864 if (hasControl) { 7865 *left = mNewLeftVolume; 7866 *right = mNewRightVolume; 7867 } 7868 return hasControl; 7869 } 7870 7871 mVolumeCtrlIdx = ctrlIdx; 7872 mLeftVolume = newLeft; 7873 mRightVolume = newRight; 7874 7875 // second get volume update from volume controller 7876 if (ctrlIdx >= 0) { 7877 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7878 mNewLeftVolume = newLeft; 7879 mNewRightVolume = newRight; 7880 } 7881 // then indicate volume to all other effects in chain. 7882 // Pass altered volume to effects before volume controller 7883 // and requested volume to effects after controller 7884 uint32_t lVol = newLeft; 7885 uint32_t rVol = newRight; 7886 7887 for (size_t i = 0; i < size; i++) { 7888 if ((int)i == ctrlIdx) continue; 7889 // this also works for ctrlIdx == -1 when there is no volume controller 7890 if ((int)i > ctrlIdx) { 7891 lVol = *left; 7892 rVol = *right; 7893 } 7894 mEffects[i]->setVolume(&lVol, &rVol, false); 7895 } 7896 *left = newLeft; 7897 *right = newRight; 7898 7899 return hasControl; 7900} 7901 7902status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7903{ 7904 const size_t SIZE = 256; 7905 char buffer[SIZE]; 7906 String8 result; 7907 7908 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7909 result.append(buffer); 7910 7911 bool locked = tryLock(mLock); 7912 // failed to lock - AudioFlinger is probably deadlocked 7913 if (!locked) { 7914 result.append("\tCould not lock mutex:\n"); 7915 } 7916 7917 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7918 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7919 mEffects.size(), 7920 (uint32_t)mInBuffer, 7921 (uint32_t)mOutBuffer, 7922 mActiveTrackCnt); 7923 result.append(buffer); 7924 write(fd, result.string(), result.size()); 7925 7926 for (size_t i = 0; i < mEffects.size(); ++i) { 7927 sp<EffectModule> effect = mEffects[i]; 7928 if (effect != 0) { 7929 effect->dump(fd, args); 7930 } 7931 } 7932 7933 if (locked) { 7934 mLock.unlock(); 7935 } 7936 7937 return NO_ERROR; 7938} 7939 7940// must be called with ThreadBase::mLock held 7941void AudioFlinger::EffectChain::setEffectSuspended_l( 7942 const effect_uuid_t *type, bool suspend) 7943{ 7944 sp<SuspendedEffectDesc> desc; 7945 // use effect type UUID timelow as key as there is no real risk of identical 7946 // timeLow fields among effect type UUIDs. 7947 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7948 if (suspend) { 7949 if (index >= 0) { 7950 desc = mSuspendedEffects.valueAt(index); 7951 } else { 7952 desc = new SuspendedEffectDesc(); 7953 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7954 mSuspendedEffects.add(type->timeLow, desc); 7955 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7956 } 7957 if (desc->mRefCount++ == 0) { 7958 sp<EffectModule> effect = getEffectIfEnabled(type); 7959 if (effect != 0) { 7960 desc->mEffect = effect; 7961 effect->setSuspended(true); 7962 effect->setEnabled(false); 7963 } 7964 } 7965 } else { 7966 if (index < 0) { 7967 return; 7968 } 7969 desc = mSuspendedEffects.valueAt(index); 7970 if (desc->mRefCount <= 0) { 7971 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7972 desc->mRefCount = 1; 7973 } 7974 if (--desc->mRefCount == 0) { 7975 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7976 if (desc->mEffect != 0) { 7977 sp<EffectModule> effect = desc->mEffect.promote(); 7978 if (effect != 0) { 7979 effect->setSuspended(false); 7980 sp<EffectHandle> handle = effect->controlHandle(); 7981 if (handle != 0) { 7982 effect->setEnabled(handle->enabled()); 7983 } 7984 } 7985 desc->mEffect.clear(); 7986 } 7987 mSuspendedEffects.removeItemsAt(index); 7988 } 7989 } 7990} 7991 7992// must be called with ThreadBase::mLock held 7993void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7994{ 7995 sp<SuspendedEffectDesc> desc; 7996 7997 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7998 if (suspend) { 7999 if (index >= 0) { 8000 desc = mSuspendedEffects.valueAt(index); 8001 } else { 8002 desc = new SuspendedEffectDesc(); 8003 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 8004 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 8005 } 8006 if (desc->mRefCount++ == 0) { 8007 Vector< sp<EffectModule> > effects; 8008 getSuspendEligibleEffects(effects); 8009 for (size_t i = 0; i < effects.size(); i++) { 8010 setEffectSuspended_l(&effects[i]->desc().type, true); 8011 } 8012 } 8013 } else { 8014 if (index < 0) { 8015 return; 8016 } 8017 desc = mSuspendedEffects.valueAt(index); 8018 if (desc->mRefCount <= 0) { 8019 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 8020 desc->mRefCount = 1; 8021 } 8022 if (--desc->mRefCount == 0) { 8023 Vector<const effect_uuid_t *> types; 8024 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 8025 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 8026 continue; 8027 } 8028 types.add(&mSuspendedEffects.valueAt(i)->mType); 8029 } 8030 for (size_t i = 0; i < types.size(); i++) { 8031 setEffectSuspended_l(types[i], false); 8032 } 8033 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 8034 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 8035 } 8036 } 8037} 8038 8039 8040// The volume effect is used for automated tests only 8041#ifndef OPENSL_ES_H_ 8042static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 8043 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 8044const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 8045#endif //OPENSL_ES_H_ 8046 8047bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 8048{ 8049 // auxiliary effects and visualizer are never suspended on output mix 8050 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 8051 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 8052 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 8053 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 8054 return false; 8055 } 8056 return true; 8057} 8058 8059void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8060{ 8061 effects.clear(); 8062 for (size_t i = 0; i < mEffects.size(); i++) { 8063 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8064 effects.add(mEffects[i]); 8065 } 8066 } 8067} 8068 8069sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8070 const effect_uuid_t *type) 8071{ 8072 sp<EffectModule> effect = getEffectFromType_l(type); 8073 return effect != 0 && effect->isEnabled() ? effect : 0; 8074} 8075 8076void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8077 bool enabled) 8078{ 8079 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8080 if (enabled) { 8081 if (index < 0) { 8082 // if the effect is not suspend check if all effects are suspended 8083 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8084 if (index < 0) { 8085 return; 8086 } 8087 if (!isEffectEligibleForSuspend(effect->desc())) { 8088 return; 8089 } 8090 setEffectSuspended_l(&effect->desc().type, enabled); 8091 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8092 if (index < 0) { 8093 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8094 return; 8095 } 8096 } 8097 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8098 effect->desc().type.timeLow); 8099 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8100 // if effect is requested to suspended but was not yet enabled, supend it now. 8101 if (desc->mEffect == 0) { 8102 desc->mEffect = effect; 8103 effect->setEnabled(false); 8104 effect->setSuspended(true); 8105 } 8106 } else { 8107 if (index < 0) { 8108 return; 8109 } 8110 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8111 effect->desc().type.timeLow); 8112 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8113 desc->mEffect.clear(); 8114 effect->setSuspended(false); 8115 } 8116} 8117 8118#undef LOG_TAG 8119#define LOG_TAG "AudioFlinger" 8120 8121// ---------------------------------------------------------------------------- 8122 8123status_t AudioFlinger::onTransact( 8124 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8125{ 8126 return BnAudioFlinger::onTransact(code, data, reply, flags); 8127} 8128 8129}; // namespace android 8130