AudioFlinger.cpp revision 18868c5db2f90309c6d11e5837822135e4a0c0fa
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_IDLE; 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_IDLE; 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 count\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->mSessionid, r->mPid, r->mCnt); 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 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 827 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 828 // check calling permissions 829 if (!settingsAllowed()) { 830 return PERMISSION_DENIED; 831 } 832 833 // ioHandle == 0 means the parameters are global to the audio hardware interface 834 if (ioHandle == 0) { 835 status_t final_result = NO_ERROR; 836 { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs[i]; 841 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 842 final_result = result ?: final_result; 843 } 844 mHardwareStatus = AUDIO_HW_IDLE; 845 } 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 char *s; 909 { 910 AutoMutex lock(mHardwareLock); 911 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 912 audio_hw_device_t *dev = mAudioHwDevs[i]; 913 s = dev->get_parameters(dev, keys.string()); 914 mHardwareStatus = AUDIO_HW_IDLE; 915 } 916 out_s8 += String8(s ? s : ""); 917 free(s); 918 } 919 return out_s8; 920 } 921 922 Mutex::Autolock _l(mLock); 923 924 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 925 if (playbackThread != NULL) { 926 return playbackThread->getParameters(keys); 927 } 928 RecordThread *recordThread = checkRecordThread_l(ioHandle); 929 if (recordThread != NULL) { 930 return recordThread->getParameters(keys); 931 } 932 return String8(""); 933} 934 935size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 936{ 937 status_t ret = initCheck(); 938 if (ret != NO_ERROR) { 939 return 0; 940 } 941 942 AutoMutex lock(mHardwareLock); 943 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 944 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 945 mHardwareStatus = AUDIO_HW_IDLE; 946 return size; 947} 948 949unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 950{ 951 if (ioHandle == 0) { 952 return 0; 953 } 954 955 Mutex::Autolock _l(mLock); 956 957 RecordThread *recordThread = checkRecordThread_l(ioHandle); 958 if (recordThread != NULL) { 959 return recordThread->getInputFramesLost(); 960 } 961 return 0; 962} 963 964status_t AudioFlinger::setVoiceVolume(float value) 965{ 966 status_t ret = initCheck(); 967 if (ret != NO_ERROR) { 968 return ret; 969 } 970 971 // check calling permissions 972 if (!settingsAllowed()) { 973 return PERMISSION_DENIED; 974 } 975 976 AutoMutex lock(mHardwareLock); 977 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 978 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 979 mHardwareStatus = AUDIO_HW_IDLE; 980 981 return ret; 982} 983 984status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 985 audio_io_handle_t output) const 986{ 987 status_t status; 988 989 Mutex::Autolock _l(mLock); 990 991 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 992 if (playbackThread != NULL) { 993 return playbackThread->getRenderPosition(halFrames, dspFrames); 994 } 995 996 return BAD_VALUE; 997} 998 999void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1000{ 1001 1002 Mutex::Autolock _l(mLock); 1003 1004 pid_t pid = IPCThreadState::self()->getCallingPid(); 1005 if (mNotificationClients.indexOfKey(pid) < 0) { 1006 sp<NotificationClient> notificationClient = new NotificationClient(this, 1007 client, 1008 pid); 1009 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1010 1011 mNotificationClients.add(pid, notificationClient); 1012 1013 sp<IBinder> binder = client->asBinder(); 1014 binder->linkToDeath(notificationClient); 1015 1016 // the config change is always sent from playback or record threads to avoid deadlock 1017 // with AudioSystem::gLock 1018 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1019 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1020 } 1021 1022 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1023 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1024 } 1025 } 1026} 1027 1028void AudioFlinger::removeNotificationClient(pid_t pid) 1029{ 1030 Mutex::Autolock _l(mLock); 1031 1032 mNotificationClients.removeItem(pid); 1033 1034 ALOGV("%d died, releasing its sessions", pid); 1035 size_t num = mAudioSessionRefs.size(); 1036 bool removed = false; 1037 for (size_t i = 0; i< num; ) { 1038 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1039 ALOGV(" pid %d @ %d", ref->mPid, i); 1040 if (ref->mPid == pid) { 1041 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1042 mAudioSessionRefs.removeAt(i); 1043 delete ref; 1044 removed = true; 1045 num--; 1046 } else { 1047 i++; 1048 } 1049 } 1050 if (removed) { 1051 purgeStaleEffects_l(); 1052 } 1053} 1054 1055// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1056void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1057{ 1058 size_t size = mNotificationClients.size(); 1059 for (size_t i = 0; i < size; i++) { 1060 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1061 param2); 1062 } 1063} 1064 1065// removeClient_l() must be called with AudioFlinger::mLock held 1066void AudioFlinger::removeClient_l(pid_t pid) 1067{ 1068 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1069 mClients.removeItem(pid); 1070} 1071 1072 1073// ---------------------------------------------------------------------------- 1074 1075AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1076 uint32_t device, type_t type) 1077 : Thread(false), 1078 mType(type), 1079 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1080 // mChannelMask 1081 mChannelCount(0), 1082 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1083 mParamStatus(NO_ERROR), 1084 mStandby(false), mId(id), 1085 mDevice(device), 1086 mDeathRecipient(new PMDeathRecipient(this)) 1087{ 1088} 1089 1090AudioFlinger::ThreadBase::~ThreadBase() 1091{ 1092 mParamCond.broadcast(); 1093 // do not lock the mutex in destructor 1094 releaseWakeLock_l(); 1095 if (mPowerManager != 0) { 1096 sp<IBinder> binder = mPowerManager->asBinder(); 1097 binder->unlinkToDeath(mDeathRecipient); 1098 } 1099} 1100 1101void AudioFlinger::ThreadBase::exit() 1102{ 1103 ALOGV("ThreadBase::exit"); 1104 { 1105 // This lock prevents the following race in thread (uniprocessor for illustration): 1106 // if (!exitPending()) { 1107 // // context switch from here to exit() 1108 // // exit() calls requestExit(), what exitPending() observes 1109 // // exit() calls signal(), which is dropped since no waiters 1110 // // context switch back from exit() to here 1111 // mWaitWorkCV.wait(...); 1112 // // now thread is hung 1113 // } 1114 AutoMutex lock(mLock); 1115 requestExit(); 1116 mWaitWorkCV.signal(); 1117 } 1118 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1119 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1120 requestExitAndWait(); 1121} 1122 1123status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1124{ 1125 status_t status; 1126 1127 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1128 Mutex::Autolock _l(mLock); 1129 1130 mNewParameters.add(keyValuePairs); 1131 mWaitWorkCV.signal(); 1132 // wait condition with timeout in case the thread loop has exited 1133 // before the request could be processed 1134 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1135 status = mParamStatus; 1136 mWaitWorkCV.signal(); 1137 } else { 1138 status = TIMED_OUT; 1139 } 1140 return status; 1141} 1142 1143void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1144{ 1145 Mutex::Autolock _l(mLock); 1146 sendConfigEvent_l(event, param); 1147} 1148 1149// sendConfigEvent_l() must be called with ThreadBase::mLock held 1150void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1151{ 1152 ConfigEvent configEvent; 1153 configEvent.mEvent = event; 1154 configEvent.mParam = param; 1155 mConfigEvents.add(configEvent); 1156 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1157 mWaitWorkCV.signal(); 1158} 1159 1160void AudioFlinger::ThreadBase::processConfigEvents() 1161{ 1162 mLock.lock(); 1163 while(!mConfigEvents.isEmpty()) { 1164 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1165 ConfigEvent configEvent = mConfigEvents[0]; 1166 mConfigEvents.removeAt(0); 1167 // release mLock before locking AudioFlinger mLock: lock order is always 1168 // AudioFlinger then ThreadBase to avoid cross deadlock 1169 mLock.unlock(); 1170 mAudioFlinger->mLock.lock(); 1171 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1172 mAudioFlinger->mLock.unlock(); 1173 mLock.lock(); 1174 } 1175 mLock.unlock(); 1176} 1177 1178status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1179{ 1180 const size_t SIZE = 256; 1181 char buffer[SIZE]; 1182 String8 result; 1183 1184 bool locked = tryLock(mLock); 1185 if (!locked) { 1186 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1187 write(fd, buffer, strlen(buffer)); 1188 } 1189 1190 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1191 result.append(buffer); 1192 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1193 result.append(buffer); 1194 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1197 result.append(buffer); 1198 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1199 result.append(buffer); 1200 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1201 result.append(buffer); 1202 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1203 result.append(buffer); 1204 1205 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1206 result.append(buffer); 1207 result.append(" Index Command"); 1208 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1209 snprintf(buffer, SIZE, "\n %02d ", i); 1210 result.append(buffer); 1211 result.append(mNewParameters[i]); 1212 } 1213 1214 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, " Index event param\n"); 1217 result.append(buffer); 1218 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1219 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1220 result.append(buffer); 1221 } 1222 result.append("\n"); 1223 1224 write(fd, result.string(), result.size()); 1225 1226 if (locked) { 1227 mLock.unlock(); 1228 } 1229 return NO_ERROR; 1230} 1231 1232status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1233{ 1234 const size_t SIZE = 256; 1235 char buffer[SIZE]; 1236 String8 result; 1237 1238 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1239 write(fd, buffer, strlen(buffer)); 1240 1241 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1242 sp<EffectChain> chain = mEffectChains[i]; 1243 if (chain != 0) { 1244 chain->dump(fd, args); 1245 } 1246 } 1247 return NO_ERROR; 1248} 1249 1250void AudioFlinger::ThreadBase::acquireWakeLock() 1251{ 1252 Mutex::Autolock _l(mLock); 1253 acquireWakeLock_l(); 1254} 1255 1256void AudioFlinger::ThreadBase::acquireWakeLock_l() 1257{ 1258 if (mPowerManager == 0) { 1259 // use checkService() to avoid blocking if power service is not up yet 1260 sp<IBinder> binder = 1261 defaultServiceManager()->checkService(String16("power")); 1262 if (binder == 0) { 1263 ALOGW("Thread %s cannot connect to the power manager service", mName); 1264 } else { 1265 mPowerManager = interface_cast<IPowerManager>(binder); 1266 binder->linkToDeath(mDeathRecipient); 1267 } 1268 } 1269 if (mPowerManager != 0) { 1270 sp<IBinder> binder = new BBinder(); 1271 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1272 binder, 1273 String16(mName)); 1274 if (status == NO_ERROR) { 1275 mWakeLockToken = binder; 1276 } 1277 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1278 } 1279} 1280 1281void AudioFlinger::ThreadBase::releaseWakeLock() 1282{ 1283 Mutex::Autolock _l(mLock); 1284 releaseWakeLock_l(); 1285} 1286 1287void AudioFlinger::ThreadBase::releaseWakeLock_l() 1288{ 1289 if (mWakeLockToken != 0) { 1290 ALOGV("releaseWakeLock_l() %s", mName); 1291 if (mPowerManager != 0) { 1292 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1293 } 1294 mWakeLockToken.clear(); 1295 } 1296} 1297 1298void AudioFlinger::ThreadBase::clearPowerManager() 1299{ 1300 Mutex::Autolock _l(mLock); 1301 releaseWakeLock_l(); 1302 mPowerManager.clear(); 1303} 1304 1305void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1306{ 1307 sp<ThreadBase> thread = mThread.promote(); 1308 if (thread != 0) { 1309 thread->clearPowerManager(); 1310 } 1311 ALOGW("power manager service died !!!"); 1312} 1313 1314void AudioFlinger::ThreadBase::setEffectSuspended( 1315 const effect_uuid_t *type, bool suspend, int sessionId) 1316{ 1317 Mutex::Autolock _l(mLock); 1318 setEffectSuspended_l(type, suspend, sessionId); 1319} 1320 1321void AudioFlinger::ThreadBase::setEffectSuspended_l( 1322 const effect_uuid_t *type, bool suspend, int sessionId) 1323{ 1324 sp<EffectChain> chain = getEffectChain_l(sessionId); 1325 if (chain != 0) { 1326 if (type != NULL) { 1327 chain->setEffectSuspended_l(type, suspend); 1328 } else { 1329 chain->setEffectSuspendedAll_l(suspend); 1330 } 1331 } 1332 1333 updateSuspendedSessions_l(type, suspend, sessionId); 1334} 1335 1336void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1337{ 1338 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1339 if (index < 0) { 1340 return; 1341 } 1342 1343 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1344 mSuspendedSessions.editValueAt(index); 1345 1346 for (size_t i = 0; i < sessionEffects.size(); i++) { 1347 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1348 for (int j = 0; j < desc->mRefCount; j++) { 1349 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1350 chain->setEffectSuspendedAll_l(true); 1351 } else { 1352 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1353 desc->mType.timeLow); 1354 chain->setEffectSuspended_l(&desc->mType, true); 1355 } 1356 } 1357 } 1358} 1359 1360void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1361 bool suspend, 1362 int sessionId) 1363{ 1364 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1365 1366 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1367 1368 if (suspend) { 1369 if (index >= 0) { 1370 sessionEffects = mSuspendedSessions.editValueAt(index); 1371 } else { 1372 mSuspendedSessions.add(sessionId, sessionEffects); 1373 } 1374 } else { 1375 if (index < 0) { 1376 return; 1377 } 1378 sessionEffects = mSuspendedSessions.editValueAt(index); 1379 } 1380 1381 1382 int key = EffectChain::kKeyForSuspendAll; 1383 if (type != NULL) { 1384 key = type->timeLow; 1385 } 1386 index = sessionEffects.indexOfKey(key); 1387 1388 sp <SuspendedSessionDesc> desc; 1389 if (suspend) { 1390 if (index >= 0) { 1391 desc = sessionEffects.valueAt(index); 1392 } else { 1393 desc = new SuspendedSessionDesc(); 1394 if (type != NULL) { 1395 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1396 } 1397 sessionEffects.add(key, desc); 1398 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1399 } 1400 desc->mRefCount++; 1401 } else { 1402 if (index < 0) { 1403 return; 1404 } 1405 desc = sessionEffects.valueAt(index); 1406 if (--desc->mRefCount == 0) { 1407 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1408 sessionEffects.removeItemsAt(index); 1409 if (sessionEffects.isEmpty()) { 1410 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1411 sessionId); 1412 mSuspendedSessions.removeItem(sessionId); 1413 } 1414 } 1415 } 1416 if (!sessionEffects.isEmpty()) { 1417 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1418 } 1419} 1420 1421void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1422 bool enabled, 1423 int sessionId) 1424{ 1425 Mutex::Autolock _l(mLock); 1426 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1427} 1428 1429void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1430 bool enabled, 1431 int sessionId) 1432{ 1433 if (mType != RECORD) { 1434 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1435 // another session. This gives the priority to well behaved effect control panels 1436 // and applications not using global effects. 1437 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1438 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1439 } 1440 } 1441 1442 sp<EffectChain> chain = getEffectChain_l(sessionId); 1443 if (chain != 0) { 1444 chain->checkSuspendOnEffectEnabled(effect, enabled); 1445 } 1446} 1447 1448// ---------------------------------------------------------------------------- 1449 1450AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1451 AudioStreamOut* output, 1452 audio_io_handle_t id, 1453 uint32_t device, 1454 type_t type) 1455 : ThreadBase(audioFlinger, id, device, type), 1456 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1457 // Assumes constructor is called by AudioFlinger with it's mLock held, 1458 // but it would be safer to explicitly pass initial masterMute as parameter 1459 mMasterMute(audioFlinger->masterMute_l()), 1460 // mStreamTypes[] initialized in constructor body 1461 mOutput(output), 1462 // Assumes constructor is called by AudioFlinger with it's mLock held, 1463 // but it would be safer to explicitly pass initial masterVolume as parameter 1464 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1465 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1466 // mMixerStatus 1467 mPrevMixerStatus(MIXER_IDLE) 1468{ 1469 snprintf(mName, kNameLength, "AudioOut_%X", id); 1470 1471 readOutputParameters(); 1472 1473 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1474 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1475 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1476 stream = (audio_stream_type_t) (stream + 1)) { 1477 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1478 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1479 // initialized by stream_type_t default constructor 1480 // mStreamTypes[stream].valid = true; 1481 } 1482 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1483 // because mAudioFlinger doesn't have one to copy from 1484} 1485 1486AudioFlinger::PlaybackThread::~PlaybackThread() 1487{ 1488 delete [] mMixBuffer; 1489} 1490 1491status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1492{ 1493 dumpInternals(fd, args); 1494 dumpTracks(fd, args); 1495 dumpEffectChains(fd, args); 1496 return NO_ERROR; 1497} 1498 1499status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1500{ 1501 const size_t SIZE = 256; 1502 char buffer[SIZE]; 1503 String8 result; 1504 1505 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1506 result.append(buffer); 1507 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1508 for (size_t i = 0; i < mTracks.size(); ++i) { 1509 sp<Track> track = mTracks[i]; 1510 if (track != 0) { 1511 track->dump(buffer, SIZE); 1512 result.append(buffer); 1513 } 1514 } 1515 1516 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1517 result.append(buffer); 1518 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1519 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1520 sp<Track> track = mActiveTracks[i].promote(); 1521 if (track != 0) { 1522 track->dump(buffer, SIZE); 1523 result.append(buffer); 1524 } 1525 } 1526 write(fd, result.string(), result.size()); 1527 return NO_ERROR; 1528} 1529 1530status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1531{ 1532 const size_t SIZE = 256; 1533 char buffer[SIZE]; 1534 String8 result; 1535 1536 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1537 result.append(buffer); 1538 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1541 result.append(buffer); 1542 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1545 result.append(buffer); 1546 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1547 result.append(buffer); 1548 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1549 result.append(buffer); 1550 write(fd, result.string(), result.size()); 1551 1552 dumpBase(fd, args); 1553 1554 return NO_ERROR; 1555} 1556 1557// Thread virtuals 1558status_t AudioFlinger::PlaybackThread::readyToRun() 1559{ 1560 status_t status = initCheck(); 1561 if (status == NO_ERROR) { 1562 ALOGI("AudioFlinger's thread %p ready to run", this); 1563 } else { 1564 ALOGE("No working audio driver found."); 1565 } 1566 return status; 1567} 1568 1569void AudioFlinger::PlaybackThread::onFirstRef() 1570{ 1571 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1572} 1573 1574// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1575sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1576 const sp<AudioFlinger::Client>& client, 1577 audio_stream_type_t streamType, 1578 uint32_t sampleRate, 1579 audio_format_t format, 1580 uint32_t channelMask, 1581 int frameCount, 1582 const sp<IMemory>& sharedBuffer, 1583 int sessionId, 1584 bool isTimed, 1585 status_t *status) 1586{ 1587 sp<Track> track; 1588 status_t lStatus; 1589 1590 if (mType == DIRECT) { 1591 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1592 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1593 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1594 "for output %p with format %d", 1595 sampleRate, format, channelMask, mOutput, mFormat); 1596 lStatus = BAD_VALUE; 1597 goto Exit; 1598 } 1599 } 1600 } else { 1601 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1602 if (sampleRate > mSampleRate*2) { 1603 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1604 lStatus = BAD_VALUE; 1605 goto Exit; 1606 } 1607 } 1608 1609 lStatus = initCheck(); 1610 if (lStatus != NO_ERROR) { 1611 ALOGE("Audio driver not initialized."); 1612 goto Exit; 1613 } 1614 1615 { // scope for mLock 1616 Mutex::Autolock _l(mLock); 1617 1618 // all tracks in same audio session must share the same routing strategy otherwise 1619 // conflicts will happen when tracks are moved from one output to another by audio policy 1620 // manager 1621 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1622 for (size_t i = 0; i < mTracks.size(); ++i) { 1623 sp<Track> t = mTracks[i]; 1624 if (t != 0 && !t->isOutputTrack()) { 1625 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1626 if (sessionId == t->sessionId() && strategy != actual) { 1627 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1628 strategy, actual); 1629 lStatus = BAD_VALUE; 1630 goto Exit; 1631 } 1632 } 1633 } 1634 1635 if (!isTimed) { 1636 track = new Track(this, client, streamType, sampleRate, format, 1637 channelMask, frameCount, sharedBuffer, sessionId); 1638 } else { 1639 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1640 channelMask, frameCount, sharedBuffer, sessionId); 1641 } 1642 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1643 lStatus = NO_MEMORY; 1644 goto Exit; 1645 } 1646 mTracks.add(track); 1647 1648 sp<EffectChain> chain = getEffectChain_l(sessionId); 1649 if (chain != 0) { 1650 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1651 track->setMainBuffer(chain->inBuffer()); 1652 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1653 chain->incTrackCnt(); 1654 } 1655 1656 // invalidate track immediately if the stream type was moved to another thread since 1657 // createTrack() was called by the client process. 1658 if (!mStreamTypes[streamType].valid) { 1659 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1660 this, streamType); 1661 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1662 } 1663 } 1664 lStatus = NO_ERROR; 1665 1666Exit: 1667 if(status) { 1668 *status = lStatus; 1669 } 1670 return track; 1671} 1672 1673uint32_t AudioFlinger::PlaybackThread::latency() const 1674{ 1675 Mutex::Autolock _l(mLock); 1676 if (initCheck() == NO_ERROR) { 1677 return mOutput->stream->get_latency(mOutput->stream); 1678 } else { 1679 return 0; 1680 } 1681} 1682 1683void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1684{ 1685 Mutex::Autolock _l(mLock); 1686 mMasterVolume = value; 1687} 1688 1689void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1690{ 1691 Mutex::Autolock _l(mLock); 1692 setMasterMute_l(muted); 1693} 1694 1695void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1696{ 1697 Mutex::Autolock _l(mLock); 1698 mStreamTypes[stream].volume = value; 1699} 1700 1701void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1702{ 1703 Mutex::Autolock _l(mLock); 1704 mStreamTypes[stream].mute = muted; 1705} 1706 1707float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1708{ 1709 Mutex::Autolock _l(mLock); 1710 return mStreamTypes[stream].volume; 1711} 1712 1713// addTrack_l() must be called with ThreadBase::mLock held 1714status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1715{ 1716 status_t status = ALREADY_EXISTS; 1717 1718 // set retry count for buffer fill 1719 track->mRetryCount = kMaxTrackStartupRetries; 1720 if (mActiveTracks.indexOf(track) < 0) { 1721 // the track is newly added, make sure it fills up all its 1722 // buffers before playing. This is to ensure the client will 1723 // effectively get the latency it requested. 1724 track->mFillingUpStatus = Track::FS_FILLING; 1725 track->mResetDone = false; 1726 mActiveTracks.add(track); 1727 if (track->mainBuffer() != mMixBuffer) { 1728 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1729 if (chain != 0) { 1730 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1731 chain->incActiveTrackCnt(); 1732 } 1733 } 1734 1735 status = NO_ERROR; 1736 } 1737 1738 ALOGV("mWaitWorkCV.broadcast"); 1739 mWaitWorkCV.broadcast(); 1740 1741 return status; 1742} 1743 1744// destroyTrack_l() must be called with ThreadBase::mLock held 1745void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1746{ 1747 track->mState = TrackBase::TERMINATED; 1748 if (mActiveTracks.indexOf(track) < 0) { 1749 removeTrack_l(track); 1750 } 1751} 1752 1753void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1754{ 1755 mTracks.remove(track); 1756 deleteTrackName_l(track->name()); 1757 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1758 if (chain != 0) { 1759 chain->decTrackCnt(); 1760 } 1761} 1762 1763String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1764{ 1765 String8 out_s8 = String8(""); 1766 char *s; 1767 1768 Mutex::Autolock _l(mLock); 1769 if (initCheck() != NO_ERROR) { 1770 return out_s8; 1771 } 1772 1773 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1774 out_s8 = String8(s); 1775 free(s); 1776 return out_s8; 1777} 1778 1779// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1780void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1781 AudioSystem::OutputDescriptor desc; 1782 void *param2 = NULL; 1783 1784 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1785 1786 switch (event) { 1787 case AudioSystem::OUTPUT_OPENED: 1788 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1789 desc.channels = mChannelMask; 1790 desc.samplingRate = mSampleRate; 1791 desc.format = mFormat; 1792 desc.frameCount = mFrameCount; 1793 desc.latency = latency(); 1794 param2 = &desc; 1795 break; 1796 1797 case AudioSystem::STREAM_CONFIG_CHANGED: 1798 param2 = ¶m; 1799 case AudioSystem::OUTPUT_CLOSED: 1800 default: 1801 break; 1802 } 1803 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1804} 1805 1806void AudioFlinger::PlaybackThread::readOutputParameters() 1807{ 1808 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1809 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1810 mChannelCount = (uint16_t)popcount(mChannelMask); 1811 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1812 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1813 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1814 1815 // FIXME - Current mixer implementation only supports stereo output: Always 1816 // Allocate a stereo buffer even if HW output is mono. 1817 delete[] mMixBuffer; 1818 mMixBuffer = new int16_t[mFrameCount * 2]; 1819 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1820 1821 // force reconfiguration of effect chains and engines to take new buffer size and audio 1822 // parameters into account 1823 // Note that mLock is not held when readOutputParameters() is called from the constructor 1824 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1825 // matter. 1826 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1827 Vector< sp<EffectChain> > effectChains = mEffectChains; 1828 for (size_t i = 0; i < effectChains.size(); i ++) { 1829 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1830 } 1831} 1832 1833status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1834{ 1835 if (halFrames == NULL || dspFrames == NULL) { 1836 return BAD_VALUE; 1837 } 1838 Mutex::Autolock _l(mLock); 1839 if (initCheck() != NO_ERROR) { 1840 return INVALID_OPERATION; 1841 } 1842 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1843 1844 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1845} 1846 1847uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1848{ 1849 Mutex::Autolock _l(mLock); 1850 uint32_t result = 0; 1851 if (getEffectChain_l(sessionId) != 0) { 1852 result = EFFECT_SESSION; 1853 } 1854 1855 for (size_t i = 0; i < mTracks.size(); ++i) { 1856 sp<Track> track = mTracks[i]; 1857 if (sessionId == track->sessionId() && 1858 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1859 result |= TRACK_SESSION; 1860 break; 1861 } 1862 } 1863 1864 return result; 1865} 1866 1867uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1868{ 1869 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1870 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1871 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1872 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1873 } 1874 for (size_t i = 0; i < mTracks.size(); i++) { 1875 sp<Track> track = mTracks[i]; 1876 if (sessionId == track->sessionId() && 1877 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1878 return AudioSystem::getStrategyForStream(track->streamType()); 1879 } 1880 } 1881 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1882} 1883 1884 1885AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1886{ 1887 Mutex::Autolock _l(mLock); 1888 return mOutput; 1889} 1890 1891AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1892{ 1893 Mutex::Autolock _l(mLock); 1894 AudioStreamOut *output = mOutput; 1895 mOutput = NULL; 1896 return output; 1897} 1898 1899// this method must always be called either with ThreadBase mLock held or inside the thread loop 1900audio_stream_t* AudioFlinger::PlaybackThread::stream() 1901{ 1902 if (mOutput == NULL) { 1903 return NULL; 1904 } 1905 return &mOutput->stream->common; 1906} 1907 1908uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1909{ 1910 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1911 // decoding and transfer time. So sleeping for half of the latency would likely cause 1912 // underruns 1913 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1914 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1915 } else { 1916 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1917 } 1918} 1919 1920// ---------------------------------------------------------------------------- 1921 1922AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1923 audio_io_handle_t id, uint32_t device, type_t type) 1924 : PlaybackThread(audioFlinger, output, id, device, type) 1925{ 1926 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1927 // FIXME - Current mixer implementation only supports stereo output 1928 if (mChannelCount == 1) { 1929 ALOGE("Invalid audio hardware channel count"); 1930 } 1931} 1932 1933AudioFlinger::MixerThread::~MixerThread() 1934{ 1935 delete mAudioMixer; 1936} 1937 1938class CpuStats { 1939public: 1940 void sample(); 1941#ifdef DEBUG_CPU_USAGE 1942private: 1943 ThreadCpuUsage mCpu; 1944#endif 1945}; 1946 1947void CpuStats::sample() { 1948#ifdef DEBUG_CPU_USAGE 1949 const CentralTendencyStatistics& stats = mCpu.statistics(); 1950 mCpu.sampleAndEnable(); 1951 unsigned n = stats.n(); 1952 // mCpu.elapsed() is expensive, so don't call it every loop 1953 if ((n & 127) == 1) { 1954 long long elapsed = mCpu.elapsed(); 1955 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1956 double perLoop = elapsed / (double) n; 1957 double perLoop100 = perLoop * 0.01; 1958 double mean = stats.mean(); 1959 double stddev = stats.stddev(); 1960 double minimum = stats.minimum(); 1961 double maximum = stats.maximum(); 1962 mCpu.resetStatistics(); 1963 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", 1964 elapsed * .000000001, n, perLoop * .000001, 1965 mean * .001, 1966 stddev * .001, 1967 minimum * .001, 1968 maximum * .001, 1969 mean / perLoop100, 1970 stddev / perLoop100, 1971 minimum / perLoop100, 1972 maximum / perLoop100); 1973 } 1974 } 1975#endif 1976}; 1977 1978void AudioFlinger::PlaybackThread::checkSilentMode_l() 1979{ 1980 if (!mMasterMute) { 1981 char value[PROPERTY_VALUE_MAX]; 1982 if (property_get("ro.audio.silent", value, "0") > 0) { 1983 char *endptr; 1984 unsigned long ul = strtoul(value, &endptr, 0); 1985 if (*endptr == '\0' && ul != 0) { 1986 ALOGD("Silence is golden"); 1987 // The setprop command will not allow a property to be changed after 1988 // the first time it is set, so we don't have to worry about un-muting. 1989 setMasterMute_l(true); 1990 } 1991 } 1992 } 1993} 1994 1995bool AudioFlinger::PlaybackThread::threadLoop() 1996{ 1997 Vector< sp<Track> > tracksToRemove; 1998 1999 standbyTime = systemTime(); 2000 mixBufferSize = mFrameCount * mFrameSize; 2001 2002 // MIXER 2003 // FIXME: Relaxed timing because of a certain device that can't meet latency 2004 // Should be reduced to 2x after the vendor fixes the driver issue 2005 // increase threshold again due to low power audio mode. The way this warning threshold is 2006 // calculated and its usefulness should be reconsidered anyway. 2007 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2008 nsecs_t lastWarning = 0; 2009if (mType == MIXER) { 2010 longStandbyExit = false; 2011} 2012 2013 // DUPLICATING 2014 // FIXME could this be made local to while loop? 2015 writeFrames = 0; 2016 2017 activeSleepTime = activeSleepTimeUs(); 2018 idleSleepTime = idleSleepTimeUs(); 2019 sleepTime = idleSleepTime; 2020 2021if (mType == MIXER) { 2022 sleepTimeShift = 0; 2023} 2024 2025 // MIXER 2026 CpuStats cpuStats; 2027 2028 // DIRECT 2029if (mType == DIRECT) { 2030 // use shorter standby delay as on normal output to release 2031 // hardware resources as soon as possible 2032 standbyDelay = microseconds(activeSleepTime*2); 2033} 2034 2035 acquireWakeLock(); 2036 2037 while (!exitPending()) 2038 { 2039if (mType == MIXER) { 2040 cpuStats.sample(); 2041} 2042 2043 Vector< sp<EffectChain> > effectChains; 2044 2045 processConfigEvents(); 2046 2047 mMixerStatus = MIXER_IDLE; 2048 { // scope for mLock 2049 2050 Mutex::Autolock _l(mLock); 2051 2052 if (checkForNewParameters_l()) { 2053 mixBufferSize = mFrameCount * mFrameSize; 2054 2055if (mType == MIXER) { 2056 // FIXME: Relaxed timing because of a certain device that can't meet latency 2057 // Should be reduced to 2x after the vendor fixes the driver issue 2058 // increase threshold again due to low power audio mode. The way this warning 2059 // threshold is calculated and its usefulness should be reconsidered anyway. 2060 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2061} 2062 2063 updateWaitTime_l(); 2064 2065 activeSleepTime = activeSleepTimeUs(); 2066 idleSleepTime = idleSleepTimeUs(); 2067 2068if (mType == DIRECT) { 2069 standbyDelay = microseconds(activeSleepTime*2); 2070} 2071 2072 } 2073 2074 saveOutputTracks(); 2075 2076 // put audio hardware into standby after short delay 2077 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2078 mSuspended > 0)) { 2079 if (!mStandby) { 2080 2081 threadLoop_standby(); 2082 2083 mStandby = true; 2084 mBytesWritten = 0; 2085 } 2086 2087 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2088 // we're about to wait, flush the binder command buffer 2089 IPCThreadState::self()->flushCommands(); 2090 2091 clearOutputTracks(); 2092 2093 if (exitPending()) break; 2094 2095 releaseWakeLock_l(); 2096 // wait until we have something to do... 2097 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2098 mWaitWorkCV.wait(mLock); 2099 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2100 acquireWakeLock_l(); 2101 2102 mPrevMixerStatus = MIXER_IDLE; 2103 2104 checkSilentMode_l(); 2105 2106if (mType == MIXER || mType == DUPLICATING) { 2107 standbyTime = systemTime() + mStandbyTimeInNsecs; 2108} 2109 2110if (mType == DIRECT) { 2111 standbyTime = systemTime() + standbyDelay; 2112} 2113 2114 sleepTime = idleSleepTime; 2115 2116if (mType == MIXER) { 2117 sleepTimeShift = 0; 2118} 2119 2120 continue; 2121 } 2122 } 2123 2124 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2125 // Shift in the new status; this could be a queue if it's 2126 // useful to filter the mixer status over several cycles. 2127 mPrevMixerStatus = mMixerStatus; 2128 mMixerStatus = newMixerStatus; 2129 2130 // prevent any changes in effect chain list and in each effect chain 2131 // during mixing and effect process as the audio buffers could be deleted 2132 // or modified if an effect is created or deleted 2133 lockEffectChains_l(effectChains); 2134 } 2135 2136 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2137 threadLoop_mix(); 2138 } else { 2139 threadLoop_sleepTime(); 2140 } 2141 2142 if (mSuspended > 0) { 2143 sleepTime = suspendSleepTimeUs(); 2144 } 2145 2146 // only process effects if we're going to write 2147 if (sleepTime == 0) { 2148 for (size_t i = 0; i < effectChains.size(); i ++) { 2149 effectChains[i]->process_l(); 2150 } 2151 } 2152 2153 // enable changes in effect chain 2154 unlockEffectChains(effectChains); 2155 2156 // sleepTime == 0 means we must write to audio hardware 2157 if (sleepTime == 0) { 2158 2159 threadLoop_write(); 2160 2161if (mType == MIXER) { 2162 // write blocked detection 2163 nsecs_t now = systemTime(); 2164 nsecs_t delta = now - mLastWriteTime; 2165 if (!mStandby && delta > maxPeriod) { 2166 mNumDelayedWrites++; 2167 if ((now - lastWarning) > kWarningThrottleNs) { 2168 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2169 ns2ms(delta), mNumDelayedWrites, this); 2170 lastWarning = now; 2171 } 2172 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2173 // a different threshold. Or completely removed for what it is worth anyway... 2174 if (mStandby) { 2175 longStandbyExit = true; 2176 } 2177 } 2178} 2179 2180 mStandby = false; 2181 } else { 2182 usleep(sleepTime); 2183 } 2184 2185 // finally let go of removed track(s), without the lock held 2186 // since we can't guarantee the destructors won't acquire that 2187 // same lock. 2188 tracksToRemove.clear(); 2189 2190 // FIXME I don't understand the need for this here; 2191 // it was in the original code but maybe the 2192 // assignment in saveOutputTracks() makes this unnecessary? 2193 clearOutputTracks(); 2194 2195 // Effect chains will be actually deleted here if they were removed from 2196 // mEffectChains list during mixing or effects processing 2197 effectChains.clear(); 2198 2199 // FIXME Note that the above .clear() is no longer necessary since effectChains 2200 // is now local to this block, but will keep it for now (at least until merge done). 2201 } 2202 2203if (mType == MIXER || mType == DIRECT) { 2204 // put output stream into standby mode 2205 if (!mStandby) { 2206 mOutput->stream->common.standby(&mOutput->stream->common); 2207 } 2208} 2209if (mType == DUPLICATING) { 2210 // for DuplicatingThread, standby mode is handled by the outputTracks 2211} 2212 2213 releaseWakeLock(); 2214 2215 ALOGV("Thread %p type %d exiting", this, mType); 2216 return false; 2217} 2218 2219// shared by MIXER and DIRECT, overridden by DUPLICATING 2220void AudioFlinger::PlaybackThread::threadLoop_write() 2221{ 2222 // FIXME rewrite to reduce number of system calls 2223 mLastWriteTime = systemTime(); 2224 mInWrite = true; 2225 mBytesWritten += mixBufferSize; 2226 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2227 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2228 mNumWrites++; 2229 mInWrite = false; 2230} 2231 2232// shared by MIXER and DIRECT, overridden by DUPLICATING 2233void AudioFlinger::PlaybackThread::threadLoop_standby() 2234{ 2235 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2236 mOutput->stream->common.standby(&mOutput->stream->common); 2237} 2238 2239void AudioFlinger::MixerThread::threadLoop_mix() 2240{ 2241 // obtain the presentation timestamp of the next output buffer 2242 int64_t pts; 2243 status_t status = INVALID_OPERATION; 2244 2245 if (NULL != mOutput->stream->get_next_write_timestamp) { 2246 status = mOutput->stream->get_next_write_timestamp( 2247 mOutput->stream, &pts); 2248 } 2249 2250 if (status != NO_ERROR) { 2251 pts = AudioBufferProvider::kInvalidPTS; 2252 } 2253 2254 // mix buffers... 2255 mAudioMixer->process(pts); 2256 // increase sleep time progressively when application underrun condition clears. 2257 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2258 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2259 // such that we would underrun the audio HAL. 2260 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2261 sleepTimeShift--; 2262 } 2263 sleepTime = 0; 2264 standbyTime = systemTime() + mStandbyTimeInNsecs; 2265 //TODO: delay standby when effects have a tail 2266} 2267 2268void AudioFlinger::MixerThread::threadLoop_sleepTime() 2269{ 2270 // If no tracks are ready, sleep once for the duration of an output 2271 // buffer size, then write 0s to the output 2272 if (sleepTime == 0) { 2273 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2274 sleepTime = activeSleepTime >> sleepTimeShift; 2275 if (sleepTime < kMinThreadSleepTimeUs) { 2276 sleepTime = kMinThreadSleepTimeUs; 2277 } 2278 // reduce sleep time in case of consecutive application underruns to avoid 2279 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2280 // duration we would end up writing less data than needed by the audio HAL if 2281 // the condition persists. 2282 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2283 sleepTimeShift++; 2284 } 2285 } else { 2286 sleepTime = idleSleepTime; 2287 } 2288 } else if (mBytesWritten != 0 || 2289 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2290 memset (mMixBuffer, 0, mixBufferSize); 2291 sleepTime = 0; 2292 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2293 } 2294 // TODO add standby time extension fct of effect tail 2295} 2296 2297// prepareTracks_l() must be called with ThreadBase::mLock held 2298AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2299 Vector< sp<Track> > *tracksToRemove) 2300{ 2301 2302 mixer_state mixerStatus = MIXER_IDLE; 2303 // find out which tracks need to be processed 2304 size_t count = mActiveTracks.size(); 2305 size_t mixedTracks = 0; 2306 size_t tracksWithEffect = 0; 2307 2308 float masterVolume = mMasterVolume; 2309 bool masterMute = mMasterMute; 2310 2311 if (masterMute) { 2312 masterVolume = 0; 2313 } 2314 // Delegate master volume control to effect in output mix effect chain if needed 2315 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2316 if (chain != 0) { 2317 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2318 chain->setVolume_l(&v, &v); 2319 masterVolume = (float)((v + (1 << 23)) >> 24); 2320 chain.clear(); 2321 } 2322 2323 for (size_t i=0 ; i<count ; i++) { 2324 sp<Track> t = mActiveTracks[i].promote(); 2325 if (t == 0) continue; 2326 2327 // this const just means the local variable doesn't change 2328 Track* const track = t.get(); 2329 audio_track_cblk_t* cblk = track->cblk(); 2330 2331 // The first time a track is added we wait 2332 // for all its buffers to be filled before processing it 2333 int name = track->name(); 2334 // make sure that we have enough frames to mix one full buffer. 2335 // enforce this condition only once to enable draining the buffer in case the client 2336 // app does not call stop() and relies on underrun to stop: 2337 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2338 // during last round 2339 uint32_t minFrames = 1; 2340 if (!track->isStopped() && !track->isPausing() && 2341 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2342 if (t->sampleRate() == (int)mSampleRate) { 2343 minFrames = mFrameCount; 2344 } else { 2345 // +1 for rounding and +1 for additional sample needed for interpolation 2346 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2347 // add frames already consumed but not yet released by the resampler 2348 // because cblk->framesReady() will include these frames 2349 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2350 // the minimum track buffer size is normally twice the number of frames necessary 2351 // to fill one buffer and the resampler should not leave more than one buffer worth 2352 // of unreleased frames after each pass, but just in case... 2353 ALOG_ASSERT(minFrames <= cblk->frameCount); 2354 } 2355 } 2356 if ((track->framesReady() >= minFrames) && track->isReady() && 2357 !track->isPaused() && !track->isTerminated()) 2358 { 2359 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2360 2361 mixedTracks++; 2362 2363 // track->mainBuffer() != mMixBuffer means there is an effect chain 2364 // connected to the track 2365 chain.clear(); 2366 if (track->mainBuffer() != mMixBuffer) { 2367 chain = getEffectChain_l(track->sessionId()); 2368 // Delegate volume control to effect in track effect chain if needed 2369 if (chain != 0) { 2370 tracksWithEffect++; 2371 } else { 2372 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2373 name, track->sessionId()); 2374 } 2375 } 2376 2377 2378 int param = AudioMixer::VOLUME; 2379 if (track->mFillingUpStatus == Track::FS_FILLED) { 2380 // no ramp for the first volume setting 2381 track->mFillingUpStatus = Track::FS_ACTIVE; 2382 if (track->mState == TrackBase::RESUMING) { 2383 track->mState = TrackBase::ACTIVE; 2384 param = AudioMixer::RAMP_VOLUME; 2385 } 2386 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2387 } else if (cblk->server != 0) { 2388 // If the track is stopped before the first frame was mixed, 2389 // do not apply ramp 2390 param = AudioMixer::RAMP_VOLUME; 2391 } 2392 2393 // compute volume for this track 2394 uint32_t vl, vr, va; 2395 if (track->isMuted() || track->isPausing() || 2396 mStreamTypes[track->streamType()].mute) { 2397 vl = vr = va = 0; 2398 if (track->isPausing()) { 2399 track->setPaused(); 2400 } 2401 } else { 2402 2403 // read original volumes with volume control 2404 float typeVolume = mStreamTypes[track->streamType()].volume; 2405 float v = masterVolume * typeVolume; 2406 uint32_t vlr = cblk->getVolumeLR(); 2407 vl = vlr & 0xFFFF; 2408 vr = vlr >> 16; 2409 // track volumes come from shared memory, so can't be trusted and must be clamped 2410 if (vl > MAX_GAIN_INT) { 2411 ALOGV("Track left volume out of range: %04X", vl); 2412 vl = MAX_GAIN_INT; 2413 } 2414 if (vr > MAX_GAIN_INT) { 2415 ALOGV("Track right volume out of range: %04X", vr); 2416 vr = MAX_GAIN_INT; 2417 } 2418 // now apply the master volume and stream type volume 2419 vl = (uint32_t)(v * vl) << 12; 2420 vr = (uint32_t)(v * vr) << 12; 2421 // assuming master volume and stream type volume each go up to 1.0, 2422 // vl and vr are now in 8.24 format 2423 2424 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2425 // send level comes from shared memory and so may be corrupt 2426 if (sendLevel > MAX_GAIN_INT) { 2427 ALOGV("Track send level out of range: %04X", sendLevel); 2428 sendLevel = MAX_GAIN_INT; 2429 } 2430 va = (uint32_t)(v * sendLevel); 2431 } 2432 // Delegate volume control to effect in track effect chain if needed 2433 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2434 // Do not ramp volume if volume is controlled by effect 2435 param = AudioMixer::VOLUME; 2436 track->mHasVolumeController = true; 2437 } else { 2438 // force no volume ramp when volume controller was just disabled or removed 2439 // from effect chain to avoid volume spike 2440 if (track->mHasVolumeController) { 2441 param = AudioMixer::VOLUME; 2442 } 2443 track->mHasVolumeController = false; 2444 } 2445 2446 // Convert volumes from 8.24 to 4.12 format 2447 // This additional clamping is needed in case chain->setVolume_l() overshot 2448 vl = (vl + (1 << 11)) >> 12; 2449 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2450 vr = (vr + (1 << 11)) >> 12; 2451 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2452 2453 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2454 2455 // XXX: these things DON'T need to be done each time 2456 mAudioMixer->setBufferProvider(name, track); 2457 mAudioMixer->enable(name); 2458 2459 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2460 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2461 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2462 mAudioMixer->setParameter( 2463 name, 2464 AudioMixer::TRACK, 2465 AudioMixer::FORMAT, (void *)track->format()); 2466 mAudioMixer->setParameter( 2467 name, 2468 AudioMixer::TRACK, 2469 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2470 mAudioMixer->setParameter( 2471 name, 2472 AudioMixer::RESAMPLE, 2473 AudioMixer::SAMPLE_RATE, 2474 (void *)(cblk->sampleRate)); 2475 mAudioMixer->setParameter( 2476 name, 2477 AudioMixer::TRACK, 2478 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2479 mAudioMixer->setParameter( 2480 name, 2481 AudioMixer::TRACK, 2482 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2483 2484 // reset retry count 2485 track->mRetryCount = kMaxTrackRetries; 2486 // If one track is ready, set the mixer ready if: 2487 // - the mixer was not ready during previous round OR 2488 // - no other track is not ready 2489 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2490 mixerStatus != MIXER_TRACKS_ENABLED) { 2491 mixerStatus = MIXER_TRACKS_READY; 2492 } 2493 } else { 2494 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2495 if (track->isStopped()) { 2496 track->reset(); 2497 } 2498 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2499 // We have consumed all the buffers of this track. 2500 // Remove it from the list of active tracks. 2501 tracksToRemove->add(track); 2502 } else { 2503 // No buffers for this track. Give it a few chances to 2504 // fill a buffer, then remove it from active list. 2505 if (--(track->mRetryCount) <= 0) { 2506 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2507 tracksToRemove->add(track); 2508 // indicate to client process that the track was disabled because of underrun 2509 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2510 // If one track is not ready, mark the mixer also not ready if: 2511 // - the mixer was ready during previous round OR 2512 // - no other track is ready 2513 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2514 mixerStatus != MIXER_TRACKS_READY) { 2515 mixerStatus = MIXER_TRACKS_ENABLED; 2516 } 2517 } 2518 mAudioMixer->disable(name); 2519 } 2520 } 2521 2522 // remove all the tracks that need to be... 2523 count = tracksToRemove->size(); 2524 if (CC_UNLIKELY(count)) { 2525 for (size_t i=0 ; i<count ; i++) { 2526 const sp<Track>& track = tracksToRemove->itemAt(i); 2527 mActiveTracks.remove(track); 2528 if (track->mainBuffer() != mMixBuffer) { 2529 chain = getEffectChain_l(track->sessionId()); 2530 if (chain != 0) { 2531 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2532 chain->decActiveTrackCnt(); 2533 } 2534 } 2535 if (track->isTerminated()) { 2536 removeTrack_l(track); 2537 } 2538 } 2539 } 2540 2541 // mix buffer must be cleared if all tracks are connected to an 2542 // effect chain as in this case the mixer will not write to 2543 // mix buffer and track effects will accumulate into it 2544 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2545 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2546 } 2547 2548 return mixerStatus; 2549} 2550 2551void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2552{ 2553 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2554 this, streamType, mTracks.size()); 2555 Mutex::Autolock _l(mLock); 2556 2557 size_t size = mTracks.size(); 2558 for (size_t i = 0; i < size; i++) { 2559 sp<Track> t = mTracks[i]; 2560 if (t->streamType() == streamType) { 2561 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2562 t->mCblk->cv.signal(); 2563 } 2564 } 2565} 2566 2567void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2568{ 2569 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2570 this, streamType, valid); 2571 Mutex::Autolock _l(mLock); 2572 2573 mStreamTypes[streamType].valid = valid; 2574} 2575 2576// getTrackName_l() must be called with ThreadBase::mLock held 2577int AudioFlinger::MixerThread::getTrackName_l() 2578{ 2579 return mAudioMixer->getTrackName(); 2580} 2581 2582// deleteTrackName_l() must be called with ThreadBase::mLock held 2583void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2584{ 2585 ALOGV("remove track (%d) and delete from mixer", name); 2586 mAudioMixer->deleteTrackName(name); 2587} 2588 2589// checkForNewParameters_l() must be called with ThreadBase::mLock held 2590bool AudioFlinger::MixerThread::checkForNewParameters_l() 2591{ 2592 bool reconfig = false; 2593 2594 while (!mNewParameters.isEmpty()) { 2595 status_t status = NO_ERROR; 2596 String8 keyValuePair = mNewParameters[0]; 2597 AudioParameter param = AudioParameter(keyValuePair); 2598 int value; 2599 2600 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2601 reconfig = true; 2602 } 2603 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2604 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2605 status = BAD_VALUE; 2606 } else { 2607 reconfig = true; 2608 } 2609 } 2610 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2611 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2612 status = BAD_VALUE; 2613 } else { 2614 reconfig = true; 2615 } 2616 } 2617 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2618 // do not accept frame count changes if tracks are open as the track buffer 2619 // size depends on frame count and correct behavior would not be guaranteed 2620 // if frame count is changed after track creation 2621 if (!mTracks.isEmpty()) { 2622 status = INVALID_OPERATION; 2623 } else { 2624 reconfig = true; 2625 } 2626 } 2627 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2628 // when changing the audio output device, call addBatteryData to notify 2629 // the change 2630 if ((int)mDevice != value) { 2631 uint32_t params = 0; 2632 // check whether speaker is on 2633 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2634 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2635 } 2636 2637 int deviceWithoutSpeaker 2638 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2639 // check if any other device (except speaker) is on 2640 if (value & deviceWithoutSpeaker ) { 2641 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2642 } 2643 2644 if (params != 0) { 2645 addBatteryData(params); 2646 } 2647 } 2648 2649 // forward device change to effects that have requested to be 2650 // aware of attached audio device. 2651 mDevice = (uint32_t)value; 2652 for (size_t i = 0; i < mEffectChains.size(); i++) { 2653 mEffectChains[i]->setDevice_l(mDevice); 2654 } 2655 } 2656 2657 if (status == NO_ERROR) { 2658 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2659 keyValuePair.string()); 2660 if (!mStandby && status == INVALID_OPERATION) { 2661 mOutput->stream->common.standby(&mOutput->stream->common); 2662 mStandby = true; 2663 mBytesWritten = 0; 2664 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2665 keyValuePair.string()); 2666 } 2667 if (status == NO_ERROR && reconfig) { 2668 delete mAudioMixer; 2669 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2670 mAudioMixer = NULL; 2671 readOutputParameters(); 2672 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2673 for (size_t i = 0; i < mTracks.size() ; i++) { 2674 int name = getTrackName_l(); 2675 if (name < 0) break; 2676 mTracks[i]->mName = name; 2677 // limit track sample rate to 2 x new output sample rate 2678 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2679 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2680 } 2681 } 2682 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2683 } 2684 } 2685 2686 mNewParameters.removeAt(0); 2687 2688 mParamStatus = status; 2689 mParamCond.signal(); 2690 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2691 // already timed out waiting for the status and will never signal the condition. 2692 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2693 } 2694 return reconfig; 2695} 2696 2697status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2698{ 2699 const size_t SIZE = 256; 2700 char buffer[SIZE]; 2701 String8 result; 2702 2703 PlaybackThread::dumpInternals(fd, args); 2704 2705 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2706 result.append(buffer); 2707 write(fd, result.string(), result.size()); 2708 return NO_ERROR; 2709} 2710 2711uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2712{ 2713 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2714} 2715 2716uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2717{ 2718 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2719} 2720 2721// ---------------------------------------------------------------------------- 2722AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2723 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2724 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2725 // mLeftVolFloat, mRightVolFloat 2726 // mLeftVolShort, mRightVolShort 2727{ 2728} 2729 2730AudioFlinger::DirectOutputThread::~DirectOutputThread() 2731{ 2732} 2733 2734void AudioFlinger::DirectOutputThread::applyVolume() 2735{ 2736 // Do not apply volume on compressed audio 2737 if (!audio_is_linear_pcm(mFormat)) { 2738 return; 2739 } 2740 2741 // convert to signed 16 bit before volume calculation 2742 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2743 size_t count = mFrameCount * mChannelCount; 2744 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2745 int16_t *dst = mMixBuffer + count-1; 2746 while(count--) { 2747 *dst-- = (int16_t)(*src--^0x80) << 8; 2748 } 2749 } 2750 2751 size_t frameCount = mFrameCount; 2752 int16_t *out = mMixBuffer; 2753 if (rampVolume) { 2754 if (mChannelCount == 1) { 2755 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2756 int32_t vlInc = d / (int32_t)frameCount; 2757 int32_t vl = ((int32_t)mLeftVolShort << 16); 2758 do { 2759 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2760 out++; 2761 vl += vlInc; 2762 } while (--frameCount); 2763 2764 } else { 2765 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2766 int32_t vlInc = d / (int32_t)frameCount; 2767 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2768 int32_t vrInc = d / (int32_t)frameCount; 2769 int32_t vl = ((int32_t)mLeftVolShort << 16); 2770 int32_t vr = ((int32_t)mRightVolShort << 16); 2771 do { 2772 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2773 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2774 out += 2; 2775 vl += vlInc; 2776 vr += vrInc; 2777 } while (--frameCount); 2778 } 2779 } else { 2780 if (mChannelCount == 1) { 2781 do { 2782 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2783 out++; 2784 } while (--frameCount); 2785 } else { 2786 do { 2787 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2788 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2789 out += 2; 2790 } while (--frameCount); 2791 } 2792 } 2793 2794 // convert back to unsigned 8 bit after volume calculation 2795 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2796 size_t count = mFrameCount * mChannelCount; 2797 int16_t *src = mMixBuffer; 2798 uint8_t *dst = (uint8_t *)mMixBuffer; 2799 while(count--) { 2800 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2801 } 2802 } 2803 2804 mLeftVolShort = leftVol; 2805 mRightVolShort = rightVol; 2806} 2807 2808AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2809 Vector< sp<Track> > *tracksToRemove 2810) 2811{ 2812 sp<Track> trackToRemove; 2813 2814 mixer_state mixerStatus = MIXER_IDLE; 2815 2816 // find out which tracks need to be processed 2817 if (mActiveTracks.size() != 0) { 2818 sp<Track> t = mActiveTracks[0].promote(); 2819 // The track died recently 2820 if (t == 0) return MIXER_IDLE; 2821 2822 Track* const track = t.get(); 2823 audio_track_cblk_t* cblk = track->cblk(); 2824 2825 // The first time a track is added we wait 2826 // for all its buffers to be filled before processing it 2827 if (cblk->framesReady() && track->isReady() && 2828 !track->isPaused() && !track->isTerminated()) 2829 { 2830 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2831 2832 if (track->mFillingUpStatus == Track::FS_FILLED) { 2833 track->mFillingUpStatus = Track::FS_ACTIVE; 2834 mLeftVolFloat = mRightVolFloat = 0; 2835 mLeftVolShort = mRightVolShort = 0; 2836 if (track->mState == TrackBase::RESUMING) { 2837 track->mState = TrackBase::ACTIVE; 2838 rampVolume = true; 2839 } 2840 } else if (cblk->server != 0) { 2841 // If the track is stopped before the first frame was mixed, 2842 // do not apply ramp 2843 rampVolume = true; 2844 } 2845 // compute volume for this track 2846 float left, right; 2847 if (track->isMuted() || mMasterMute || track->isPausing() || 2848 mStreamTypes[track->streamType()].mute) { 2849 left = right = 0; 2850 if (track->isPausing()) { 2851 track->setPaused(); 2852 } 2853 } else { 2854 float typeVolume = mStreamTypes[track->streamType()].volume; 2855 float v = mMasterVolume * typeVolume; 2856 uint32_t vlr = cblk->getVolumeLR(); 2857 float v_clamped = v * (vlr & 0xFFFF); 2858 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2859 left = v_clamped/MAX_GAIN; 2860 v_clamped = v * (vlr >> 16); 2861 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2862 right = v_clamped/MAX_GAIN; 2863 } 2864 2865 if (left != mLeftVolFloat || right != mRightVolFloat) { 2866 mLeftVolFloat = left; 2867 mRightVolFloat = right; 2868 2869 // If audio HAL implements volume control, 2870 // force software volume to nominal value 2871 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2872 left = 1.0f; 2873 right = 1.0f; 2874 } 2875 2876 // Convert volumes from float to 8.24 2877 uint32_t vl = (uint32_t)(left * (1 << 24)); 2878 uint32_t vr = (uint32_t)(right * (1 << 24)); 2879 2880 // Delegate volume control to effect in track effect chain if needed 2881 // only one effect chain can be present on DirectOutputThread, so if 2882 // there is one, the track is connected to it 2883 if (!mEffectChains.isEmpty()) { 2884 // Do not ramp volume if volume is controlled by effect 2885 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2886 rampVolume = false; 2887 } 2888 } 2889 2890 // Convert volumes from 8.24 to 4.12 format 2891 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2892 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2893 leftVol = (uint16_t)v_clamped; 2894 v_clamped = (vr + (1 << 11)) >> 12; 2895 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2896 rightVol = (uint16_t)v_clamped; 2897 } else { 2898 leftVol = mLeftVolShort; 2899 rightVol = mRightVolShort; 2900 rampVolume = false; 2901 } 2902 2903 // reset retry count 2904 track->mRetryCount = kMaxTrackRetriesDirect; 2905 mActiveTrack = t; 2906 mixerStatus = MIXER_TRACKS_READY; 2907 } else { 2908 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2909 if (track->isStopped()) { 2910 track->reset(); 2911 } 2912 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2913 // We have consumed all the buffers of this track. 2914 // Remove it from the list of active tracks. 2915 trackToRemove = track; 2916 } else { 2917 // No buffers for this track. Give it a few chances to 2918 // fill a buffer, then remove it from active list. 2919 if (--(track->mRetryCount) <= 0) { 2920 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2921 trackToRemove = track; 2922 } else { 2923 mixerStatus = MIXER_TRACKS_ENABLED; 2924 } 2925 } 2926 } 2927 } 2928 2929 // FIXME merge this with similar code for removing multiple tracks 2930 // remove all the tracks that need to be... 2931 if (CC_UNLIKELY(trackToRemove != 0)) { 2932 tracksToRemove->add(trackToRemove); 2933 mActiveTracks.remove(trackToRemove); 2934 if (!mEffectChains.isEmpty()) { 2935 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2936 trackToRemove->sessionId()); 2937 mEffectChains[0]->decActiveTrackCnt(); 2938 } 2939 if (trackToRemove->isTerminated()) { 2940 removeTrack_l(trackToRemove); 2941 } 2942 } 2943 2944 return mixerStatus; 2945} 2946 2947void AudioFlinger::DirectOutputThread::threadLoop_mix() 2948{ 2949 AudioBufferProvider::Buffer buffer; 2950 size_t frameCount = mFrameCount; 2951 int8_t *curBuf = (int8_t *)mMixBuffer; 2952 // output audio to hardware 2953 while (frameCount) { 2954 buffer.frameCount = frameCount; 2955 mActiveTrack->getNextBuffer(&buffer); 2956 if (CC_UNLIKELY(buffer.raw == NULL)) { 2957 memset(curBuf, 0, frameCount * mFrameSize); 2958 break; 2959 } 2960 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2961 frameCount -= buffer.frameCount; 2962 curBuf += buffer.frameCount * mFrameSize; 2963 mActiveTrack->releaseBuffer(&buffer); 2964 } 2965 sleepTime = 0; 2966 standbyTime = systemTime() + standbyDelay; 2967 mActiveTrack.clear(); 2968 applyVolume(); 2969} 2970 2971void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 2972{ 2973 if (sleepTime == 0) { 2974 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2975 sleepTime = activeSleepTime; 2976 } else { 2977 sleepTime = idleSleepTime; 2978 } 2979 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2980 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2981 sleepTime = 0; 2982 } 2983} 2984 2985// getTrackName_l() must be called with ThreadBase::mLock held 2986int AudioFlinger::DirectOutputThread::getTrackName_l() 2987{ 2988 return 0; 2989} 2990 2991// deleteTrackName_l() must be called with ThreadBase::mLock held 2992void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2993{ 2994} 2995 2996// checkForNewParameters_l() must be called with ThreadBase::mLock held 2997bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 2998{ 2999 bool reconfig = false; 3000 3001 while (!mNewParameters.isEmpty()) { 3002 status_t status = NO_ERROR; 3003 String8 keyValuePair = mNewParameters[0]; 3004 AudioParameter param = AudioParameter(keyValuePair); 3005 int value; 3006 3007 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3008 // do not accept frame count changes if tracks are open as the track buffer 3009 // size depends on frame count and correct behavior would not be garantied 3010 // if frame count is changed after track creation 3011 if (!mTracks.isEmpty()) { 3012 status = INVALID_OPERATION; 3013 } else { 3014 reconfig = true; 3015 } 3016 } 3017 if (status == NO_ERROR) { 3018 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3019 keyValuePair.string()); 3020 if (!mStandby && status == INVALID_OPERATION) { 3021 mOutput->stream->common.standby(&mOutput->stream->common); 3022 mStandby = true; 3023 mBytesWritten = 0; 3024 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3025 keyValuePair.string()); 3026 } 3027 if (status == NO_ERROR && reconfig) { 3028 readOutputParameters(); 3029 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3030 } 3031 } 3032 3033 mNewParameters.removeAt(0); 3034 3035 mParamStatus = status; 3036 mParamCond.signal(); 3037 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3038 // already timed out waiting for the status and will never signal the condition. 3039 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3040 } 3041 return reconfig; 3042} 3043 3044uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3045{ 3046 uint32_t time; 3047 if (audio_is_linear_pcm(mFormat)) { 3048 time = PlaybackThread::activeSleepTimeUs(); 3049 } else { 3050 time = 10000; 3051 } 3052 return time; 3053} 3054 3055uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3056{ 3057 uint32_t time; 3058 if (audio_is_linear_pcm(mFormat)) { 3059 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3060 } else { 3061 time = 10000; 3062 } 3063 return time; 3064} 3065 3066uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3067{ 3068 uint32_t time; 3069 if (audio_is_linear_pcm(mFormat)) { 3070 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3071 } else { 3072 time = 10000; 3073 } 3074 return time; 3075} 3076 3077 3078// ---------------------------------------------------------------------------- 3079 3080AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3081 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3082 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3083 mWaitTimeMs(UINT_MAX) 3084{ 3085 addOutputTrack(mainThread); 3086} 3087 3088AudioFlinger::DuplicatingThread::~DuplicatingThread() 3089{ 3090 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3091 mOutputTracks[i]->destroy(); 3092 } 3093} 3094 3095void AudioFlinger::DuplicatingThread::threadLoop_mix() 3096{ 3097 // mix buffers... 3098 if (outputsReady(outputTracks)) { 3099 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3100 } else { 3101 memset(mMixBuffer, 0, mixBufferSize); 3102 } 3103 sleepTime = 0; 3104 writeFrames = mFrameCount; 3105} 3106 3107void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3108{ 3109 if (sleepTime == 0) { 3110 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3111 sleepTime = activeSleepTime; 3112 } else { 3113 sleepTime = idleSleepTime; 3114 } 3115 } else if (mBytesWritten != 0) { 3116 // flush remaining overflow buffers in output tracks 3117 for (size_t i = 0; i < outputTracks.size(); i++) { 3118 if (outputTracks[i]->isActive()) { 3119 sleepTime = 0; 3120 writeFrames = 0; 3121 memset(mMixBuffer, 0, mixBufferSize); 3122 break; 3123 } 3124 } 3125 } 3126} 3127 3128void AudioFlinger::DuplicatingThread::threadLoop_write() 3129{ 3130 standbyTime = systemTime() + mStandbyTimeInNsecs; 3131 for (size_t i = 0; i < outputTracks.size(); i++) { 3132 outputTracks[i]->write(mMixBuffer, writeFrames); 3133 } 3134 mBytesWritten += mixBufferSize; 3135} 3136 3137void AudioFlinger::DuplicatingThread::threadLoop_standby() 3138{ 3139 // DuplicatingThread implements standby by stopping all tracks 3140 for (size_t i = 0; i < outputTracks.size(); i++) { 3141 outputTracks[i]->stop(); 3142 } 3143} 3144 3145void AudioFlinger::DuplicatingThread::saveOutputTracks() 3146{ 3147 outputTracks = mOutputTracks; 3148} 3149 3150void AudioFlinger::DuplicatingThread::clearOutputTracks() 3151{ 3152 outputTracks.clear(); 3153} 3154 3155void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3156{ 3157 Mutex::Autolock _l(mLock); 3158 // FIXME explain this formula 3159 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3160 OutputTrack *outputTrack = new OutputTrack(thread, 3161 this, 3162 mSampleRate, 3163 mFormat, 3164 mChannelMask, 3165 frameCount); 3166 if (outputTrack->cblk() != NULL) { 3167 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3168 mOutputTracks.add(outputTrack); 3169 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3170 updateWaitTime_l(); 3171 } 3172} 3173 3174void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3175{ 3176 Mutex::Autolock _l(mLock); 3177 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3178 if (mOutputTracks[i]->thread() == thread) { 3179 mOutputTracks[i]->destroy(); 3180 mOutputTracks.removeAt(i); 3181 updateWaitTime_l(); 3182 return; 3183 } 3184 } 3185 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3186} 3187 3188// caller must hold mLock 3189void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3190{ 3191 mWaitTimeMs = UINT_MAX; 3192 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3193 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3194 if (strong != 0) { 3195 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3196 if (waitTimeMs < mWaitTimeMs) { 3197 mWaitTimeMs = waitTimeMs; 3198 } 3199 } 3200 } 3201} 3202 3203 3204bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3205{ 3206 for (size_t i = 0; i < outputTracks.size(); i++) { 3207 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3208 if (thread == 0) { 3209 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3210 return false; 3211 } 3212 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3213 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3214 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3215 return false; 3216 } 3217 } 3218 return true; 3219} 3220 3221uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3222{ 3223 return (mWaitTimeMs * 1000) / 2; 3224} 3225 3226// ---------------------------------------------------------------------------- 3227 3228// TrackBase constructor must be called with AudioFlinger::mLock held 3229AudioFlinger::ThreadBase::TrackBase::TrackBase( 3230 ThreadBase *thread, 3231 const sp<Client>& client, 3232 uint32_t sampleRate, 3233 audio_format_t format, 3234 uint32_t channelMask, 3235 int frameCount, 3236 const sp<IMemory>& sharedBuffer, 3237 int sessionId) 3238 : RefBase(), 3239 mThread(thread), 3240 mClient(client), 3241 mCblk(NULL), 3242 // mBuffer 3243 // mBufferEnd 3244 mFrameCount(0), 3245 mState(IDLE), 3246 mFormat(format), 3247 mStepServerFailed(false), 3248 mSessionId(sessionId) 3249 // mChannelCount 3250 // mChannelMask 3251{ 3252 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3253 3254 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3255 size_t size = sizeof(audio_track_cblk_t); 3256 uint8_t channelCount = popcount(channelMask); 3257 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3258 if (sharedBuffer == 0) { 3259 size += bufferSize; 3260 } 3261 3262 if (client != NULL) { 3263 mCblkMemory = client->heap()->allocate(size); 3264 if (mCblkMemory != 0) { 3265 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3266 if (mCblk != NULL) { // construct the shared structure in-place. 3267 new(mCblk) audio_track_cblk_t(); 3268 // clear all buffers 3269 mCblk->frameCount = frameCount; 3270 mCblk->sampleRate = sampleRate; 3271 mChannelCount = channelCount; 3272 mChannelMask = channelMask; 3273 if (sharedBuffer == 0) { 3274 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3275 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3276 // Force underrun condition to avoid false underrun callback until first data is 3277 // written to buffer (other flags are cleared) 3278 mCblk->flags = CBLK_UNDERRUN_ON; 3279 } else { 3280 mBuffer = sharedBuffer->pointer(); 3281 } 3282 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3283 } 3284 } else { 3285 ALOGE("not enough memory for AudioTrack size=%u", size); 3286 client->heap()->dump("AudioTrack"); 3287 return; 3288 } 3289 } else { 3290 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3291 // construct the shared structure in-place. 3292 new(mCblk) audio_track_cblk_t(); 3293 // clear all buffers 3294 mCblk->frameCount = frameCount; 3295 mCblk->sampleRate = sampleRate; 3296 mChannelCount = channelCount; 3297 mChannelMask = channelMask; 3298 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3299 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3300 // Force underrun condition to avoid false underrun callback until first data is 3301 // written to buffer (other flags are cleared) 3302 mCblk->flags = CBLK_UNDERRUN_ON; 3303 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3304 } 3305} 3306 3307AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3308{ 3309 if (mCblk != NULL) { 3310 if (mClient == 0) { 3311 delete mCblk; 3312 } else { 3313 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3314 } 3315 } 3316 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3317 if (mClient != 0) { 3318 // Client destructor must run with AudioFlinger mutex locked 3319 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3320 // If the client's reference count drops to zero, the associated destructor 3321 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3322 // relying on the automatic clear() at end of scope. 3323 mClient.clear(); 3324 } 3325} 3326 3327// AudioBufferProvider interface 3328// getNextBuffer() = 0; 3329// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3330void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3331{ 3332 buffer->raw = NULL; 3333 mFrameCount = buffer->frameCount; 3334 (void) step(); // ignore return value of step() 3335 buffer->frameCount = 0; 3336} 3337 3338bool AudioFlinger::ThreadBase::TrackBase::step() { 3339 bool result; 3340 audio_track_cblk_t* cblk = this->cblk(); 3341 3342 result = cblk->stepServer(mFrameCount); 3343 if (!result) { 3344 ALOGV("stepServer failed acquiring cblk mutex"); 3345 mStepServerFailed = true; 3346 } 3347 return result; 3348} 3349 3350void AudioFlinger::ThreadBase::TrackBase::reset() { 3351 audio_track_cblk_t* cblk = this->cblk(); 3352 3353 cblk->user = 0; 3354 cblk->server = 0; 3355 cblk->userBase = 0; 3356 cblk->serverBase = 0; 3357 mStepServerFailed = false; 3358 ALOGV("TrackBase::reset"); 3359} 3360 3361int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3362 return (int)mCblk->sampleRate; 3363} 3364 3365void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3366 audio_track_cblk_t* cblk = this->cblk(); 3367 size_t frameSize = cblk->frameSize; 3368 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3369 int8_t *bufferEnd = bufferStart + frames * frameSize; 3370 3371 // Check validity of returned pointer in case the track control block would have been corrupted. 3372 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3373 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3374 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3375 server %d, serverBase %d, user %d, userBase %d", 3376 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3377 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3378 return NULL; 3379 } 3380 3381 return bufferStart; 3382} 3383 3384// ---------------------------------------------------------------------------- 3385 3386// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3387AudioFlinger::PlaybackThread::Track::Track( 3388 PlaybackThread *thread, 3389 const sp<Client>& client, 3390 audio_stream_type_t streamType, 3391 uint32_t sampleRate, 3392 audio_format_t format, 3393 uint32_t channelMask, 3394 int frameCount, 3395 const sp<IMemory>& sharedBuffer, 3396 int sessionId) 3397 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3398 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3399 mAuxEffectId(0), mHasVolumeController(false) 3400{ 3401 if (mCblk != NULL) { 3402 if (thread != NULL) { 3403 mName = thread->getTrackName_l(); 3404 mMainBuffer = thread->mixBuffer(); 3405 } 3406 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3407 if (mName < 0) { 3408 ALOGE("no more track names available"); 3409 } 3410 mStreamType = streamType; 3411 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3412 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3413 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3414 } 3415} 3416 3417AudioFlinger::PlaybackThread::Track::~Track() 3418{ 3419 ALOGV("PlaybackThread::Track destructor"); 3420 sp<ThreadBase> thread = mThread.promote(); 3421 if (thread != 0) { 3422 Mutex::Autolock _l(thread->mLock); 3423 mState = TERMINATED; 3424 } 3425} 3426 3427void AudioFlinger::PlaybackThread::Track::destroy() 3428{ 3429 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3430 // by removing it from mTracks vector, so there is a risk that this Tracks's 3431 // destructor is called. As the destructor needs to lock mLock, 3432 // we must acquire a strong reference on this Track before locking mLock 3433 // here so that the destructor is called only when exiting this function. 3434 // On the other hand, as long as Track::destroy() is only called by 3435 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3436 // this Track with its member mTrack. 3437 sp<Track> keep(this); 3438 { // scope for mLock 3439 sp<ThreadBase> thread = mThread.promote(); 3440 if (thread != 0) { 3441 if (!isOutputTrack()) { 3442 if (mState == ACTIVE || mState == RESUMING) { 3443 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3444 3445 // to track the speaker usage 3446 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3447 } 3448 AudioSystem::releaseOutput(thread->id()); 3449 } 3450 Mutex::Autolock _l(thread->mLock); 3451 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3452 playbackThread->destroyTrack_l(this); 3453 } 3454 } 3455} 3456 3457void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3458{ 3459 uint32_t vlr = mCblk->getVolumeLR(); 3460 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", 3461 mName - AudioMixer::TRACK0, 3462 (mClient == 0) ? getpid_cached : mClient->pid(), 3463 mStreamType, 3464 mFormat, 3465 mChannelMask, 3466 mSessionId, 3467 mFrameCount, 3468 mState, 3469 mMute, 3470 mFillingUpStatus, 3471 mCblk->sampleRate, 3472 vlr & 0xFFFF, 3473 vlr >> 16, 3474 mCblk->server, 3475 mCblk->user, 3476 (int)mMainBuffer, 3477 (int)mAuxBuffer); 3478} 3479 3480// AudioBufferProvider interface 3481status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3482 AudioBufferProvider::Buffer* buffer, int64_t pts) 3483{ 3484 audio_track_cblk_t* cblk = this->cblk(); 3485 uint32_t framesReady; 3486 uint32_t framesReq = buffer->frameCount; 3487 3488 // Check if last stepServer failed, try to step now 3489 if (mStepServerFailed) { 3490 if (!step()) goto getNextBuffer_exit; 3491 ALOGV("stepServer recovered"); 3492 mStepServerFailed = false; 3493 } 3494 3495 framesReady = cblk->framesReady(); 3496 3497 if (CC_LIKELY(framesReady)) { 3498 uint32_t s = cblk->server; 3499 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3500 3501 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3502 if (framesReq > framesReady) { 3503 framesReq = framesReady; 3504 } 3505 if (s + framesReq > bufferEnd) { 3506 framesReq = bufferEnd - s; 3507 } 3508 3509 buffer->raw = getBuffer(s, framesReq); 3510 if (buffer->raw == NULL) goto getNextBuffer_exit; 3511 3512 buffer->frameCount = framesReq; 3513 return NO_ERROR; 3514 } 3515 3516getNextBuffer_exit: 3517 buffer->raw = NULL; 3518 buffer->frameCount = 0; 3519 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3520 return NOT_ENOUGH_DATA; 3521} 3522 3523uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3524 return mCblk->framesReady(); 3525} 3526 3527bool AudioFlinger::PlaybackThread::Track::isReady() const { 3528 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3529 3530 if (framesReady() >= mCblk->frameCount || 3531 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3532 mFillingUpStatus = FS_FILLED; 3533 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3534 return true; 3535 } 3536 return false; 3537} 3538 3539status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3540{ 3541 status_t status = NO_ERROR; 3542 ALOGV("start(%d), calling pid %d session %d tid %d", 3543 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3544 sp<ThreadBase> thread = mThread.promote(); 3545 if (thread != 0) { 3546 Mutex::Autolock _l(thread->mLock); 3547 track_state state = mState; 3548 // here the track could be either new, or restarted 3549 // in both cases "unstop" the track 3550 if (mState == PAUSED) { 3551 mState = TrackBase::RESUMING; 3552 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3553 } else { 3554 mState = TrackBase::ACTIVE; 3555 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3556 } 3557 3558 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3559 thread->mLock.unlock(); 3560 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3561 thread->mLock.lock(); 3562 3563 // to track the speaker usage 3564 if (status == NO_ERROR) { 3565 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3566 } 3567 } 3568 if (status == NO_ERROR) { 3569 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3570 playbackThread->addTrack_l(this); 3571 } else { 3572 mState = state; 3573 } 3574 } else { 3575 status = BAD_VALUE; 3576 } 3577 return status; 3578} 3579 3580void AudioFlinger::PlaybackThread::Track::stop() 3581{ 3582 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3583 sp<ThreadBase> thread = mThread.promote(); 3584 if (thread != 0) { 3585 Mutex::Autolock _l(thread->mLock); 3586 track_state state = mState; 3587 if (mState > STOPPED) { 3588 mState = STOPPED; 3589 // If the track is not active (PAUSED and buffers full), flush buffers 3590 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3591 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3592 reset(); 3593 } 3594 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3595 } 3596 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3597 thread->mLock.unlock(); 3598 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3599 thread->mLock.lock(); 3600 3601 // to track the speaker usage 3602 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3603 } 3604 } 3605} 3606 3607void AudioFlinger::PlaybackThread::Track::pause() 3608{ 3609 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3610 sp<ThreadBase> thread = mThread.promote(); 3611 if (thread != 0) { 3612 Mutex::Autolock _l(thread->mLock); 3613 if (mState == ACTIVE || mState == RESUMING) { 3614 mState = PAUSING; 3615 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3616 if (!isOutputTrack()) { 3617 thread->mLock.unlock(); 3618 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3619 thread->mLock.lock(); 3620 3621 // to track the speaker usage 3622 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3623 } 3624 } 3625 } 3626} 3627 3628void AudioFlinger::PlaybackThread::Track::flush() 3629{ 3630 ALOGV("flush(%d)", mName); 3631 sp<ThreadBase> thread = mThread.promote(); 3632 if (thread != 0) { 3633 Mutex::Autolock _l(thread->mLock); 3634 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3635 return; 3636 } 3637 // No point remaining in PAUSED state after a flush => go to 3638 // STOPPED state 3639 mState = STOPPED; 3640 3641 // do not reset the track if it is still in the process of being stopped or paused. 3642 // this will be done by prepareTracks_l() when the track is stopped. 3643 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3644 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3645 reset(); 3646 } 3647 } 3648} 3649 3650void AudioFlinger::PlaybackThread::Track::reset() 3651{ 3652 // Do not reset twice to avoid discarding data written just after a flush and before 3653 // the audioflinger thread detects the track is stopped. 3654 if (!mResetDone) { 3655 TrackBase::reset(); 3656 // Force underrun condition to avoid false underrun callback until first data is 3657 // written to buffer 3658 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3659 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3660 mFillingUpStatus = FS_FILLING; 3661 mResetDone = true; 3662 } 3663} 3664 3665void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3666{ 3667 mMute = muted; 3668} 3669 3670status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3671{ 3672 status_t status = DEAD_OBJECT; 3673 sp<ThreadBase> thread = mThread.promote(); 3674 if (thread != 0) { 3675 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3676 status = playbackThread->attachAuxEffect(this, EffectId); 3677 } 3678 return status; 3679} 3680 3681void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3682{ 3683 mAuxEffectId = EffectId; 3684 mAuxBuffer = buffer; 3685} 3686 3687// timed audio tracks 3688 3689sp<AudioFlinger::PlaybackThread::TimedTrack> 3690AudioFlinger::PlaybackThread::TimedTrack::create( 3691 PlaybackThread *thread, 3692 const sp<Client>& client, 3693 audio_stream_type_t streamType, 3694 uint32_t sampleRate, 3695 audio_format_t format, 3696 uint32_t channelMask, 3697 int frameCount, 3698 const sp<IMemory>& sharedBuffer, 3699 int sessionId) { 3700 if (!client->reserveTimedTrack()) 3701 return NULL; 3702 3703 sp<TimedTrack> track = new TimedTrack( 3704 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3705 sharedBuffer, sessionId); 3706 3707 if (track == NULL) { 3708 client->releaseTimedTrack(); 3709 return NULL; 3710 } 3711 3712 return track; 3713} 3714 3715AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3716 PlaybackThread *thread, 3717 const sp<Client>& client, 3718 audio_stream_type_t streamType, 3719 uint32_t sampleRate, 3720 audio_format_t format, 3721 uint32_t channelMask, 3722 int frameCount, 3723 const sp<IMemory>& sharedBuffer, 3724 int sessionId) 3725 : Track(thread, client, streamType, sampleRate, format, channelMask, 3726 frameCount, sharedBuffer, sessionId), 3727 mTimedSilenceBuffer(NULL), 3728 mTimedSilenceBufferSize(0), 3729 mTimedAudioOutputOnTime(false), 3730 mMediaTimeTransformValid(false) 3731{ 3732 LocalClock lc; 3733 mLocalTimeFreq = lc.getLocalFreq(); 3734 3735 mLocalTimeToSampleTransform.a_zero = 0; 3736 mLocalTimeToSampleTransform.b_zero = 0; 3737 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3738 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3739 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3740 &mLocalTimeToSampleTransform.a_to_b_denom); 3741} 3742 3743AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3744 mClient->releaseTimedTrack(); 3745 delete [] mTimedSilenceBuffer; 3746} 3747 3748status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3749 size_t size, sp<IMemory>* buffer) { 3750 3751 Mutex::Autolock _l(mTimedBufferQueueLock); 3752 3753 trimTimedBufferQueue_l(); 3754 3755 // lazily initialize the shared memory heap for timed buffers 3756 if (mTimedMemoryDealer == NULL) { 3757 const int kTimedBufferHeapSize = 512 << 10; 3758 3759 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3760 "AudioFlingerTimed"); 3761 if (mTimedMemoryDealer == NULL) 3762 return NO_MEMORY; 3763 } 3764 3765 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3766 if (newBuffer == NULL) { 3767 newBuffer = mTimedMemoryDealer->allocate(size); 3768 if (newBuffer == NULL) 3769 return NO_MEMORY; 3770 } 3771 3772 *buffer = newBuffer; 3773 return NO_ERROR; 3774} 3775 3776// caller must hold mTimedBufferQueueLock 3777void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3778 int64_t mediaTimeNow; 3779 { 3780 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3781 if (!mMediaTimeTransformValid) 3782 return; 3783 3784 int64_t targetTimeNow; 3785 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3786 ? mCCHelper.getCommonTime(&targetTimeNow) 3787 : mCCHelper.getLocalTime(&targetTimeNow); 3788 3789 if (OK != res) 3790 return; 3791 3792 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3793 &mediaTimeNow)) { 3794 return; 3795 } 3796 } 3797 3798 size_t trimIndex; 3799 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3800 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3801 break; 3802 } 3803 3804 if (trimIndex) { 3805 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3806 } 3807} 3808 3809status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3810 const sp<IMemory>& buffer, int64_t pts) { 3811 3812 { 3813 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3814 if (!mMediaTimeTransformValid) 3815 return INVALID_OPERATION; 3816 } 3817 3818 Mutex::Autolock _l(mTimedBufferQueueLock); 3819 3820 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3821 3822 return NO_ERROR; 3823} 3824 3825status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3826 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3827 3828 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3829 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3830 target); 3831 3832 if (!(target == TimedAudioTrack::LOCAL_TIME || 3833 target == TimedAudioTrack::COMMON_TIME)) { 3834 return BAD_VALUE; 3835 } 3836 3837 Mutex::Autolock lock(mMediaTimeTransformLock); 3838 mMediaTimeTransform = xform; 3839 mMediaTimeTransformTarget = target; 3840 mMediaTimeTransformValid = true; 3841 3842 return NO_ERROR; 3843} 3844 3845#define min(a, b) ((a) < (b) ? (a) : (b)) 3846 3847// implementation of getNextBuffer for tracks whose buffers have timestamps 3848status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3849 AudioBufferProvider::Buffer* buffer, int64_t pts) 3850{ 3851 if (pts == AudioBufferProvider::kInvalidPTS) { 3852 buffer->raw = 0; 3853 buffer->frameCount = 0; 3854 return INVALID_OPERATION; 3855 } 3856 3857 Mutex::Autolock _l(mTimedBufferQueueLock); 3858 3859 while (true) { 3860 3861 // if we have no timed buffers, then fail 3862 if (mTimedBufferQueue.isEmpty()) { 3863 buffer->raw = 0; 3864 buffer->frameCount = 0; 3865 return NOT_ENOUGH_DATA; 3866 } 3867 3868 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3869 3870 // calculate the PTS of the head of the timed buffer queue expressed in 3871 // local time 3872 int64_t headLocalPTS; 3873 { 3874 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3875 3876 assert(mMediaTimeTransformValid); 3877 3878 if (mMediaTimeTransform.a_to_b_denom == 0) { 3879 // the transform represents a pause, so yield silence 3880 timedYieldSilence(buffer->frameCount, buffer); 3881 return NO_ERROR; 3882 } 3883 3884 int64_t transformedPTS; 3885 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3886 &transformedPTS)) { 3887 // the transform failed. this shouldn't happen, but if it does 3888 // then just drop this buffer 3889 ALOGW("timedGetNextBuffer transform failed"); 3890 buffer->raw = 0; 3891 buffer->frameCount = 0; 3892 mTimedBufferQueue.removeAt(0); 3893 return NO_ERROR; 3894 } 3895 3896 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3897 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3898 &headLocalPTS)) { 3899 buffer->raw = 0; 3900 buffer->frameCount = 0; 3901 return INVALID_OPERATION; 3902 } 3903 } else { 3904 headLocalPTS = transformedPTS; 3905 } 3906 } 3907 3908 // adjust the head buffer's PTS to reflect the portion of the head buffer 3909 // that has already been consumed 3910 int64_t effectivePTS = headLocalPTS + 3911 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3912 3913 // Calculate the delta in samples between the head of the input buffer 3914 // queue and the start of the next output buffer that will be written. 3915 // If the transformation fails because of over or underflow, it means 3916 // that the sample's position in the output stream is so far out of 3917 // whack that it should just be dropped. 3918 int64_t sampleDelta; 3919 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3920 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3921 mTimedBufferQueue.removeAt(0); 3922 continue; 3923 } 3924 if (!mLocalTimeToSampleTransform.doForwardTransform( 3925 (effectivePTS - pts) << 32, &sampleDelta)) { 3926 ALOGV("*** too late during sample rate transform: dropped buffer"); 3927 mTimedBufferQueue.removeAt(0); 3928 continue; 3929 } 3930 3931 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 3932 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 3933 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 3934 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 3935 3936 // if the delta between the ideal placement for the next input sample and 3937 // the current output position is within this threshold, then we will 3938 // concatenate the next input samples to the previous output 3939 const int64_t kSampleContinuityThreshold = 3940 (static_cast<int64_t>(sampleRate()) << 32) / 10; 3941 3942 // if this is the first buffer of audio that we're emitting from this track 3943 // then it should be almost exactly on time. 3944 const int64_t kSampleStartupThreshold = 1LL << 32; 3945 3946 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 3947 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 3948 // the next input is close enough to being on time, so concatenate it 3949 // with the last output 3950 timedYieldSamples(buffer); 3951 3952 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3953 return NO_ERROR; 3954 } else if (sampleDelta > 0) { 3955 // the gap between the current output position and the proper start of 3956 // the next input sample is too big, so fill it with silence 3957 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 3958 3959 timedYieldSilence(framesUntilNextInput, buffer); 3960 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 3961 return NO_ERROR; 3962 } else { 3963 // the next input sample is late 3964 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 3965 size_t onTimeSamplePosition = 3966 head.position() + lateFrames * mCblk->frameSize; 3967 3968 if (onTimeSamplePosition > head.buffer()->size()) { 3969 // all the remaining samples in the head are too late, so 3970 // drop it and move on 3971 ALOGV("*** too late: dropped buffer"); 3972 mTimedBufferQueue.removeAt(0); 3973 continue; 3974 } else { 3975 // skip over the late samples 3976 head.setPosition(onTimeSamplePosition); 3977 3978 // yield the available samples 3979 timedYieldSamples(buffer); 3980 3981 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3982 return NO_ERROR; 3983 } 3984 } 3985 } 3986} 3987 3988// Yield samples from the timed buffer queue head up to the given output 3989// buffer's capacity. 3990// 3991// Caller must hold mTimedBufferQueueLock 3992void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 3993 AudioBufferProvider::Buffer* buffer) { 3994 3995 const TimedBuffer& head = mTimedBufferQueue[0]; 3996 3997 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 3998 head.position()); 3999 4000 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4001 mCblk->frameSize); 4002 size_t framesRequested = buffer->frameCount; 4003 buffer->frameCount = min(framesLeftInHead, framesRequested); 4004 4005 mTimedAudioOutputOnTime = true; 4006} 4007 4008// Yield samples of silence up to the given output buffer's capacity 4009// 4010// Caller must hold mTimedBufferQueueLock 4011void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4012 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4013 4014 // lazily allocate a buffer filled with silence 4015 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4016 delete [] mTimedSilenceBuffer; 4017 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4018 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4019 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4020 } 4021 4022 buffer->raw = mTimedSilenceBuffer; 4023 size_t framesRequested = buffer->frameCount; 4024 buffer->frameCount = min(numFrames, framesRequested); 4025 4026 mTimedAudioOutputOnTime = false; 4027} 4028 4029// AudioBufferProvider interface 4030void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4031 AudioBufferProvider::Buffer* buffer) { 4032 4033 Mutex::Autolock _l(mTimedBufferQueueLock); 4034 4035 // If the buffer which was just released is part of the buffer at the head 4036 // of the queue, be sure to update the amt of the buffer which has been 4037 // consumed. If the buffer being returned is not part of the head of the 4038 // queue, its either because the buffer is part of the silence buffer, or 4039 // because the head of the timed queue was trimmed after the mixer called 4040 // getNextBuffer but before the mixer called releaseBuffer. 4041 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4042 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4043 4044 void* start = head.buffer()->pointer(); 4045 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4046 4047 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4048 head.setPosition(head.position() + 4049 (buffer->frameCount * mCblk->frameSize)); 4050 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4051 mTimedBufferQueue.removeAt(0); 4052 } 4053 } 4054 } 4055 4056 buffer->raw = 0; 4057 buffer->frameCount = 0; 4058} 4059 4060uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4061 Mutex::Autolock _l(mTimedBufferQueueLock); 4062 4063 uint32_t frames = 0; 4064 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4065 const TimedBuffer& tb = mTimedBufferQueue[i]; 4066 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4067 } 4068 4069 return frames; 4070} 4071 4072AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4073 : mPTS(0), mPosition(0) {} 4074 4075AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4076 const sp<IMemory>& buffer, int64_t pts) 4077 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4078 4079// ---------------------------------------------------------------------------- 4080 4081// RecordTrack constructor must be called with AudioFlinger::mLock held 4082AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4083 RecordThread *thread, 4084 const sp<Client>& client, 4085 uint32_t sampleRate, 4086 audio_format_t format, 4087 uint32_t channelMask, 4088 int frameCount, 4089 int sessionId) 4090 : TrackBase(thread, client, sampleRate, format, 4091 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4092 mOverflow(false) 4093{ 4094 if (mCblk != NULL) { 4095 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4096 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4097 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4098 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4099 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4100 } else { 4101 mCblk->frameSize = sizeof(int8_t); 4102 } 4103 } 4104} 4105 4106AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4107{ 4108 sp<ThreadBase> thread = mThread.promote(); 4109 if (thread != 0) { 4110 AudioSystem::releaseInput(thread->id()); 4111 } 4112} 4113 4114// AudioBufferProvider interface 4115status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4116{ 4117 audio_track_cblk_t* cblk = this->cblk(); 4118 uint32_t framesAvail; 4119 uint32_t framesReq = buffer->frameCount; 4120 4121 // Check if last stepServer failed, try to step now 4122 if (mStepServerFailed) { 4123 if (!step()) goto getNextBuffer_exit; 4124 ALOGV("stepServer recovered"); 4125 mStepServerFailed = false; 4126 } 4127 4128 framesAvail = cblk->framesAvailable_l(); 4129 4130 if (CC_LIKELY(framesAvail)) { 4131 uint32_t s = cblk->server; 4132 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4133 4134 if (framesReq > framesAvail) { 4135 framesReq = framesAvail; 4136 } 4137 if (s + framesReq > bufferEnd) { 4138 framesReq = bufferEnd - s; 4139 } 4140 4141 buffer->raw = getBuffer(s, framesReq); 4142 if (buffer->raw == NULL) goto getNextBuffer_exit; 4143 4144 buffer->frameCount = framesReq; 4145 return NO_ERROR; 4146 } 4147 4148getNextBuffer_exit: 4149 buffer->raw = NULL; 4150 buffer->frameCount = 0; 4151 return NOT_ENOUGH_DATA; 4152} 4153 4154status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4155{ 4156 sp<ThreadBase> thread = mThread.promote(); 4157 if (thread != 0) { 4158 RecordThread *recordThread = (RecordThread *)thread.get(); 4159 return recordThread->start(this, tid); 4160 } else { 4161 return BAD_VALUE; 4162 } 4163} 4164 4165void AudioFlinger::RecordThread::RecordTrack::stop() 4166{ 4167 sp<ThreadBase> thread = mThread.promote(); 4168 if (thread != 0) { 4169 RecordThread *recordThread = (RecordThread *)thread.get(); 4170 recordThread->stop(this); 4171 TrackBase::reset(); 4172 // Force overerrun condition to avoid false overrun callback until first data is 4173 // read from buffer 4174 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4175 } 4176} 4177 4178void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4179{ 4180 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4181 (mClient == 0) ? getpid_cached : mClient->pid(), 4182 mFormat, 4183 mChannelMask, 4184 mSessionId, 4185 mFrameCount, 4186 mState, 4187 mCblk->sampleRate, 4188 mCblk->server, 4189 mCblk->user); 4190} 4191 4192 4193// ---------------------------------------------------------------------------- 4194 4195AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4196 PlaybackThread *playbackThread, 4197 DuplicatingThread *sourceThread, 4198 uint32_t sampleRate, 4199 audio_format_t format, 4200 uint32_t channelMask, 4201 int frameCount) 4202 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4203 mActive(false), mSourceThread(sourceThread) 4204{ 4205 4206 if (mCblk != NULL) { 4207 mCblk->flags |= CBLK_DIRECTION_OUT; 4208 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4209 mOutBuffer.frameCount = 0; 4210 playbackThread->mTracks.add(this); 4211 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4212 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4213 mCblk, mBuffer, mCblk->buffers, 4214 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4215 } else { 4216 ALOGW("Error creating output track on thread %p", playbackThread); 4217 } 4218} 4219 4220AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4221{ 4222 clearBufferQueue(); 4223} 4224 4225status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4226{ 4227 status_t status = Track::start(tid); 4228 if (status != NO_ERROR) { 4229 return status; 4230 } 4231 4232 mActive = true; 4233 mRetryCount = 127; 4234 return status; 4235} 4236 4237void AudioFlinger::PlaybackThread::OutputTrack::stop() 4238{ 4239 Track::stop(); 4240 clearBufferQueue(); 4241 mOutBuffer.frameCount = 0; 4242 mActive = false; 4243} 4244 4245bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4246{ 4247 Buffer *pInBuffer; 4248 Buffer inBuffer; 4249 uint32_t channelCount = mChannelCount; 4250 bool outputBufferFull = false; 4251 inBuffer.frameCount = frames; 4252 inBuffer.i16 = data; 4253 4254 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4255 4256 if (!mActive && frames != 0) { 4257 start(0); 4258 sp<ThreadBase> thread = mThread.promote(); 4259 if (thread != 0) { 4260 MixerThread *mixerThread = (MixerThread *)thread.get(); 4261 if (mCblk->frameCount > frames){ 4262 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4263 uint32_t startFrames = (mCblk->frameCount - frames); 4264 pInBuffer = new Buffer; 4265 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4266 pInBuffer->frameCount = startFrames; 4267 pInBuffer->i16 = pInBuffer->mBuffer; 4268 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4269 mBufferQueue.add(pInBuffer); 4270 } else { 4271 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4272 } 4273 } 4274 } 4275 } 4276 4277 while (waitTimeLeftMs) { 4278 // First write pending buffers, then new data 4279 if (mBufferQueue.size()) { 4280 pInBuffer = mBufferQueue.itemAt(0); 4281 } else { 4282 pInBuffer = &inBuffer; 4283 } 4284 4285 if (pInBuffer->frameCount == 0) { 4286 break; 4287 } 4288 4289 if (mOutBuffer.frameCount == 0) { 4290 mOutBuffer.frameCount = pInBuffer->frameCount; 4291 nsecs_t startTime = systemTime(); 4292 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4293 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4294 outputBufferFull = true; 4295 break; 4296 } 4297 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4298 if (waitTimeLeftMs >= waitTimeMs) { 4299 waitTimeLeftMs -= waitTimeMs; 4300 } else { 4301 waitTimeLeftMs = 0; 4302 } 4303 } 4304 4305 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4306 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4307 mCblk->stepUser(outFrames); 4308 pInBuffer->frameCount -= outFrames; 4309 pInBuffer->i16 += outFrames * channelCount; 4310 mOutBuffer.frameCount -= outFrames; 4311 mOutBuffer.i16 += outFrames * channelCount; 4312 4313 if (pInBuffer->frameCount == 0) { 4314 if (mBufferQueue.size()) { 4315 mBufferQueue.removeAt(0); 4316 delete [] pInBuffer->mBuffer; 4317 delete pInBuffer; 4318 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4319 } else { 4320 break; 4321 } 4322 } 4323 } 4324 4325 // If we could not write all frames, allocate a buffer and queue it for next time. 4326 if (inBuffer.frameCount) { 4327 sp<ThreadBase> thread = mThread.promote(); 4328 if (thread != 0 && !thread->standby()) { 4329 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4330 pInBuffer = new Buffer; 4331 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4332 pInBuffer->frameCount = inBuffer.frameCount; 4333 pInBuffer->i16 = pInBuffer->mBuffer; 4334 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4335 mBufferQueue.add(pInBuffer); 4336 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4337 } else { 4338 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4339 } 4340 } 4341 } 4342 4343 // Calling write() with a 0 length buffer, means that no more data will be written: 4344 // If no more buffers are pending, fill output track buffer to make sure it is started 4345 // by output mixer. 4346 if (frames == 0 && mBufferQueue.size() == 0) { 4347 if (mCblk->user < mCblk->frameCount) { 4348 frames = mCblk->frameCount - mCblk->user; 4349 pInBuffer = new Buffer; 4350 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4351 pInBuffer->frameCount = frames; 4352 pInBuffer->i16 = pInBuffer->mBuffer; 4353 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4354 mBufferQueue.add(pInBuffer); 4355 } else if (mActive) { 4356 stop(); 4357 } 4358 } 4359 4360 return outputBufferFull; 4361} 4362 4363status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4364{ 4365 int active; 4366 status_t result; 4367 audio_track_cblk_t* cblk = mCblk; 4368 uint32_t framesReq = buffer->frameCount; 4369 4370// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4371 buffer->frameCount = 0; 4372 4373 uint32_t framesAvail = cblk->framesAvailable(); 4374 4375 4376 if (framesAvail == 0) { 4377 Mutex::Autolock _l(cblk->lock); 4378 goto start_loop_here; 4379 while (framesAvail == 0) { 4380 active = mActive; 4381 if (CC_UNLIKELY(!active)) { 4382 ALOGV("Not active and NO_MORE_BUFFERS"); 4383 return NO_MORE_BUFFERS; 4384 } 4385 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4386 if (result != NO_ERROR) { 4387 return NO_MORE_BUFFERS; 4388 } 4389 // read the server count again 4390 start_loop_here: 4391 framesAvail = cblk->framesAvailable_l(); 4392 } 4393 } 4394 4395// if (framesAvail < framesReq) { 4396// return NO_MORE_BUFFERS; 4397// } 4398 4399 if (framesReq > framesAvail) { 4400 framesReq = framesAvail; 4401 } 4402 4403 uint32_t u = cblk->user; 4404 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4405 4406 if (u + framesReq > bufferEnd) { 4407 framesReq = bufferEnd - u; 4408 } 4409 4410 buffer->frameCount = framesReq; 4411 buffer->raw = (void *)cblk->buffer(u); 4412 return NO_ERROR; 4413} 4414 4415 4416void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4417{ 4418 size_t size = mBufferQueue.size(); 4419 4420 for (size_t i = 0; i < size; i++) { 4421 Buffer *pBuffer = mBufferQueue.itemAt(i); 4422 delete [] pBuffer->mBuffer; 4423 delete pBuffer; 4424 } 4425 mBufferQueue.clear(); 4426} 4427 4428// ---------------------------------------------------------------------------- 4429 4430AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4431 : RefBase(), 4432 mAudioFlinger(audioFlinger), 4433 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4434 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4435 mPid(pid), 4436 mTimedTrackCount(0) 4437{ 4438 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4439} 4440 4441// Client destructor must be called with AudioFlinger::mLock held 4442AudioFlinger::Client::~Client() 4443{ 4444 mAudioFlinger->removeClient_l(mPid); 4445} 4446 4447sp<MemoryDealer> AudioFlinger::Client::heap() const 4448{ 4449 return mMemoryDealer; 4450} 4451 4452// Reserve one of the limited slots for a timed audio track associated 4453// with this client 4454bool AudioFlinger::Client::reserveTimedTrack() 4455{ 4456 const int kMaxTimedTracksPerClient = 4; 4457 4458 Mutex::Autolock _l(mTimedTrackLock); 4459 4460 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4461 ALOGW("can not create timed track - pid %d has exceeded the limit", 4462 mPid); 4463 return false; 4464 } 4465 4466 mTimedTrackCount++; 4467 return true; 4468} 4469 4470// Release a slot for a timed audio track 4471void AudioFlinger::Client::releaseTimedTrack() 4472{ 4473 Mutex::Autolock _l(mTimedTrackLock); 4474 mTimedTrackCount--; 4475} 4476 4477// ---------------------------------------------------------------------------- 4478 4479AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4480 const sp<IAudioFlingerClient>& client, 4481 pid_t pid) 4482 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4483{ 4484} 4485 4486AudioFlinger::NotificationClient::~NotificationClient() 4487{ 4488} 4489 4490void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4491{ 4492 sp<NotificationClient> keep(this); 4493 mAudioFlinger->removeNotificationClient(mPid); 4494} 4495 4496// ---------------------------------------------------------------------------- 4497 4498AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4499 : BnAudioTrack(), 4500 mTrack(track) 4501{ 4502} 4503 4504AudioFlinger::TrackHandle::~TrackHandle() { 4505 // just stop the track on deletion, associated resources 4506 // will be freed from the main thread once all pending buffers have 4507 // been played. Unless it's not in the active track list, in which 4508 // case we free everything now... 4509 mTrack->destroy(); 4510} 4511 4512sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4513 return mTrack->getCblk(); 4514} 4515 4516status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4517 return mTrack->start(tid); 4518} 4519 4520void AudioFlinger::TrackHandle::stop() { 4521 mTrack->stop(); 4522} 4523 4524void AudioFlinger::TrackHandle::flush() { 4525 mTrack->flush(); 4526} 4527 4528void AudioFlinger::TrackHandle::mute(bool e) { 4529 mTrack->mute(e); 4530} 4531 4532void AudioFlinger::TrackHandle::pause() { 4533 mTrack->pause(); 4534} 4535 4536status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4537{ 4538 return mTrack->attachAuxEffect(EffectId); 4539} 4540 4541status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4542 sp<IMemory>* buffer) { 4543 if (!mTrack->isTimedTrack()) 4544 return INVALID_OPERATION; 4545 4546 PlaybackThread::TimedTrack* tt = 4547 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4548 return tt->allocateTimedBuffer(size, buffer); 4549} 4550 4551status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4552 int64_t pts) { 4553 if (!mTrack->isTimedTrack()) 4554 return INVALID_OPERATION; 4555 4556 PlaybackThread::TimedTrack* tt = 4557 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4558 return tt->queueTimedBuffer(buffer, pts); 4559} 4560 4561status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4562 const LinearTransform& xform, int target) { 4563 4564 if (!mTrack->isTimedTrack()) 4565 return INVALID_OPERATION; 4566 4567 PlaybackThread::TimedTrack* tt = 4568 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4569 return tt->setMediaTimeTransform( 4570 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4571} 4572 4573status_t AudioFlinger::TrackHandle::onTransact( 4574 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4575{ 4576 return BnAudioTrack::onTransact(code, data, reply, flags); 4577} 4578 4579// ---------------------------------------------------------------------------- 4580 4581sp<IAudioRecord> AudioFlinger::openRecord( 4582 pid_t pid, 4583 audio_io_handle_t input, 4584 uint32_t sampleRate, 4585 audio_format_t format, 4586 uint32_t channelMask, 4587 int frameCount, 4588 // FIXME dead, remove from IAudioFlinger 4589 uint32_t flags, 4590 int *sessionId, 4591 status_t *status) 4592{ 4593 sp<RecordThread::RecordTrack> recordTrack; 4594 sp<RecordHandle> recordHandle; 4595 sp<Client> client; 4596 status_t lStatus; 4597 RecordThread *thread; 4598 size_t inFrameCount; 4599 int lSessionId; 4600 4601 // check calling permissions 4602 if (!recordingAllowed()) { 4603 lStatus = PERMISSION_DENIED; 4604 goto Exit; 4605 } 4606 4607 // add client to list 4608 { // scope for mLock 4609 Mutex::Autolock _l(mLock); 4610 thread = checkRecordThread_l(input); 4611 if (thread == NULL) { 4612 lStatus = BAD_VALUE; 4613 goto Exit; 4614 } 4615 4616 client = registerPid_l(pid); 4617 4618 // If no audio session id is provided, create one here 4619 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4620 lSessionId = *sessionId; 4621 } else { 4622 lSessionId = nextUniqueId(); 4623 if (sessionId != NULL) { 4624 *sessionId = lSessionId; 4625 } 4626 } 4627 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4628 recordTrack = thread->createRecordTrack_l(client, 4629 sampleRate, 4630 format, 4631 channelMask, 4632 frameCount, 4633 lSessionId, 4634 &lStatus); 4635 } 4636 if (lStatus != NO_ERROR) { 4637 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4638 // destructor is called by the TrackBase destructor with mLock held 4639 client.clear(); 4640 recordTrack.clear(); 4641 goto Exit; 4642 } 4643 4644 // return to handle to client 4645 recordHandle = new RecordHandle(recordTrack); 4646 lStatus = NO_ERROR; 4647 4648Exit: 4649 if (status) { 4650 *status = lStatus; 4651 } 4652 return recordHandle; 4653} 4654 4655// ---------------------------------------------------------------------------- 4656 4657AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4658 : BnAudioRecord(), 4659 mRecordTrack(recordTrack) 4660{ 4661} 4662 4663AudioFlinger::RecordHandle::~RecordHandle() { 4664 stop(); 4665} 4666 4667sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4668 return mRecordTrack->getCblk(); 4669} 4670 4671status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4672 ALOGV("RecordHandle::start()"); 4673 return mRecordTrack->start(tid); 4674} 4675 4676void AudioFlinger::RecordHandle::stop() { 4677 ALOGV("RecordHandle::stop()"); 4678 mRecordTrack->stop(); 4679} 4680 4681status_t AudioFlinger::RecordHandle::onTransact( 4682 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4683{ 4684 return BnAudioRecord::onTransact(code, data, reply, flags); 4685} 4686 4687// ---------------------------------------------------------------------------- 4688 4689AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4690 AudioStreamIn *input, 4691 uint32_t sampleRate, 4692 uint32_t channels, 4693 audio_io_handle_t id, 4694 uint32_t device) : 4695 ThreadBase(audioFlinger, id, device, RECORD), 4696 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4697 // mRsmpInIndex and mInputBytes set by readInputParameters() 4698 mReqChannelCount(popcount(channels)), 4699 mReqSampleRate(sampleRate) 4700 // mBytesRead is only meaningful while active, and so is cleared in start() 4701 // (but might be better to also clear here for dump?) 4702{ 4703 snprintf(mName, kNameLength, "AudioIn_%X", id); 4704 4705 readInputParameters(); 4706} 4707 4708 4709AudioFlinger::RecordThread::~RecordThread() 4710{ 4711 delete[] mRsmpInBuffer; 4712 delete mResampler; 4713 delete[] mRsmpOutBuffer; 4714} 4715 4716void AudioFlinger::RecordThread::onFirstRef() 4717{ 4718 run(mName, PRIORITY_URGENT_AUDIO); 4719} 4720 4721status_t AudioFlinger::RecordThread::readyToRun() 4722{ 4723 status_t status = initCheck(); 4724 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4725 return status; 4726} 4727 4728bool AudioFlinger::RecordThread::threadLoop() 4729{ 4730 AudioBufferProvider::Buffer buffer; 4731 sp<RecordTrack> activeTrack; 4732 Vector< sp<EffectChain> > effectChains; 4733 4734 nsecs_t lastWarning = 0; 4735 4736 acquireWakeLock(); 4737 4738 // start recording 4739 while (!exitPending()) { 4740 4741 processConfigEvents(); 4742 4743 { // scope for mLock 4744 Mutex::Autolock _l(mLock); 4745 checkForNewParameters_l(); 4746 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4747 if (!mStandby) { 4748 mInput->stream->common.standby(&mInput->stream->common); 4749 mStandby = true; 4750 } 4751 4752 if (exitPending()) break; 4753 4754 releaseWakeLock_l(); 4755 ALOGV("RecordThread: loop stopping"); 4756 // go to sleep 4757 mWaitWorkCV.wait(mLock); 4758 ALOGV("RecordThread: loop starting"); 4759 acquireWakeLock_l(); 4760 continue; 4761 } 4762 if (mActiveTrack != 0) { 4763 if (mActiveTrack->mState == TrackBase::PAUSING) { 4764 if (!mStandby) { 4765 mInput->stream->common.standby(&mInput->stream->common); 4766 mStandby = true; 4767 } 4768 mActiveTrack.clear(); 4769 mStartStopCond.broadcast(); 4770 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4771 if (mReqChannelCount != mActiveTrack->channelCount()) { 4772 mActiveTrack.clear(); 4773 mStartStopCond.broadcast(); 4774 } else if (mBytesRead != 0) { 4775 // record start succeeds only if first read from audio input 4776 // succeeds 4777 if (mBytesRead > 0) { 4778 mActiveTrack->mState = TrackBase::ACTIVE; 4779 } else { 4780 mActiveTrack.clear(); 4781 } 4782 mStartStopCond.broadcast(); 4783 } 4784 mStandby = false; 4785 } 4786 } 4787 lockEffectChains_l(effectChains); 4788 } 4789 4790 if (mActiveTrack != 0) { 4791 if (mActiveTrack->mState != TrackBase::ACTIVE && 4792 mActiveTrack->mState != TrackBase::RESUMING) { 4793 unlockEffectChains(effectChains); 4794 usleep(kRecordThreadSleepUs); 4795 continue; 4796 } 4797 for (size_t i = 0; i < effectChains.size(); i ++) { 4798 effectChains[i]->process_l(); 4799 } 4800 4801 buffer.frameCount = mFrameCount; 4802 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4803 size_t framesOut = buffer.frameCount; 4804 if (mResampler == NULL) { 4805 // no resampling 4806 while (framesOut) { 4807 size_t framesIn = mFrameCount - mRsmpInIndex; 4808 if (framesIn) { 4809 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4810 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4811 if (framesIn > framesOut) 4812 framesIn = framesOut; 4813 mRsmpInIndex += framesIn; 4814 framesOut -= framesIn; 4815 if ((int)mChannelCount == mReqChannelCount || 4816 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4817 memcpy(dst, src, framesIn * mFrameSize); 4818 } else { 4819 int16_t *src16 = (int16_t *)src; 4820 int16_t *dst16 = (int16_t *)dst; 4821 if (mChannelCount == 1) { 4822 while (framesIn--) { 4823 *dst16++ = *src16; 4824 *dst16++ = *src16++; 4825 } 4826 } else { 4827 while (framesIn--) { 4828 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4829 src16 += 2; 4830 } 4831 } 4832 } 4833 } 4834 if (framesOut && mFrameCount == mRsmpInIndex) { 4835 if (framesOut == mFrameCount && 4836 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4837 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4838 framesOut = 0; 4839 } else { 4840 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4841 mRsmpInIndex = 0; 4842 } 4843 if (mBytesRead < 0) { 4844 ALOGE("Error reading audio input"); 4845 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4846 // Force input into standby so that it tries to 4847 // recover at next read attempt 4848 mInput->stream->common.standby(&mInput->stream->common); 4849 usleep(kRecordThreadSleepUs); 4850 } 4851 mRsmpInIndex = mFrameCount; 4852 framesOut = 0; 4853 buffer.frameCount = 0; 4854 } 4855 } 4856 } 4857 } else { 4858 // resampling 4859 4860 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4861 // alter output frame count as if we were expecting stereo samples 4862 if (mChannelCount == 1 && mReqChannelCount == 1) { 4863 framesOut >>= 1; 4864 } 4865 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4866 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4867 // are 32 bit aligned which should be always true. 4868 if (mChannelCount == 2 && mReqChannelCount == 1) { 4869 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4870 // the resampler always outputs stereo samples: do post stereo to mono conversion 4871 int16_t *src = (int16_t *)mRsmpOutBuffer; 4872 int16_t *dst = buffer.i16; 4873 while (framesOut--) { 4874 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4875 src += 2; 4876 } 4877 } else { 4878 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4879 } 4880 4881 } 4882 mActiveTrack->releaseBuffer(&buffer); 4883 mActiveTrack->overflow(); 4884 } 4885 // client isn't retrieving buffers fast enough 4886 else { 4887 if (!mActiveTrack->setOverflow()) { 4888 nsecs_t now = systemTime(); 4889 if ((now - lastWarning) > kWarningThrottleNs) { 4890 ALOGW("RecordThread: buffer overflow"); 4891 lastWarning = now; 4892 } 4893 } 4894 // Release the processor for a while before asking for a new buffer. 4895 // This will give the application more chance to read from the buffer and 4896 // clear the overflow. 4897 usleep(kRecordThreadSleepUs); 4898 } 4899 } 4900 // enable changes in effect chain 4901 unlockEffectChains(effectChains); 4902 effectChains.clear(); 4903 } 4904 4905 if (!mStandby) { 4906 mInput->stream->common.standby(&mInput->stream->common); 4907 } 4908 mActiveTrack.clear(); 4909 4910 mStartStopCond.broadcast(); 4911 4912 releaseWakeLock(); 4913 4914 ALOGV("RecordThread %p exiting", this); 4915 return false; 4916} 4917 4918 4919sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4920 const sp<AudioFlinger::Client>& client, 4921 uint32_t sampleRate, 4922 audio_format_t format, 4923 int channelMask, 4924 int frameCount, 4925 int sessionId, 4926 status_t *status) 4927{ 4928 sp<RecordTrack> track; 4929 status_t lStatus; 4930 4931 lStatus = initCheck(); 4932 if (lStatus != NO_ERROR) { 4933 ALOGE("Audio driver not initialized."); 4934 goto Exit; 4935 } 4936 4937 { // scope for mLock 4938 Mutex::Autolock _l(mLock); 4939 4940 track = new RecordTrack(this, client, sampleRate, 4941 format, channelMask, frameCount, sessionId); 4942 4943 if (track->getCblk() == 0) { 4944 lStatus = NO_MEMORY; 4945 goto Exit; 4946 } 4947 4948 mTrack = track.get(); 4949 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4950 bool suspend = audio_is_bluetooth_sco_device( 4951 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4952 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4953 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4954 } 4955 lStatus = NO_ERROR; 4956 4957Exit: 4958 if (status) { 4959 *status = lStatus; 4960 } 4961 return track; 4962} 4963 4964status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 4965{ 4966 ALOGV("RecordThread::start tid=%d", tid); 4967 sp <ThreadBase> strongMe = this; 4968 status_t status = NO_ERROR; 4969 { 4970 AutoMutex lock(mLock); 4971 if (mActiveTrack != 0) { 4972 if (recordTrack != mActiveTrack.get()) { 4973 status = -EBUSY; 4974 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4975 mActiveTrack->mState = TrackBase::ACTIVE; 4976 } 4977 return status; 4978 } 4979 4980 recordTrack->mState = TrackBase::IDLE; 4981 mActiveTrack = recordTrack; 4982 mLock.unlock(); 4983 status_t status = AudioSystem::startInput(mId); 4984 mLock.lock(); 4985 if (status != NO_ERROR) { 4986 mActiveTrack.clear(); 4987 return status; 4988 } 4989 mRsmpInIndex = mFrameCount; 4990 mBytesRead = 0; 4991 if (mResampler != NULL) { 4992 mResampler->reset(); 4993 } 4994 mActiveTrack->mState = TrackBase::RESUMING; 4995 // signal thread to start 4996 ALOGV("Signal record thread"); 4997 mWaitWorkCV.signal(); 4998 // do not wait for mStartStopCond if exiting 4999 if (exitPending()) { 5000 mActiveTrack.clear(); 5001 status = INVALID_OPERATION; 5002 goto startError; 5003 } 5004 mStartStopCond.wait(mLock); 5005 if (mActiveTrack == 0) { 5006 ALOGV("Record failed to start"); 5007 status = BAD_VALUE; 5008 goto startError; 5009 } 5010 ALOGV("Record started OK"); 5011 return status; 5012 } 5013startError: 5014 AudioSystem::stopInput(mId); 5015 return status; 5016} 5017 5018void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5019 ALOGV("RecordThread::stop"); 5020 sp <ThreadBase> strongMe = this; 5021 { 5022 AutoMutex lock(mLock); 5023 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5024 mActiveTrack->mState = TrackBase::PAUSING; 5025 // do not wait for mStartStopCond if exiting 5026 if (exitPending()) { 5027 return; 5028 } 5029 mStartStopCond.wait(mLock); 5030 // if we have been restarted, recordTrack == mActiveTrack.get() here 5031 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5032 mLock.unlock(); 5033 AudioSystem::stopInput(mId); 5034 mLock.lock(); 5035 ALOGV("Record stopped OK"); 5036 } 5037 } 5038 } 5039} 5040 5041status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5042{ 5043 const size_t SIZE = 256; 5044 char buffer[SIZE]; 5045 String8 result; 5046 5047 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5048 result.append(buffer); 5049 5050 if (mActiveTrack != 0) { 5051 result.append("Active Track:\n"); 5052 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5053 mActiveTrack->dump(buffer, SIZE); 5054 result.append(buffer); 5055 5056 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5057 result.append(buffer); 5058 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5059 result.append(buffer); 5060 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5061 result.append(buffer); 5062 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5063 result.append(buffer); 5064 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5065 result.append(buffer); 5066 5067 5068 } else { 5069 result.append("No record client\n"); 5070 } 5071 write(fd, result.string(), result.size()); 5072 5073 dumpBase(fd, args); 5074 dumpEffectChains(fd, args); 5075 5076 return NO_ERROR; 5077} 5078 5079// AudioBufferProvider interface 5080status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5081{ 5082 size_t framesReq = buffer->frameCount; 5083 size_t framesReady = mFrameCount - mRsmpInIndex; 5084 int channelCount; 5085 5086 if (framesReady == 0) { 5087 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5088 if (mBytesRead < 0) { 5089 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5090 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5091 // Force input into standby so that it tries to 5092 // recover at next read attempt 5093 mInput->stream->common.standby(&mInput->stream->common); 5094 usleep(kRecordThreadSleepUs); 5095 } 5096 buffer->raw = NULL; 5097 buffer->frameCount = 0; 5098 return NOT_ENOUGH_DATA; 5099 } 5100 mRsmpInIndex = 0; 5101 framesReady = mFrameCount; 5102 } 5103 5104 if (framesReq > framesReady) { 5105 framesReq = framesReady; 5106 } 5107 5108 if (mChannelCount == 1 && mReqChannelCount == 2) { 5109 channelCount = 1; 5110 } else { 5111 channelCount = 2; 5112 } 5113 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5114 buffer->frameCount = framesReq; 5115 return NO_ERROR; 5116} 5117 5118// AudioBufferProvider interface 5119void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5120{ 5121 mRsmpInIndex += buffer->frameCount; 5122 buffer->frameCount = 0; 5123} 5124 5125bool AudioFlinger::RecordThread::checkForNewParameters_l() 5126{ 5127 bool reconfig = false; 5128 5129 while (!mNewParameters.isEmpty()) { 5130 status_t status = NO_ERROR; 5131 String8 keyValuePair = mNewParameters[0]; 5132 AudioParameter param = AudioParameter(keyValuePair); 5133 int value; 5134 audio_format_t reqFormat = mFormat; 5135 int reqSamplingRate = mReqSampleRate; 5136 int reqChannelCount = mReqChannelCount; 5137 5138 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5139 reqSamplingRate = value; 5140 reconfig = true; 5141 } 5142 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5143 reqFormat = (audio_format_t) value; 5144 reconfig = true; 5145 } 5146 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5147 reqChannelCount = popcount(value); 5148 reconfig = true; 5149 } 5150 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5151 // do not accept frame count changes if tracks are open as the track buffer 5152 // size depends on frame count and correct behavior would not be guaranteed 5153 // if frame count is changed after track creation 5154 if (mActiveTrack != 0) { 5155 status = INVALID_OPERATION; 5156 } else { 5157 reconfig = true; 5158 } 5159 } 5160 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5161 // forward device change to effects that have requested to be 5162 // aware of attached audio device. 5163 for (size_t i = 0; i < mEffectChains.size(); i++) { 5164 mEffectChains[i]->setDevice_l(value); 5165 } 5166 // store input device and output device but do not forward output device to audio HAL. 5167 // Note that status is ignored by the caller for output device 5168 // (see AudioFlinger::setParameters() 5169 if (value & AUDIO_DEVICE_OUT_ALL) { 5170 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5171 status = BAD_VALUE; 5172 } else { 5173 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5174 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5175 if (mTrack != NULL) { 5176 bool suspend = audio_is_bluetooth_sco_device( 5177 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5178 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5179 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5180 } 5181 } 5182 mDevice |= (uint32_t)value; 5183 } 5184 if (status == NO_ERROR) { 5185 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5186 if (status == INVALID_OPERATION) { 5187 mInput->stream->common.standby(&mInput->stream->common); 5188 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5189 } 5190 if (reconfig) { 5191 if (status == BAD_VALUE && 5192 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5193 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5194 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5195 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5196 (reqChannelCount <= FCC_2)) { 5197 status = NO_ERROR; 5198 } 5199 if (status == NO_ERROR) { 5200 readInputParameters(); 5201 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5202 } 5203 } 5204 } 5205 5206 mNewParameters.removeAt(0); 5207 5208 mParamStatus = status; 5209 mParamCond.signal(); 5210 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5211 // already timed out waiting for the status and will never signal the condition. 5212 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5213 } 5214 return reconfig; 5215} 5216 5217String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5218{ 5219 char *s; 5220 String8 out_s8 = String8(); 5221 5222 Mutex::Autolock _l(mLock); 5223 if (initCheck() != NO_ERROR) { 5224 return out_s8; 5225 } 5226 5227 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5228 out_s8 = String8(s); 5229 free(s); 5230 return out_s8; 5231} 5232 5233void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5234 AudioSystem::OutputDescriptor desc; 5235 void *param2 = NULL; 5236 5237 switch (event) { 5238 case AudioSystem::INPUT_OPENED: 5239 case AudioSystem::INPUT_CONFIG_CHANGED: 5240 desc.channels = mChannelMask; 5241 desc.samplingRate = mSampleRate; 5242 desc.format = mFormat; 5243 desc.frameCount = mFrameCount; 5244 desc.latency = 0; 5245 param2 = &desc; 5246 break; 5247 5248 case AudioSystem::INPUT_CLOSED: 5249 default: 5250 break; 5251 } 5252 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5253} 5254 5255void AudioFlinger::RecordThread::readInputParameters() 5256{ 5257 delete mRsmpInBuffer; 5258 // mRsmpInBuffer is always assigned a new[] below 5259 delete mRsmpOutBuffer; 5260 mRsmpOutBuffer = NULL; 5261 delete mResampler; 5262 mResampler = NULL; 5263 5264 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5265 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5266 mChannelCount = (uint16_t)popcount(mChannelMask); 5267 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5268 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5269 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5270 mFrameCount = mInputBytes / mFrameSize; 5271 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5272 5273 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5274 { 5275 int channelCount; 5276 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5277 // stereo to mono post process as the resampler always outputs stereo. 5278 if (mChannelCount == 1 && mReqChannelCount == 2) { 5279 channelCount = 1; 5280 } else { 5281 channelCount = 2; 5282 } 5283 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5284 mResampler->setSampleRate(mSampleRate); 5285 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5286 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5287 5288 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5289 if (mChannelCount == 1 && mReqChannelCount == 1) { 5290 mFrameCount >>= 1; 5291 } 5292 5293 } 5294 mRsmpInIndex = mFrameCount; 5295} 5296 5297unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5298{ 5299 Mutex::Autolock _l(mLock); 5300 if (initCheck() != NO_ERROR) { 5301 return 0; 5302 } 5303 5304 return mInput->stream->get_input_frames_lost(mInput->stream); 5305} 5306 5307uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5308{ 5309 Mutex::Autolock _l(mLock); 5310 uint32_t result = 0; 5311 if (getEffectChain_l(sessionId) != 0) { 5312 result = EFFECT_SESSION; 5313 } 5314 5315 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5316 result |= TRACK_SESSION; 5317 } 5318 5319 return result; 5320} 5321 5322AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5323{ 5324 Mutex::Autolock _l(mLock); 5325 return mTrack; 5326} 5327 5328AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5329{ 5330 Mutex::Autolock _l(mLock); 5331 return mInput; 5332} 5333 5334AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5335{ 5336 Mutex::Autolock _l(mLock); 5337 AudioStreamIn *input = mInput; 5338 mInput = NULL; 5339 return input; 5340} 5341 5342// this method must always be called either with ThreadBase mLock held or inside the thread loop 5343audio_stream_t* AudioFlinger::RecordThread::stream() 5344{ 5345 if (mInput == NULL) { 5346 return NULL; 5347 } 5348 return &mInput->stream->common; 5349} 5350 5351 5352// ---------------------------------------------------------------------------- 5353 5354audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5355 uint32_t *pSamplingRate, 5356 audio_format_t *pFormat, 5357 uint32_t *pChannels, 5358 uint32_t *pLatencyMs, 5359 audio_policy_output_flags_t flags) 5360{ 5361 status_t status; 5362 PlaybackThread *thread = NULL; 5363 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5364 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5365 uint32_t channels = pChannels ? *pChannels : 0; 5366 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5367 audio_stream_out_t *outStream; 5368 audio_hw_device_t *outHwDev; 5369 5370 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5371 pDevices ? *pDevices : 0, 5372 samplingRate, 5373 format, 5374 channels, 5375 flags); 5376 5377 if (pDevices == NULL || *pDevices == 0) { 5378 return 0; 5379 } 5380 5381 Mutex::Autolock _l(mLock); 5382 5383 outHwDev = findSuitableHwDev_l(*pDevices); 5384 if (outHwDev == NULL) 5385 return 0; 5386 5387 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5388 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5389 &channels, &samplingRate, &outStream); 5390 mHardwareStatus = AUDIO_HW_IDLE; 5391 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5392 outStream, 5393 samplingRate, 5394 format, 5395 channels, 5396 status); 5397 5398 if (outStream != NULL) { 5399 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5400 audio_io_handle_t id = nextUniqueId(); 5401 5402 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5403 (format != AUDIO_FORMAT_PCM_16_BIT) || 5404 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5405 thread = new DirectOutputThread(this, output, id, *pDevices); 5406 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5407 } else { 5408 thread = new MixerThread(this, output, id, *pDevices); 5409 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5410 } 5411 mPlaybackThreads.add(id, thread); 5412 5413 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5414 if (pFormat != NULL) *pFormat = format; 5415 if (pChannels != NULL) *pChannels = channels; 5416 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5417 5418 // notify client processes of the new output creation 5419 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5420 return id; 5421 } 5422 5423 return 0; 5424} 5425 5426audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5427 audio_io_handle_t output2) 5428{ 5429 Mutex::Autolock _l(mLock); 5430 MixerThread *thread1 = checkMixerThread_l(output1); 5431 MixerThread *thread2 = checkMixerThread_l(output2); 5432 5433 if (thread1 == NULL || thread2 == NULL) { 5434 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5435 return 0; 5436 } 5437 5438 audio_io_handle_t id = nextUniqueId(); 5439 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5440 thread->addOutputTrack(thread2); 5441 mPlaybackThreads.add(id, thread); 5442 // notify client processes of the new output creation 5443 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5444 return id; 5445} 5446 5447status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5448{ 5449 // keep strong reference on the playback thread so that 5450 // it is not destroyed while exit() is executed 5451 sp <PlaybackThread> thread; 5452 { 5453 Mutex::Autolock _l(mLock); 5454 thread = checkPlaybackThread_l(output); 5455 if (thread == NULL) { 5456 return BAD_VALUE; 5457 } 5458 5459 ALOGV("closeOutput() %d", output); 5460 5461 if (thread->type() == ThreadBase::MIXER) { 5462 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5463 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5464 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5465 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5466 } 5467 } 5468 } 5469 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5470 mPlaybackThreads.removeItem(output); 5471 } 5472 thread->exit(); 5473 // The thread entity (active unit of execution) is no longer running here, 5474 // but the ThreadBase container still exists. 5475 5476 if (thread->type() != ThreadBase::DUPLICATING) { 5477 AudioStreamOut *out = thread->clearOutput(); 5478 assert(out != NULL); 5479 // from now on thread->mOutput is NULL 5480 out->hwDev->close_output_stream(out->hwDev, out->stream); 5481 delete out; 5482 } 5483 return NO_ERROR; 5484} 5485 5486status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5487{ 5488 Mutex::Autolock _l(mLock); 5489 PlaybackThread *thread = checkPlaybackThread_l(output); 5490 5491 if (thread == NULL) { 5492 return BAD_VALUE; 5493 } 5494 5495 ALOGV("suspendOutput() %d", output); 5496 thread->suspend(); 5497 5498 return NO_ERROR; 5499} 5500 5501status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5502{ 5503 Mutex::Autolock _l(mLock); 5504 PlaybackThread *thread = checkPlaybackThread_l(output); 5505 5506 if (thread == NULL) { 5507 return BAD_VALUE; 5508 } 5509 5510 ALOGV("restoreOutput() %d", output); 5511 5512 thread->restore(); 5513 5514 return NO_ERROR; 5515} 5516 5517audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5518 uint32_t *pSamplingRate, 5519 audio_format_t *pFormat, 5520 uint32_t *pChannels, 5521 audio_in_acoustics_t acoustics) 5522{ 5523 status_t status; 5524 RecordThread *thread = NULL; 5525 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5526 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5527 uint32_t channels = pChannels ? *pChannels : 0; 5528 uint32_t reqSamplingRate = samplingRate; 5529 audio_format_t reqFormat = format; 5530 uint32_t reqChannels = channels; 5531 audio_stream_in_t *inStream; 5532 audio_hw_device_t *inHwDev; 5533 5534 if (pDevices == NULL || *pDevices == 0) { 5535 return 0; 5536 } 5537 5538 Mutex::Autolock _l(mLock); 5539 5540 inHwDev = findSuitableHwDev_l(*pDevices); 5541 if (inHwDev == NULL) 5542 return 0; 5543 5544 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5545 &channels, &samplingRate, 5546 acoustics, 5547 &inStream); 5548 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5549 inStream, 5550 samplingRate, 5551 format, 5552 channels, 5553 acoustics, 5554 status); 5555 5556 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5557 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5558 // or stereo to mono conversions on 16 bit PCM inputs. 5559 if (inStream == NULL && status == BAD_VALUE && 5560 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5561 (samplingRate <= 2 * reqSamplingRate) && 5562 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5563 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5564 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5565 &channels, &samplingRate, 5566 acoustics, 5567 &inStream); 5568 } 5569 5570 if (inStream != NULL) { 5571 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5572 5573 audio_io_handle_t id = nextUniqueId(); 5574 // Start record thread 5575 // RecorThread require both input and output device indication to forward to audio 5576 // pre processing modules 5577 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5578 thread = new RecordThread(this, 5579 input, 5580 reqSamplingRate, 5581 reqChannels, 5582 id, 5583 device); 5584 mRecordThreads.add(id, thread); 5585 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5586 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5587 if (pFormat != NULL) *pFormat = format; 5588 if (pChannels != NULL) *pChannels = reqChannels; 5589 5590 input->stream->common.standby(&input->stream->common); 5591 5592 // notify client processes of the new input creation 5593 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5594 return id; 5595 } 5596 5597 return 0; 5598} 5599 5600status_t AudioFlinger::closeInput(audio_io_handle_t input) 5601{ 5602 // keep strong reference on the record thread so that 5603 // it is not destroyed while exit() is executed 5604 sp <RecordThread> thread; 5605 { 5606 Mutex::Autolock _l(mLock); 5607 thread = checkRecordThread_l(input); 5608 if (thread == NULL) { 5609 return BAD_VALUE; 5610 } 5611 5612 ALOGV("closeInput() %d", input); 5613 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5614 mRecordThreads.removeItem(input); 5615 } 5616 thread->exit(); 5617 // The thread entity (active unit of execution) is no longer running here, 5618 // but the ThreadBase container still exists. 5619 5620 AudioStreamIn *in = thread->clearInput(); 5621 assert(in != NULL); 5622 // from now on thread->mInput is NULL 5623 in->hwDev->close_input_stream(in->hwDev, in->stream); 5624 delete in; 5625 5626 return NO_ERROR; 5627} 5628 5629status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5630{ 5631 Mutex::Autolock _l(mLock); 5632 MixerThread *dstThread = checkMixerThread_l(output); 5633 if (dstThread == NULL) { 5634 ALOGW("setStreamOutput() bad output id %d", output); 5635 return BAD_VALUE; 5636 } 5637 5638 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5639 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5640 5641 dstThread->setStreamValid(stream, true); 5642 5643 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5644 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5645 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5646 MixerThread *srcThread = (MixerThread *)thread; 5647 srcThread->setStreamValid(stream, false); 5648 srcThread->invalidateTracks(stream); 5649 } 5650 } 5651 5652 return NO_ERROR; 5653} 5654 5655 5656int AudioFlinger::newAudioSessionId() 5657{ 5658 return nextUniqueId(); 5659} 5660 5661void AudioFlinger::acquireAudioSessionId(int audioSession) 5662{ 5663 Mutex::Autolock _l(mLock); 5664 pid_t caller = IPCThreadState::self()->getCallingPid(); 5665 ALOGV("acquiring %d from %d", audioSession, caller); 5666 size_t num = mAudioSessionRefs.size(); 5667 for (size_t i = 0; i< num; i++) { 5668 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5669 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5670 ref->mCnt++; 5671 ALOGV(" incremented refcount to %d", ref->mCnt); 5672 return; 5673 } 5674 } 5675 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5676 ALOGV(" added new entry for %d", audioSession); 5677} 5678 5679void AudioFlinger::releaseAudioSessionId(int audioSession) 5680{ 5681 Mutex::Autolock _l(mLock); 5682 pid_t caller = IPCThreadState::self()->getCallingPid(); 5683 ALOGV("releasing %d from %d", audioSession, caller); 5684 size_t num = mAudioSessionRefs.size(); 5685 for (size_t i = 0; i< num; i++) { 5686 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5687 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5688 ref->mCnt--; 5689 ALOGV(" decremented refcount to %d", ref->mCnt); 5690 if (ref->mCnt == 0) { 5691 mAudioSessionRefs.removeAt(i); 5692 delete ref; 5693 purgeStaleEffects_l(); 5694 } 5695 return; 5696 } 5697 } 5698 ALOGW("session id %d not found for pid %d", audioSession, caller); 5699} 5700 5701void AudioFlinger::purgeStaleEffects_l() { 5702 5703 ALOGV("purging stale effects"); 5704 5705 Vector< sp<EffectChain> > chains; 5706 5707 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5708 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5709 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5710 sp<EffectChain> ec = t->mEffectChains[j]; 5711 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5712 chains.push(ec); 5713 } 5714 } 5715 } 5716 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5717 sp<RecordThread> t = mRecordThreads.valueAt(i); 5718 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5719 sp<EffectChain> ec = t->mEffectChains[j]; 5720 chains.push(ec); 5721 } 5722 } 5723 5724 for (size_t i = 0; i < chains.size(); i++) { 5725 sp<EffectChain> ec = chains[i]; 5726 int sessionid = ec->sessionId(); 5727 sp<ThreadBase> t = ec->mThread.promote(); 5728 if (t == 0) { 5729 continue; 5730 } 5731 size_t numsessionrefs = mAudioSessionRefs.size(); 5732 bool found = false; 5733 for (size_t k = 0; k < numsessionrefs; k++) { 5734 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5735 if (ref->mSessionid == sessionid) { 5736 ALOGV(" session %d still exists for %d with %d refs", 5737 sessionid, ref->mPid, ref->mCnt); 5738 found = true; 5739 break; 5740 } 5741 } 5742 if (!found) { 5743 // remove all effects from the chain 5744 while (ec->mEffects.size()) { 5745 sp<EffectModule> effect = ec->mEffects[0]; 5746 effect->unPin(); 5747 Mutex::Autolock _l (t->mLock); 5748 t->removeEffect_l(effect); 5749 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5750 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5751 if (handle != 0) { 5752 handle->mEffect.clear(); 5753 if (handle->mHasControl && handle->mEnabled) { 5754 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5755 } 5756 } 5757 } 5758 AudioSystem::unregisterEffect(effect->id()); 5759 } 5760 } 5761 } 5762 return; 5763} 5764 5765// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5766AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5767{ 5768 return mPlaybackThreads.valueFor(output).get(); 5769} 5770 5771// checkMixerThread_l() must be called with AudioFlinger::mLock held 5772AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5773{ 5774 PlaybackThread *thread = checkPlaybackThread_l(output); 5775 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5776} 5777 5778// checkRecordThread_l() must be called with AudioFlinger::mLock held 5779AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5780{ 5781 return mRecordThreads.valueFor(input).get(); 5782} 5783 5784uint32_t AudioFlinger::nextUniqueId() 5785{ 5786 return android_atomic_inc(&mNextUniqueId); 5787} 5788 5789AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5790{ 5791 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5792 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5793 AudioStreamOut *output = thread->getOutput(); 5794 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5795 return thread; 5796 } 5797 } 5798 return NULL; 5799} 5800 5801uint32_t AudioFlinger::primaryOutputDevice_l() const 5802{ 5803 PlaybackThread *thread = primaryPlaybackThread_l(); 5804 5805 if (thread == NULL) { 5806 return 0; 5807 } 5808 5809 return thread->device(); 5810} 5811 5812 5813// ---------------------------------------------------------------------------- 5814// Effect management 5815// ---------------------------------------------------------------------------- 5816 5817 5818status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5819{ 5820 Mutex::Autolock _l(mLock); 5821 return EffectQueryNumberEffects(numEffects); 5822} 5823 5824status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5825{ 5826 Mutex::Autolock _l(mLock); 5827 return EffectQueryEffect(index, descriptor); 5828} 5829 5830status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5831 effect_descriptor_t *descriptor) const 5832{ 5833 Mutex::Autolock _l(mLock); 5834 return EffectGetDescriptor(pUuid, descriptor); 5835} 5836 5837 5838sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5839 effect_descriptor_t *pDesc, 5840 const sp<IEffectClient>& effectClient, 5841 int32_t priority, 5842 audio_io_handle_t io, 5843 int sessionId, 5844 status_t *status, 5845 int *id, 5846 int *enabled) 5847{ 5848 status_t lStatus = NO_ERROR; 5849 sp<EffectHandle> handle; 5850 effect_descriptor_t desc; 5851 5852 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5853 pid, effectClient.get(), priority, sessionId, io); 5854 5855 if (pDesc == NULL) { 5856 lStatus = BAD_VALUE; 5857 goto Exit; 5858 } 5859 5860 // check audio settings permission for global effects 5861 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5862 lStatus = PERMISSION_DENIED; 5863 goto Exit; 5864 } 5865 5866 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5867 // that can only be created by audio policy manager (running in same process) 5868 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5869 lStatus = PERMISSION_DENIED; 5870 goto Exit; 5871 } 5872 5873 if (io == 0) { 5874 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5875 // output must be specified by AudioPolicyManager when using session 5876 // AUDIO_SESSION_OUTPUT_STAGE 5877 lStatus = BAD_VALUE; 5878 goto Exit; 5879 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5880 // if the output returned by getOutputForEffect() is removed before we lock the 5881 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5882 // and we will exit safely 5883 io = AudioSystem::getOutputForEffect(&desc); 5884 } 5885 } 5886 5887 { 5888 Mutex::Autolock _l(mLock); 5889 5890 5891 if (!EffectIsNullUuid(&pDesc->uuid)) { 5892 // if uuid is specified, request effect descriptor 5893 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5894 if (lStatus < 0) { 5895 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5896 goto Exit; 5897 } 5898 } else { 5899 // if uuid is not specified, look for an available implementation 5900 // of the required type in effect factory 5901 if (EffectIsNullUuid(&pDesc->type)) { 5902 ALOGW("createEffect() no effect type"); 5903 lStatus = BAD_VALUE; 5904 goto Exit; 5905 } 5906 uint32_t numEffects = 0; 5907 effect_descriptor_t d; 5908 d.flags = 0; // prevent compiler warning 5909 bool found = false; 5910 5911 lStatus = EffectQueryNumberEffects(&numEffects); 5912 if (lStatus < 0) { 5913 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5914 goto Exit; 5915 } 5916 for (uint32_t i = 0; i < numEffects; i++) { 5917 lStatus = EffectQueryEffect(i, &desc); 5918 if (lStatus < 0) { 5919 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5920 continue; 5921 } 5922 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5923 // If matching type found save effect descriptor. If the session is 5924 // 0 and the effect is not auxiliary, continue enumeration in case 5925 // an auxiliary version of this effect type is available 5926 found = true; 5927 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5928 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5929 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5930 break; 5931 } 5932 } 5933 } 5934 if (!found) { 5935 lStatus = BAD_VALUE; 5936 ALOGW("createEffect() effect not found"); 5937 goto Exit; 5938 } 5939 // For same effect type, chose auxiliary version over insert version if 5940 // connect to output mix (Compliance to OpenSL ES) 5941 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5942 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5943 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5944 } 5945 } 5946 5947 // Do not allow auxiliary effects on a session different from 0 (output mix) 5948 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5949 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5950 lStatus = INVALID_OPERATION; 5951 goto Exit; 5952 } 5953 5954 // check recording permission for visualizer 5955 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5956 !recordingAllowed()) { 5957 lStatus = PERMISSION_DENIED; 5958 goto Exit; 5959 } 5960 5961 // return effect descriptor 5962 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5963 5964 // If output is not specified try to find a matching audio session ID in one of the 5965 // output threads. 5966 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5967 // because of code checking output when entering the function. 5968 // Note: io is never 0 when creating an effect on an input 5969 if (io == 0) { 5970 // look for the thread where the specified audio session is present 5971 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5972 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5973 io = mPlaybackThreads.keyAt(i); 5974 break; 5975 } 5976 } 5977 if (io == 0) { 5978 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5979 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5980 io = mRecordThreads.keyAt(i); 5981 break; 5982 } 5983 } 5984 } 5985 // If no output thread contains the requested session ID, default to 5986 // first output. The effect chain will be moved to the correct output 5987 // thread when a track with the same session ID is created 5988 if (io == 0 && mPlaybackThreads.size()) { 5989 io = mPlaybackThreads.keyAt(0); 5990 } 5991 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5992 } 5993 ThreadBase *thread = checkRecordThread_l(io); 5994 if (thread == NULL) { 5995 thread = checkPlaybackThread_l(io); 5996 if (thread == NULL) { 5997 ALOGE("createEffect() unknown output thread"); 5998 lStatus = BAD_VALUE; 5999 goto Exit; 6000 } 6001 } 6002 6003 sp<Client> client = registerPid_l(pid); 6004 6005 // create effect on selected output thread 6006 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6007 &desc, enabled, &lStatus); 6008 if (handle != 0 && id != NULL) { 6009 *id = handle->id(); 6010 } 6011 } 6012 6013Exit: 6014 if(status) { 6015 *status = lStatus; 6016 } 6017 return handle; 6018} 6019 6020status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6021 audio_io_handle_t dstOutput) 6022{ 6023 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6024 sessionId, srcOutput, dstOutput); 6025 Mutex::Autolock _l(mLock); 6026 if (srcOutput == dstOutput) { 6027 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6028 return NO_ERROR; 6029 } 6030 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6031 if (srcThread == NULL) { 6032 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6033 return BAD_VALUE; 6034 } 6035 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6036 if (dstThread == NULL) { 6037 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6038 return BAD_VALUE; 6039 } 6040 6041 Mutex::Autolock _dl(dstThread->mLock); 6042 Mutex::Autolock _sl(srcThread->mLock); 6043 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6044 6045 return NO_ERROR; 6046} 6047 6048// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6049status_t AudioFlinger::moveEffectChain_l(int sessionId, 6050 AudioFlinger::PlaybackThread *srcThread, 6051 AudioFlinger::PlaybackThread *dstThread, 6052 bool reRegister) 6053{ 6054 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6055 sessionId, srcThread, dstThread); 6056 6057 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6058 if (chain == 0) { 6059 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6060 sessionId, srcThread); 6061 return INVALID_OPERATION; 6062 } 6063 6064 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6065 // so that a new chain is created with correct parameters when first effect is added. This is 6066 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6067 // removed. 6068 srcThread->removeEffectChain_l(chain); 6069 6070 // transfer all effects one by one so that new effect chain is created on new thread with 6071 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6072 audio_io_handle_t dstOutput = dstThread->id(); 6073 sp<EffectChain> dstChain; 6074 uint32_t strategy = 0; // prevent compiler warning 6075 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6076 while (effect != 0) { 6077 srcThread->removeEffect_l(effect); 6078 dstThread->addEffect_l(effect); 6079 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6080 if (effect->state() == EffectModule::ACTIVE || 6081 effect->state() == EffectModule::STOPPING) { 6082 effect->start(); 6083 } 6084 // if the move request is not received from audio policy manager, the effect must be 6085 // re-registered with the new strategy and output 6086 if (dstChain == 0) { 6087 dstChain = effect->chain().promote(); 6088 if (dstChain == 0) { 6089 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6090 srcThread->addEffect_l(effect); 6091 return NO_INIT; 6092 } 6093 strategy = dstChain->strategy(); 6094 } 6095 if (reRegister) { 6096 AudioSystem::unregisterEffect(effect->id()); 6097 AudioSystem::registerEffect(&effect->desc(), 6098 dstOutput, 6099 strategy, 6100 sessionId, 6101 effect->id()); 6102 } 6103 effect = chain->getEffectFromId_l(0); 6104 } 6105 6106 return NO_ERROR; 6107} 6108 6109 6110// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6111sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6112 const sp<AudioFlinger::Client>& client, 6113 const sp<IEffectClient>& effectClient, 6114 int32_t priority, 6115 int sessionId, 6116 effect_descriptor_t *desc, 6117 int *enabled, 6118 status_t *status 6119 ) 6120{ 6121 sp<EffectModule> effect; 6122 sp<EffectHandle> handle; 6123 status_t lStatus; 6124 sp<EffectChain> chain; 6125 bool chainCreated = false; 6126 bool effectCreated = false; 6127 bool effectRegistered = false; 6128 6129 lStatus = initCheck(); 6130 if (lStatus != NO_ERROR) { 6131 ALOGW("createEffect_l() Audio driver not initialized."); 6132 goto Exit; 6133 } 6134 6135 // Do not allow effects with session ID 0 on direct output or duplicating threads 6136 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6137 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6138 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6139 desc->name, sessionId); 6140 lStatus = BAD_VALUE; 6141 goto Exit; 6142 } 6143 // Only Pre processor effects are allowed on input threads and only on input threads 6144 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6145 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6146 desc->name, desc->flags, mType); 6147 lStatus = BAD_VALUE; 6148 goto Exit; 6149 } 6150 6151 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6152 6153 { // scope for mLock 6154 Mutex::Autolock _l(mLock); 6155 6156 // check for existing effect chain with the requested audio session 6157 chain = getEffectChain_l(sessionId); 6158 if (chain == 0) { 6159 // create a new chain for this session 6160 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6161 chain = new EffectChain(this, sessionId); 6162 addEffectChain_l(chain); 6163 chain->setStrategy(getStrategyForSession_l(sessionId)); 6164 chainCreated = true; 6165 } else { 6166 effect = chain->getEffectFromDesc_l(desc); 6167 } 6168 6169 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6170 6171 if (effect == 0) { 6172 int id = mAudioFlinger->nextUniqueId(); 6173 // Check CPU and memory usage 6174 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6175 if (lStatus != NO_ERROR) { 6176 goto Exit; 6177 } 6178 effectRegistered = true; 6179 // create a new effect module if none present in the chain 6180 effect = new EffectModule(this, chain, desc, id, sessionId); 6181 lStatus = effect->status(); 6182 if (lStatus != NO_ERROR) { 6183 goto Exit; 6184 } 6185 lStatus = chain->addEffect_l(effect); 6186 if (lStatus != NO_ERROR) { 6187 goto Exit; 6188 } 6189 effectCreated = true; 6190 6191 effect->setDevice(mDevice); 6192 effect->setMode(mAudioFlinger->getMode()); 6193 } 6194 // create effect handle and connect it to effect module 6195 handle = new EffectHandle(effect, client, effectClient, priority); 6196 lStatus = effect->addHandle(handle); 6197 if (enabled != NULL) { 6198 *enabled = (int)effect->isEnabled(); 6199 } 6200 } 6201 6202Exit: 6203 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6204 Mutex::Autolock _l(mLock); 6205 if (effectCreated) { 6206 chain->removeEffect_l(effect); 6207 } 6208 if (effectRegistered) { 6209 AudioSystem::unregisterEffect(effect->id()); 6210 } 6211 if (chainCreated) { 6212 removeEffectChain_l(chain); 6213 } 6214 handle.clear(); 6215 } 6216 6217 if(status) { 6218 *status = lStatus; 6219 } 6220 return handle; 6221} 6222 6223sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6224{ 6225 sp<EffectChain> chain = getEffectChain_l(sessionId); 6226 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6227} 6228 6229// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6230// PlaybackThread::mLock held 6231status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6232{ 6233 // check for existing effect chain with the requested audio session 6234 int sessionId = effect->sessionId(); 6235 sp<EffectChain> chain = getEffectChain_l(sessionId); 6236 bool chainCreated = false; 6237 6238 if (chain == 0) { 6239 // create a new chain for this session 6240 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6241 chain = new EffectChain(this, sessionId); 6242 addEffectChain_l(chain); 6243 chain->setStrategy(getStrategyForSession_l(sessionId)); 6244 chainCreated = true; 6245 } 6246 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6247 6248 if (chain->getEffectFromId_l(effect->id()) != 0) { 6249 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6250 this, effect->desc().name, chain.get()); 6251 return BAD_VALUE; 6252 } 6253 6254 status_t status = chain->addEffect_l(effect); 6255 if (status != NO_ERROR) { 6256 if (chainCreated) { 6257 removeEffectChain_l(chain); 6258 } 6259 return status; 6260 } 6261 6262 effect->setDevice(mDevice); 6263 effect->setMode(mAudioFlinger->getMode()); 6264 return NO_ERROR; 6265} 6266 6267void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6268 6269 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6270 effect_descriptor_t desc = effect->desc(); 6271 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6272 detachAuxEffect_l(effect->id()); 6273 } 6274 6275 sp<EffectChain> chain = effect->chain().promote(); 6276 if (chain != 0) { 6277 // remove effect chain if removing last effect 6278 if (chain->removeEffect_l(effect) == 0) { 6279 removeEffectChain_l(chain); 6280 } 6281 } else { 6282 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6283 } 6284} 6285 6286void AudioFlinger::ThreadBase::lockEffectChains_l( 6287 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6288{ 6289 effectChains = mEffectChains; 6290 for (size_t i = 0; i < mEffectChains.size(); i++) { 6291 mEffectChains[i]->lock(); 6292 } 6293} 6294 6295void AudioFlinger::ThreadBase::unlockEffectChains( 6296 const Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6297{ 6298 for (size_t i = 0; i < effectChains.size(); i++) { 6299 effectChains[i]->unlock(); 6300 } 6301} 6302 6303sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6304{ 6305 Mutex::Autolock _l(mLock); 6306 return getEffectChain_l(sessionId); 6307} 6308 6309sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6310{ 6311 size_t size = mEffectChains.size(); 6312 for (size_t i = 0; i < size; i++) { 6313 if (mEffectChains[i]->sessionId() == sessionId) { 6314 return mEffectChains[i]; 6315 } 6316 } 6317 return 0; 6318} 6319 6320void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6321{ 6322 Mutex::Autolock _l(mLock); 6323 size_t size = mEffectChains.size(); 6324 for (size_t i = 0; i < size; i++) { 6325 mEffectChains[i]->setMode_l(mode); 6326 } 6327} 6328 6329void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6330 const wp<EffectHandle>& handle, 6331 bool unpinIfLast) { 6332 6333 Mutex::Autolock _l(mLock); 6334 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6335 // delete the effect module if removing last handle on it 6336 if (effect->removeHandle(handle) == 0) { 6337 if (!effect->isPinned() || unpinIfLast) { 6338 removeEffect_l(effect); 6339 AudioSystem::unregisterEffect(effect->id()); 6340 } 6341 } 6342} 6343 6344status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6345{ 6346 int session = chain->sessionId(); 6347 int16_t *buffer = mMixBuffer; 6348 bool ownsBuffer = false; 6349 6350 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6351 if (session > 0) { 6352 // Only one effect chain can be present in direct output thread and it uses 6353 // the mix buffer as input 6354 if (mType != DIRECT) { 6355 size_t numSamples = mFrameCount * mChannelCount; 6356 buffer = new int16_t[numSamples]; 6357 memset(buffer, 0, numSamples * sizeof(int16_t)); 6358 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6359 ownsBuffer = true; 6360 } 6361 6362 // Attach all tracks with same session ID to this chain. 6363 for (size_t i = 0; i < mTracks.size(); ++i) { 6364 sp<Track> track = mTracks[i]; 6365 if (session == track->sessionId()) { 6366 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6367 track->setMainBuffer(buffer); 6368 chain->incTrackCnt(); 6369 } 6370 } 6371 6372 // indicate all active tracks in the chain 6373 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6374 sp<Track> track = mActiveTracks[i].promote(); 6375 if (track == 0) continue; 6376 if (session == track->sessionId()) { 6377 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6378 chain->incActiveTrackCnt(); 6379 } 6380 } 6381 } 6382 6383 chain->setInBuffer(buffer, ownsBuffer); 6384 chain->setOutBuffer(mMixBuffer); 6385 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6386 // chains list in order to be processed last as it contains output stage effects 6387 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6388 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6389 // after track specific effects and before output stage 6390 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6391 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6392 // Effect chain for other sessions are inserted at beginning of effect 6393 // chains list to be processed before output mix effects. Relative order between other 6394 // sessions is not important 6395 size_t size = mEffectChains.size(); 6396 size_t i = 0; 6397 for (i = 0; i < size; i++) { 6398 if (mEffectChains[i]->sessionId() < session) break; 6399 } 6400 mEffectChains.insertAt(chain, i); 6401 checkSuspendOnAddEffectChain_l(chain); 6402 6403 return NO_ERROR; 6404} 6405 6406size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6407{ 6408 int session = chain->sessionId(); 6409 6410 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6411 6412 for (size_t i = 0; i < mEffectChains.size(); i++) { 6413 if (chain == mEffectChains[i]) { 6414 mEffectChains.removeAt(i); 6415 // detach all active tracks from the chain 6416 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6417 sp<Track> track = mActiveTracks[i].promote(); 6418 if (track == 0) continue; 6419 if (session == track->sessionId()) { 6420 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6421 chain.get(), session); 6422 chain->decActiveTrackCnt(); 6423 } 6424 } 6425 6426 // detach all tracks with same session ID from this chain 6427 for (size_t i = 0; i < mTracks.size(); ++i) { 6428 sp<Track> track = mTracks[i]; 6429 if (session == track->sessionId()) { 6430 track->setMainBuffer(mMixBuffer); 6431 chain->decTrackCnt(); 6432 } 6433 } 6434 break; 6435 } 6436 } 6437 return mEffectChains.size(); 6438} 6439 6440status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6441 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6442{ 6443 Mutex::Autolock _l(mLock); 6444 return attachAuxEffect_l(track, EffectId); 6445} 6446 6447status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6448 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6449{ 6450 status_t status = NO_ERROR; 6451 6452 if (EffectId == 0) { 6453 track->setAuxBuffer(0, NULL); 6454 } else { 6455 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6456 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6457 if (effect != 0) { 6458 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6459 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6460 } else { 6461 status = INVALID_OPERATION; 6462 } 6463 } else { 6464 status = BAD_VALUE; 6465 } 6466 } 6467 return status; 6468} 6469 6470void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6471{ 6472 for (size_t i = 0; i < mTracks.size(); ++i) { 6473 sp<Track> track = mTracks[i]; 6474 if (track->auxEffectId() == effectId) { 6475 attachAuxEffect_l(track, 0); 6476 } 6477 } 6478} 6479 6480status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6481{ 6482 // only one chain per input thread 6483 if (mEffectChains.size() != 0) { 6484 return INVALID_OPERATION; 6485 } 6486 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6487 6488 chain->setInBuffer(NULL); 6489 chain->setOutBuffer(NULL); 6490 6491 checkSuspendOnAddEffectChain_l(chain); 6492 6493 mEffectChains.add(chain); 6494 6495 return NO_ERROR; 6496} 6497 6498size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6499{ 6500 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6501 ALOGW_IF(mEffectChains.size() != 1, 6502 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6503 chain.get(), mEffectChains.size(), this); 6504 if (mEffectChains.size() == 1) { 6505 mEffectChains.removeAt(0); 6506 } 6507 return 0; 6508} 6509 6510// ---------------------------------------------------------------------------- 6511// EffectModule implementation 6512// ---------------------------------------------------------------------------- 6513 6514#undef LOG_TAG 6515#define LOG_TAG "AudioFlinger::EffectModule" 6516 6517AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6518 const wp<AudioFlinger::EffectChain>& chain, 6519 effect_descriptor_t *desc, 6520 int id, 6521 int sessionId) 6522 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6523 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6524{ 6525 ALOGV("Constructor %p", this); 6526 int lStatus; 6527 if (thread == NULL) { 6528 return; 6529 } 6530 6531 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6532 6533 // create effect engine from effect factory 6534 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6535 6536 if (mStatus != NO_ERROR) { 6537 return; 6538 } 6539 lStatus = init(); 6540 if (lStatus < 0) { 6541 mStatus = lStatus; 6542 goto Error; 6543 } 6544 6545 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6546 mPinned = true; 6547 } 6548 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6549 return; 6550Error: 6551 EffectRelease(mEffectInterface); 6552 mEffectInterface = NULL; 6553 ALOGV("Constructor Error %d", mStatus); 6554} 6555 6556AudioFlinger::EffectModule::~EffectModule() 6557{ 6558 ALOGV("Destructor %p", this); 6559 if (mEffectInterface != NULL) { 6560 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6561 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6562 sp<ThreadBase> thread = mThread.promote(); 6563 if (thread != 0) { 6564 audio_stream_t *stream = thread->stream(); 6565 if (stream != NULL) { 6566 stream->remove_audio_effect(stream, mEffectInterface); 6567 } 6568 } 6569 } 6570 // release effect engine 6571 EffectRelease(mEffectInterface); 6572 } 6573} 6574 6575status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6576{ 6577 status_t status; 6578 6579 Mutex::Autolock _l(mLock); 6580 int priority = handle->priority(); 6581 size_t size = mHandles.size(); 6582 sp<EffectHandle> h; 6583 size_t i; 6584 for (i = 0; i < size; i++) { 6585 h = mHandles[i].promote(); 6586 if (h == 0) continue; 6587 if (h->priority() <= priority) break; 6588 } 6589 // if inserted in first place, move effect control from previous owner to this handle 6590 if (i == 0) { 6591 bool enabled = false; 6592 if (h != 0) { 6593 enabled = h->enabled(); 6594 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6595 } 6596 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6597 status = NO_ERROR; 6598 } else { 6599 status = ALREADY_EXISTS; 6600 } 6601 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6602 mHandles.insertAt(handle, i); 6603 return status; 6604} 6605 6606size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6607{ 6608 Mutex::Autolock _l(mLock); 6609 size_t size = mHandles.size(); 6610 size_t i; 6611 for (i = 0; i < size; i++) { 6612 if (mHandles[i] == handle) break; 6613 } 6614 if (i == size) { 6615 return size; 6616 } 6617 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6618 6619 bool enabled = false; 6620 EffectHandle *hdl = handle.unsafe_get(); 6621 if (hdl != NULL) { 6622 ALOGV("removeHandle() unsafe_get OK"); 6623 enabled = hdl->enabled(); 6624 } 6625 mHandles.removeAt(i); 6626 size = mHandles.size(); 6627 // if removed from first place, move effect control from this handle to next in line 6628 if (i == 0 && size != 0) { 6629 sp<EffectHandle> h = mHandles[0].promote(); 6630 if (h != 0) { 6631 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6632 } 6633 } 6634 6635 // Prevent calls to process() and other functions on effect interface from now on. 6636 // The effect engine will be released by the destructor when the last strong reference on 6637 // this object is released which can happen after next process is called. 6638 if (size == 0 && !mPinned) { 6639 mState = DESTROYED; 6640 } 6641 6642 return size; 6643} 6644 6645sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6646{ 6647 Mutex::Autolock _l(mLock); 6648 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6649} 6650 6651void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6652{ 6653 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6654 // keep a strong reference on this EffectModule to avoid calling the 6655 // destructor before we exit 6656 sp<EffectModule> keep(this); 6657 { 6658 sp<ThreadBase> thread = mThread.promote(); 6659 if (thread != 0) { 6660 thread->disconnectEffect(keep, handle, unpinIfLast); 6661 } 6662 } 6663} 6664 6665void AudioFlinger::EffectModule::updateState() { 6666 Mutex::Autolock _l(mLock); 6667 6668 switch (mState) { 6669 case RESTART: 6670 reset_l(); 6671 // FALL THROUGH 6672 6673 case STARTING: 6674 // clear auxiliary effect input buffer for next accumulation 6675 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6676 memset(mConfig.inputCfg.buffer.raw, 6677 0, 6678 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6679 } 6680 start_l(); 6681 mState = ACTIVE; 6682 break; 6683 case STOPPING: 6684 stop_l(); 6685 mDisableWaitCnt = mMaxDisableWaitCnt; 6686 mState = STOPPED; 6687 break; 6688 case STOPPED: 6689 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6690 // turn off sequence. 6691 if (--mDisableWaitCnt == 0) { 6692 reset_l(); 6693 mState = IDLE; 6694 } 6695 break; 6696 default: //IDLE , ACTIVE, DESTROYED 6697 break; 6698 } 6699} 6700 6701void AudioFlinger::EffectModule::process() 6702{ 6703 Mutex::Autolock _l(mLock); 6704 6705 if (mState == DESTROYED || mEffectInterface == NULL || 6706 mConfig.inputCfg.buffer.raw == NULL || 6707 mConfig.outputCfg.buffer.raw == NULL) { 6708 return; 6709 } 6710 6711 if (isProcessEnabled()) { 6712 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6713 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6714 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6715 mConfig.inputCfg.buffer.s32, 6716 mConfig.inputCfg.buffer.frameCount/2); 6717 } 6718 6719 // do the actual processing in the effect engine 6720 int ret = (*mEffectInterface)->process(mEffectInterface, 6721 &mConfig.inputCfg.buffer, 6722 &mConfig.outputCfg.buffer); 6723 6724 // force transition to IDLE state when engine is ready 6725 if (mState == STOPPED && ret == -ENODATA) { 6726 mDisableWaitCnt = 1; 6727 } 6728 6729 // clear auxiliary effect input buffer for next accumulation 6730 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6731 memset(mConfig.inputCfg.buffer.raw, 0, 6732 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6733 } 6734 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6735 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6736 // If an insert effect is idle and input buffer is different from output buffer, 6737 // accumulate input onto output 6738 sp<EffectChain> chain = mChain.promote(); 6739 if (chain != 0 && chain->activeTrackCnt() != 0) { 6740 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6741 int16_t *in = mConfig.inputCfg.buffer.s16; 6742 int16_t *out = mConfig.outputCfg.buffer.s16; 6743 for (size_t i = 0; i < frameCnt; i++) { 6744 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6745 } 6746 } 6747 } 6748} 6749 6750void AudioFlinger::EffectModule::reset_l() 6751{ 6752 if (mEffectInterface == NULL) { 6753 return; 6754 } 6755 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6756} 6757 6758status_t AudioFlinger::EffectModule::configure() 6759{ 6760 uint32_t channels; 6761 if (mEffectInterface == NULL) { 6762 return NO_INIT; 6763 } 6764 6765 sp<ThreadBase> thread = mThread.promote(); 6766 if (thread == 0) { 6767 return DEAD_OBJECT; 6768 } 6769 6770 // TODO: handle configuration of effects replacing track process 6771 if (thread->channelCount() == 1) { 6772 channels = AUDIO_CHANNEL_OUT_MONO; 6773 } else { 6774 channels = AUDIO_CHANNEL_OUT_STEREO; 6775 } 6776 6777 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6778 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6779 } else { 6780 mConfig.inputCfg.channels = channels; 6781 } 6782 mConfig.outputCfg.channels = channels; 6783 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6784 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6785 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6786 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6787 mConfig.inputCfg.bufferProvider.cookie = NULL; 6788 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6789 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6790 mConfig.outputCfg.bufferProvider.cookie = NULL; 6791 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6792 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6793 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6794 // Insert effect: 6795 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6796 // always overwrites output buffer: input buffer == output buffer 6797 // - in other sessions: 6798 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6799 // other effect: overwrites output buffer: input buffer == output buffer 6800 // Auxiliary effect: 6801 // accumulates in output buffer: input buffer != output buffer 6802 // Therefore: accumulate <=> input buffer != output buffer 6803 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6804 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6805 } else { 6806 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6807 } 6808 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6809 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6810 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6811 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6812 6813 ALOGV("configure() %p thread %p buffer %p framecount %d", 6814 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6815 6816 status_t cmdStatus; 6817 uint32_t size = sizeof(int); 6818 status_t status = (*mEffectInterface)->command(mEffectInterface, 6819 EFFECT_CMD_SET_CONFIG, 6820 sizeof(effect_config_t), 6821 &mConfig, 6822 &size, 6823 &cmdStatus); 6824 if (status == 0) { 6825 status = cmdStatus; 6826 } 6827 6828 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6829 (1000 * mConfig.outputCfg.buffer.frameCount); 6830 6831 return status; 6832} 6833 6834status_t AudioFlinger::EffectModule::init() 6835{ 6836 Mutex::Autolock _l(mLock); 6837 if (mEffectInterface == NULL) { 6838 return NO_INIT; 6839 } 6840 status_t cmdStatus; 6841 uint32_t size = sizeof(status_t); 6842 status_t status = (*mEffectInterface)->command(mEffectInterface, 6843 EFFECT_CMD_INIT, 6844 0, 6845 NULL, 6846 &size, 6847 &cmdStatus); 6848 if (status == 0) { 6849 status = cmdStatus; 6850 } 6851 return status; 6852} 6853 6854status_t AudioFlinger::EffectModule::start() 6855{ 6856 Mutex::Autolock _l(mLock); 6857 return start_l(); 6858} 6859 6860status_t AudioFlinger::EffectModule::start_l() 6861{ 6862 if (mEffectInterface == NULL) { 6863 return NO_INIT; 6864 } 6865 status_t cmdStatus; 6866 uint32_t size = sizeof(status_t); 6867 status_t status = (*mEffectInterface)->command(mEffectInterface, 6868 EFFECT_CMD_ENABLE, 6869 0, 6870 NULL, 6871 &size, 6872 &cmdStatus); 6873 if (status == 0) { 6874 status = cmdStatus; 6875 } 6876 if (status == 0 && 6877 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6878 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6879 sp<ThreadBase> thread = mThread.promote(); 6880 if (thread != 0) { 6881 audio_stream_t *stream = thread->stream(); 6882 if (stream != NULL) { 6883 stream->add_audio_effect(stream, mEffectInterface); 6884 } 6885 } 6886 } 6887 return status; 6888} 6889 6890status_t AudioFlinger::EffectModule::stop() 6891{ 6892 Mutex::Autolock _l(mLock); 6893 return stop_l(); 6894} 6895 6896status_t AudioFlinger::EffectModule::stop_l() 6897{ 6898 if (mEffectInterface == NULL) { 6899 return NO_INIT; 6900 } 6901 status_t cmdStatus; 6902 uint32_t size = sizeof(status_t); 6903 status_t status = (*mEffectInterface)->command(mEffectInterface, 6904 EFFECT_CMD_DISABLE, 6905 0, 6906 NULL, 6907 &size, 6908 &cmdStatus); 6909 if (status == 0) { 6910 status = cmdStatus; 6911 } 6912 if (status == 0 && 6913 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6914 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6915 sp<ThreadBase> thread = mThread.promote(); 6916 if (thread != 0) { 6917 audio_stream_t *stream = thread->stream(); 6918 if (stream != NULL) { 6919 stream->remove_audio_effect(stream, mEffectInterface); 6920 } 6921 } 6922 } 6923 return status; 6924} 6925 6926status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6927 uint32_t cmdSize, 6928 void *pCmdData, 6929 uint32_t *replySize, 6930 void *pReplyData) 6931{ 6932 Mutex::Autolock _l(mLock); 6933// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6934 6935 if (mState == DESTROYED || mEffectInterface == NULL) { 6936 return NO_INIT; 6937 } 6938 status_t status = (*mEffectInterface)->command(mEffectInterface, 6939 cmdCode, 6940 cmdSize, 6941 pCmdData, 6942 replySize, 6943 pReplyData); 6944 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6945 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6946 for (size_t i = 1; i < mHandles.size(); i++) { 6947 sp<EffectHandle> h = mHandles[i].promote(); 6948 if (h != 0) { 6949 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6950 } 6951 } 6952 } 6953 return status; 6954} 6955 6956status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6957{ 6958 6959 Mutex::Autolock _l(mLock); 6960 ALOGV("setEnabled %p enabled %d", this, enabled); 6961 6962 if (enabled != isEnabled()) { 6963 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6964 if (enabled && status != NO_ERROR) { 6965 return status; 6966 } 6967 6968 switch (mState) { 6969 // going from disabled to enabled 6970 case IDLE: 6971 mState = STARTING; 6972 break; 6973 case STOPPED: 6974 mState = RESTART; 6975 break; 6976 case STOPPING: 6977 mState = ACTIVE; 6978 break; 6979 6980 // going from enabled to disabled 6981 case RESTART: 6982 mState = STOPPED; 6983 break; 6984 case STARTING: 6985 mState = IDLE; 6986 break; 6987 case ACTIVE: 6988 mState = STOPPING; 6989 break; 6990 case DESTROYED: 6991 return NO_ERROR; // simply ignore as we are being destroyed 6992 } 6993 for (size_t i = 1; i < mHandles.size(); i++) { 6994 sp<EffectHandle> h = mHandles[i].promote(); 6995 if (h != 0) { 6996 h->setEnabled(enabled); 6997 } 6998 } 6999 } 7000 return NO_ERROR; 7001} 7002 7003bool AudioFlinger::EffectModule::isEnabled() const 7004{ 7005 switch (mState) { 7006 case RESTART: 7007 case STARTING: 7008 case ACTIVE: 7009 return true; 7010 case IDLE: 7011 case STOPPING: 7012 case STOPPED: 7013 case DESTROYED: 7014 default: 7015 return false; 7016 } 7017} 7018 7019bool AudioFlinger::EffectModule::isProcessEnabled() const 7020{ 7021 switch (mState) { 7022 case RESTART: 7023 case ACTIVE: 7024 case STOPPING: 7025 case STOPPED: 7026 return true; 7027 case IDLE: 7028 case STARTING: 7029 case DESTROYED: 7030 default: 7031 return false; 7032 } 7033} 7034 7035status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7036{ 7037 Mutex::Autolock _l(mLock); 7038 status_t status = NO_ERROR; 7039 7040 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7041 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7042 if (isProcessEnabled() && 7043 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7044 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7045 status_t cmdStatus; 7046 uint32_t volume[2]; 7047 uint32_t *pVolume = NULL; 7048 uint32_t size = sizeof(volume); 7049 volume[0] = *left; 7050 volume[1] = *right; 7051 if (controller) { 7052 pVolume = volume; 7053 } 7054 status = (*mEffectInterface)->command(mEffectInterface, 7055 EFFECT_CMD_SET_VOLUME, 7056 size, 7057 volume, 7058 &size, 7059 pVolume); 7060 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7061 *left = volume[0]; 7062 *right = volume[1]; 7063 } 7064 } 7065 return status; 7066} 7067 7068status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7069{ 7070 Mutex::Autolock _l(mLock); 7071 status_t status = NO_ERROR; 7072 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7073 // audio pre processing modules on RecordThread can receive both output and 7074 // input device indication in the same call 7075 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7076 if (dev) { 7077 status_t cmdStatus; 7078 uint32_t size = sizeof(status_t); 7079 7080 status = (*mEffectInterface)->command(mEffectInterface, 7081 EFFECT_CMD_SET_DEVICE, 7082 sizeof(uint32_t), 7083 &dev, 7084 &size, 7085 &cmdStatus); 7086 if (status == NO_ERROR) { 7087 status = cmdStatus; 7088 } 7089 } 7090 dev = device & AUDIO_DEVICE_IN_ALL; 7091 if (dev) { 7092 status_t cmdStatus; 7093 uint32_t size = sizeof(status_t); 7094 7095 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7096 EFFECT_CMD_SET_INPUT_DEVICE, 7097 sizeof(uint32_t), 7098 &dev, 7099 &size, 7100 &cmdStatus); 7101 if (status2 == NO_ERROR) { 7102 status2 = cmdStatus; 7103 } 7104 if (status == NO_ERROR) { 7105 status = status2; 7106 } 7107 } 7108 } 7109 return status; 7110} 7111 7112status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7113{ 7114 Mutex::Autolock _l(mLock); 7115 status_t status = NO_ERROR; 7116 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7117 status_t cmdStatus; 7118 uint32_t size = sizeof(status_t); 7119 status = (*mEffectInterface)->command(mEffectInterface, 7120 EFFECT_CMD_SET_AUDIO_MODE, 7121 sizeof(audio_mode_t), 7122 &mode, 7123 &size, 7124 &cmdStatus); 7125 if (status == NO_ERROR) { 7126 status = cmdStatus; 7127 } 7128 } 7129 return status; 7130} 7131 7132void AudioFlinger::EffectModule::setSuspended(bool suspended) 7133{ 7134 Mutex::Autolock _l(mLock); 7135 mSuspended = suspended; 7136} 7137 7138bool AudioFlinger::EffectModule::suspended() const 7139{ 7140 Mutex::Autolock _l(mLock); 7141 return mSuspended; 7142} 7143 7144status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7145{ 7146 const size_t SIZE = 256; 7147 char buffer[SIZE]; 7148 String8 result; 7149 7150 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7151 result.append(buffer); 7152 7153 bool locked = tryLock(mLock); 7154 // failed to lock - AudioFlinger is probably deadlocked 7155 if (!locked) { 7156 result.append("\t\tCould not lock Fx mutex:\n"); 7157 } 7158 7159 result.append("\t\tSession Status State Engine:\n"); 7160 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7161 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7162 result.append(buffer); 7163 7164 result.append("\t\tDescriptor:\n"); 7165 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7166 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7167 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7168 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7169 result.append(buffer); 7170 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7171 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7172 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7173 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7174 result.append(buffer); 7175 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7176 mDescriptor.apiVersion, 7177 mDescriptor.flags); 7178 result.append(buffer); 7179 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7180 mDescriptor.name); 7181 result.append(buffer); 7182 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7183 mDescriptor.implementor); 7184 result.append(buffer); 7185 7186 result.append("\t\t- Input configuration:\n"); 7187 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7188 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7189 (uint32_t)mConfig.inputCfg.buffer.raw, 7190 mConfig.inputCfg.buffer.frameCount, 7191 mConfig.inputCfg.samplingRate, 7192 mConfig.inputCfg.channels, 7193 mConfig.inputCfg.format); 7194 result.append(buffer); 7195 7196 result.append("\t\t- Output configuration:\n"); 7197 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7198 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7199 (uint32_t)mConfig.outputCfg.buffer.raw, 7200 mConfig.outputCfg.buffer.frameCount, 7201 mConfig.outputCfg.samplingRate, 7202 mConfig.outputCfg.channels, 7203 mConfig.outputCfg.format); 7204 result.append(buffer); 7205 7206 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7207 result.append(buffer); 7208 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7209 for (size_t i = 0; i < mHandles.size(); ++i) { 7210 sp<EffectHandle> handle = mHandles[i].promote(); 7211 if (handle != 0) { 7212 handle->dump(buffer, SIZE); 7213 result.append(buffer); 7214 } 7215 } 7216 7217 result.append("\n"); 7218 7219 write(fd, result.string(), result.length()); 7220 7221 if (locked) { 7222 mLock.unlock(); 7223 } 7224 7225 return NO_ERROR; 7226} 7227 7228// ---------------------------------------------------------------------------- 7229// EffectHandle implementation 7230// ---------------------------------------------------------------------------- 7231 7232#undef LOG_TAG 7233#define LOG_TAG "AudioFlinger::EffectHandle" 7234 7235AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7236 const sp<AudioFlinger::Client>& client, 7237 const sp<IEffectClient>& effectClient, 7238 int32_t priority) 7239 : BnEffect(), 7240 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7241 mPriority(priority), mHasControl(false), mEnabled(false) 7242{ 7243 ALOGV("constructor %p", this); 7244 7245 if (client == 0) { 7246 return; 7247 } 7248 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7249 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7250 if (mCblkMemory != 0) { 7251 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7252 7253 if (mCblk != NULL) { 7254 new(mCblk) effect_param_cblk_t(); 7255 mBuffer = (uint8_t *)mCblk + bufOffset; 7256 } 7257 } else { 7258 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7259 return; 7260 } 7261} 7262 7263AudioFlinger::EffectHandle::~EffectHandle() 7264{ 7265 ALOGV("Destructor %p", this); 7266 disconnect(false); 7267 ALOGV("Destructor DONE %p", this); 7268} 7269 7270status_t AudioFlinger::EffectHandle::enable() 7271{ 7272 ALOGV("enable %p", this); 7273 if (!mHasControl) return INVALID_OPERATION; 7274 if (mEffect == 0) return DEAD_OBJECT; 7275 7276 if (mEnabled) { 7277 return NO_ERROR; 7278 } 7279 7280 mEnabled = true; 7281 7282 sp<ThreadBase> thread = mEffect->thread().promote(); 7283 if (thread != 0) { 7284 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7285 } 7286 7287 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7288 if (mEffect->suspended()) { 7289 return NO_ERROR; 7290 } 7291 7292 status_t status = mEffect->setEnabled(true); 7293 if (status != NO_ERROR) { 7294 if (thread != 0) { 7295 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7296 } 7297 mEnabled = false; 7298 } 7299 return status; 7300} 7301 7302status_t AudioFlinger::EffectHandle::disable() 7303{ 7304 ALOGV("disable %p", this); 7305 if (!mHasControl) return INVALID_OPERATION; 7306 if (mEffect == 0) return DEAD_OBJECT; 7307 7308 if (!mEnabled) { 7309 return NO_ERROR; 7310 } 7311 mEnabled = false; 7312 7313 if (mEffect->suspended()) { 7314 return NO_ERROR; 7315 } 7316 7317 status_t status = mEffect->setEnabled(false); 7318 7319 sp<ThreadBase> thread = mEffect->thread().promote(); 7320 if (thread != 0) { 7321 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7322 } 7323 7324 return status; 7325} 7326 7327void AudioFlinger::EffectHandle::disconnect() 7328{ 7329 disconnect(true); 7330} 7331 7332void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7333{ 7334 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7335 if (mEffect == 0) { 7336 return; 7337 } 7338 mEffect->disconnect(this, unpinIfLast); 7339 7340 if (mHasControl && mEnabled) { 7341 sp<ThreadBase> thread = mEffect->thread().promote(); 7342 if (thread != 0) { 7343 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7344 } 7345 } 7346 7347 // release sp on module => module destructor can be called now 7348 mEffect.clear(); 7349 if (mClient != 0) { 7350 if (mCblk != NULL) { 7351 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7352 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7353 } 7354 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7355 // Client destructor must run with AudioFlinger mutex locked 7356 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7357 mClient.clear(); 7358 } 7359} 7360 7361status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7362 uint32_t cmdSize, 7363 void *pCmdData, 7364 uint32_t *replySize, 7365 void *pReplyData) 7366{ 7367// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7368// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7369 7370 // only get parameter command is permitted for applications not controlling the effect 7371 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7372 return INVALID_OPERATION; 7373 } 7374 if (mEffect == 0) return DEAD_OBJECT; 7375 if (mClient == 0) return INVALID_OPERATION; 7376 7377 // handle commands that are not forwarded transparently to effect engine 7378 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7379 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7380 // no risk to block the whole media server process or mixer threads is we are stuck here 7381 Mutex::Autolock _l(mCblk->lock); 7382 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7383 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7384 mCblk->serverIndex = 0; 7385 mCblk->clientIndex = 0; 7386 return BAD_VALUE; 7387 } 7388 status_t status = NO_ERROR; 7389 while (mCblk->serverIndex < mCblk->clientIndex) { 7390 int reply; 7391 uint32_t rsize = sizeof(int); 7392 int *p = (int *)(mBuffer + mCblk->serverIndex); 7393 int size = *p++; 7394 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7395 ALOGW("command(): invalid parameter block size"); 7396 break; 7397 } 7398 effect_param_t *param = (effect_param_t *)p; 7399 if (param->psize == 0 || param->vsize == 0) { 7400 ALOGW("command(): null parameter or value size"); 7401 mCblk->serverIndex += size; 7402 continue; 7403 } 7404 uint32_t psize = sizeof(effect_param_t) + 7405 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7406 param->vsize; 7407 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7408 psize, 7409 p, 7410 &rsize, 7411 &reply); 7412 // stop at first error encountered 7413 if (ret != NO_ERROR) { 7414 status = ret; 7415 *(int *)pReplyData = reply; 7416 break; 7417 } else if (reply != NO_ERROR) { 7418 *(int *)pReplyData = reply; 7419 break; 7420 } 7421 mCblk->serverIndex += size; 7422 } 7423 mCblk->serverIndex = 0; 7424 mCblk->clientIndex = 0; 7425 return status; 7426 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7427 *(int *)pReplyData = NO_ERROR; 7428 return enable(); 7429 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7430 *(int *)pReplyData = NO_ERROR; 7431 return disable(); 7432 } 7433 7434 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7435} 7436 7437void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7438{ 7439 ALOGV("setControl %p control %d", this, hasControl); 7440 7441 mHasControl = hasControl; 7442 mEnabled = enabled; 7443 7444 if (signal && mEffectClient != 0) { 7445 mEffectClient->controlStatusChanged(hasControl); 7446 } 7447} 7448 7449void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7450 uint32_t cmdSize, 7451 void *pCmdData, 7452 uint32_t replySize, 7453 void *pReplyData) 7454{ 7455 if (mEffectClient != 0) { 7456 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7457 } 7458} 7459 7460 7461 7462void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7463{ 7464 if (mEffectClient != 0) { 7465 mEffectClient->enableStatusChanged(enabled); 7466 } 7467} 7468 7469status_t AudioFlinger::EffectHandle::onTransact( 7470 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7471{ 7472 return BnEffect::onTransact(code, data, reply, flags); 7473} 7474 7475 7476void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7477{ 7478 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7479 7480 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7481 (mClient == 0) ? getpid_cached : mClient->pid(), 7482 mPriority, 7483 mHasControl, 7484 !locked, 7485 mCblk ? mCblk->clientIndex : 0, 7486 mCblk ? mCblk->serverIndex : 0 7487 ); 7488 7489 if (locked) { 7490 mCblk->lock.unlock(); 7491 } 7492} 7493 7494#undef LOG_TAG 7495#define LOG_TAG "AudioFlinger::EffectChain" 7496 7497AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7498 int sessionId) 7499 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7500 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7501 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7502{ 7503 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7504 if (thread == NULL) { 7505 return; 7506 } 7507 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7508 thread->frameCount(); 7509} 7510 7511AudioFlinger::EffectChain::~EffectChain() 7512{ 7513 if (mOwnInBuffer) { 7514 delete mInBuffer; 7515 } 7516 7517} 7518 7519// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7520sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7521{ 7522 size_t size = mEffects.size(); 7523 7524 for (size_t i = 0; i < size; i++) { 7525 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7526 return mEffects[i]; 7527 } 7528 } 7529 return 0; 7530} 7531 7532// getEffectFromId_l() must be called with ThreadBase::mLock held 7533sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7534{ 7535 size_t size = mEffects.size(); 7536 7537 for (size_t i = 0; i < size; i++) { 7538 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7539 if (id == 0 || mEffects[i]->id() == id) { 7540 return mEffects[i]; 7541 } 7542 } 7543 return 0; 7544} 7545 7546// getEffectFromType_l() must be called with ThreadBase::mLock held 7547sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7548 const effect_uuid_t *type) 7549{ 7550 size_t size = mEffects.size(); 7551 7552 for (size_t i = 0; i < size; i++) { 7553 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7554 return mEffects[i]; 7555 } 7556 } 7557 return 0; 7558} 7559 7560// Must be called with EffectChain::mLock locked 7561void AudioFlinger::EffectChain::process_l() 7562{ 7563 sp<ThreadBase> thread = mThread.promote(); 7564 if (thread == 0) { 7565 ALOGW("process_l(): cannot promote mixer thread"); 7566 return; 7567 } 7568 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7569 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7570 // always process effects unless no more tracks are on the session and the effect tail 7571 // has been rendered 7572 bool doProcess = true; 7573 if (!isGlobalSession) { 7574 bool tracksOnSession = (trackCnt() != 0); 7575 7576 if (!tracksOnSession && mTailBufferCount == 0) { 7577 doProcess = false; 7578 } 7579 7580 if (activeTrackCnt() == 0) { 7581 // if no track is active and the effect tail has not been rendered, 7582 // the input buffer must be cleared here as the mixer process will not do it 7583 if (tracksOnSession || mTailBufferCount > 0) { 7584 size_t numSamples = thread->frameCount() * thread->channelCount(); 7585 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7586 if (mTailBufferCount > 0) { 7587 mTailBufferCount--; 7588 } 7589 } 7590 } 7591 } 7592 7593 size_t size = mEffects.size(); 7594 if (doProcess) { 7595 for (size_t i = 0; i < size; i++) { 7596 mEffects[i]->process(); 7597 } 7598 } 7599 for (size_t i = 0; i < size; i++) { 7600 mEffects[i]->updateState(); 7601 } 7602} 7603 7604// addEffect_l() must be called with PlaybackThread::mLock held 7605status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7606{ 7607 effect_descriptor_t desc = effect->desc(); 7608 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7609 7610 Mutex::Autolock _l(mLock); 7611 effect->setChain(this); 7612 sp<ThreadBase> thread = mThread.promote(); 7613 if (thread == 0) { 7614 return NO_INIT; 7615 } 7616 effect->setThread(thread); 7617 7618 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7619 // Auxiliary effects are inserted at the beginning of mEffects vector as 7620 // they are processed first and accumulated in chain input buffer 7621 mEffects.insertAt(effect, 0); 7622 7623 // the input buffer for auxiliary effect contains mono samples in 7624 // 32 bit format. This is to avoid saturation in AudoMixer 7625 // accumulation stage. Saturation is done in EffectModule::process() before 7626 // calling the process in effect engine 7627 size_t numSamples = thread->frameCount(); 7628 int32_t *buffer = new int32_t[numSamples]; 7629 memset(buffer, 0, numSamples * sizeof(int32_t)); 7630 effect->setInBuffer((int16_t *)buffer); 7631 // auxiliary effects output samples to chain input buffer for further processing 7632 // by insert effects 7633 effect->setOutBuffer(mInBuffer); 7634 } else { 7635 // Insert effects are inserted at the end of mEffects vector as they are processed 7636 // after track and auxiliary effects. 7637 // Insert effect order as a function of indicated preference: 7638 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7639 // another effect is present 7640 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7641 // last effect claiming first position 7642 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7643 // first effect claiming last position 7644 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7645 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7646 // already present 7647 7648 size_t size = mEffects.size(); 7649 size_t idx_insert = size; 7650 ssize_t idx_insert_first = -1; 7651 ssize_t idx_insert_last = -1; 7652 7653 for (size_t i = 0; i < size; i++) { 7654 effect_descriptor_t d = mEffects[i]->desc(); 7655 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7656 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7657 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7658 // check invalid effect chaining combinations 7659 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7660 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7661 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7662 return INVALID_OPERATION; 7663 } 7664 // remember position of first insert effect and by default 7665 // select this as insert position for new effect 7666 if (idx_insert == size) { 7667 idx_insert = i; 7668 } 7669 // remember position of last insert effect claiming 7670 // first position 7671 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7672 idx_insert_first = i; 7673 } 7674 // remember position of first insert effect claiming 7675 // last position 7676 if (iPref == EFFECT_FLAG_INSERT_LAST && 7677 idx_insert_last == -1) { 7678 idx_insert_last = i; 7679 } 7680 } 7681 } 7682 7683 // modify idx_insert from first position if needed 7684 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7685 if (idx_insert_last != -1) { 7686 idx_insert = idx_insert_last; 7687 } else { 7688 idx_insert = size; 7689 } 7690 } else { 7691 if (idx_insert_first != -1) { 7692 idx_insert = idx_insert_first + 1; 7693 } 7694 } 7695 7696 // always read samples from chain input buffer 7697 effect->setInBuffer(mInBuffer); 7698 7699 // if last effect in the chain, output samples to chain 7700 // output buffer, otherwise to chain input buffer 7701 if (idx_insert == size) { 7702 if (idx_insert != 0) { 7703 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7704 mEffects[idx_insert-1]->configure(); 7705 } 7706 effect->setOutBuffer(mOutBuffer); 7707 } else { 7708 effect->setOutBuffer(mInBuffer); 7709 } 7710 mEffects.insertAt(effect, idx_insert); 7711 7712 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7713 } 7714 effect->configure(); 7715 return NO_ERROR; 7716} 7717 7718// removeEffect_l() must be called with PlaybackThread::mLock held 7719size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7720{ 7721 Mutex::Autolock _l(mLock); 7722 size_t size = mEffects.size(); 7723 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7724 7725 for (size_t i = 0; i < size; i++) { 7726 if (effect == mEffects[i]) { 7727 // calling stop here will remove pre-processing effect from the audio HAL. 7728 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7729 // the middle of a read from audio HAL 7730 if (mEffects[i]->state() == EffectModule::ACTIVE || 7731 mEffects[i]->state() == EffectModule::STOPPING) { 7732 mEffects[i]->stop(); 7733 } 7734 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7735 delete[] effect->inBuffer(); 7736 } else { 7737 if (i == size - 1 && i != 0) { 7738 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7739 mEffects[i - 1]->configure(); 7740 } 7741 } 7742 mEffects.removeAt(i); 7743 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7744 break; 7745 } 7746 } 7747 7748 return mEffects.size(); 7749} 7750 7751// setDevice_l() must be called with PlaybackThread::mLock held 7752void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7753{ 7754 size_t size = mEffects.size(); 7755 for (size_t i = 0; i < size; i++) { 7756 mEffects[i]->setDevice(device); 7757 } 7758} 7759 7760// setMode_l() must be called with PlaybackThread::mLock held 7761void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7762{ 7763 size_t size = mEffects.size(); 7764 for (size_t i = 0; i < size; i++) { 7765 mEffects[i]->setMode(mode); 7766 } 7767} 7768 7769// setVolume_l() must be called with PlaybackThread::mLock held 7770bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7771{ 7772 uint32_t newLeft = *left; 7773 uint32_t newRight = *right; 7774 bool hasControl = false; 7775 int ctrlIdx = -1; 7776 size_t size = mEffects.size(); 7777 7778 // first update volume controller 7779 for (size_t i = size; i > 0; i--) { 7780 if (mEffects[i - 1]->isProcessEnabled() && 7781 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7782 ctrlIdx = i - 1; 7783 hasControl = true; 7784 break; 7785 } 7786 } 7787 7788 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7789 if (hasControl) { 7790 *left = mNewLeftVolume; 7791 *right = mNewRightVolume; 7792 } 7793 return hasControl; 7794 } 7795 7796 mVolumeCtrlIdx = ctrlIdx; 7797 mLeftVolume = newLeft; 7798 mRightVolume = newRight; 7799 7800 // second get volume update from volume controller 7801 if (ctrlIdx >= 0) { 7802 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7803 mNewLeftVolume = newLeft; 7804 mNewRightVolume = newRight; 7805 } 7806 // then indicate volume to all other effects in chain. 7807 // Pass altered volume to effects before volume controller 7808 // and requested volume to effects after controller 7809 uint32_t lVol = newLeft; 7810 uint32_t rVol = newRight; 7811 7812 for (size_t i = 0; i < size; i++) { 7813 if ((int)i == ctrlIdx) continue; 7814 // this also works for ctrlIdx == -1 when there is no volume controller 7815 if ((int)i > ctrlIdx) { 7816 lVol = *left; 7817 rVol = *right; 7818 } 7819 mEffects[i]->setVolume(&lVol, &rVol, false); 7820 } 7821 *left = newLeft; 7822 *right = newRight; 7823 7824 return hasControl; 7825} 7826 7827status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7828{ 7829 const size_t SIZE = 256; 7830 char buffer[SIZE]; 7831 String8 result; 7832 7833 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7834 result.append(buffer); 7835 7836 bool locked = tryLock(mLock); 7837 // failed to lock - AudioFlinger is probably deadlocked 7838 if (!locked) { 7839 result.append("\tCould not lock mutex:\n"); 7840 } 7841 7842 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7843 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7844 mEffects.size(), 7845 (uint32_t)mInBuffer, 7846 (uint32_t)mOutBuffer, 7847 mActiveTrackCnt); 7848 result.append(buffer); 7849 write(fd, result.string(), result.size()); 7850 7851 for (size_t i = 0; i < mEffects.size(); ++i) { 7852 sp<EffectModule> effect = mEffects[i]; 7853 if (effect != 0) { 7854 effect->dump(fd, args); 7855 } 7856 } 7857 7858 if (locked) { 7859 mLock.unlock(); 7860 } 7861 7862 return NO_ERROR; 7863} 7864 7865// must be called with ThreadBase::mLock held 7866void AudioFlinger::EffectChain::setEffectSuspended_l( 7867 const effect_uuid_t *type, bool suspend) 7868{ 7869 sp<SuspendedEffectDesc> desc; 7870 // use effect type UUID timelow as key as there is no real risk of identical 7871 // timeLow fields among effect type UUIDs. 7872 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7873 if (suspend) { 7874 if (index >= 0) { 7875 desc = mSuspendedEffects.valueAt(index); 7876 } else { 7877 desc = new SuspendedEffectDesc(); 7878 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7879 mSuspendedEffects.add(type->timeLow, desc); 7880 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7881 } 7882 if (desc->mRefCount++ == 0) { 7883 sp<EffectModule> effect = getEffectIfEnabled(type); 7884 if (effect != 0) { 7885 desc->mEffect = effect; 7886 effect->setSuspended(true); 7887 effect->setEnabled(false); 7888 } 7889 } 7890 } else { 7891 if (index < 0) { 7892 return; 7893 } 7894 desc = mSuspendedEffects.valueAt(index); 7895 if (desc->mRefCount <= 0) { 7896 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7897 desc->mRefCount = 1; 7898 } 7899 if (--desc->mRefCount == 0) { 7900 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7901 if (desc->mEffect != 0) { 7902 sp<EffectModule> effect = desc->mEffect.promote(); 7903 if (effect != 0) { 7904 effect->setSuspended(false); 7905 sp<EffectHandle> handle = effect->controlHandle(); 7906 if (handle != 0) { 7907 effect->setEnabled(handle->enabled()); 7908 } 7909 } 7910 desc->mEffect.clear(); 7911 } 7912 mSuspendedEffects.removeItemsAt(index); 7913 } 7914 } 7915} 7916 7917// must be called with ThreadBase::mLock held 7918void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7919{ 7920 sp<SuspendedEffectDesc> desc; 7921 7922 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7923 if (suspend) { 7924 if (index >= 0) { 7925 desc = mSuspendedEffects.valueAt(index); 7926 } else { 7927 desc = new SuspendedEffectDesc(); 7928 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7929 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7930 } 7931 if (desc->mRefCount++ == 0) { 7932 Vector< sp<EffectModule> > effects; 7933 getSuspendEligibleEffects(effects); 7934 for (size_t i = 0; i < effects.size(); i++) { 7935 setEffectSuspended_l(&effects[i]->desc().type, true); 7936 } 7937 } 7938 } else { 7939 if (index < 0) { 7940 return; 7941 } 7942 desc = mSuspendedEffects.valueAt(index); 7943 if (desc->mRefCount <= 0) { 7944 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7945 desc->mRefCount = 1; 7946 } 7947 if (--desc->mRefCount == 0) { 7948 Vector<const effect_uuid_t *> types; 7949 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7950 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7951 continue; 7952 } 7953 types.add(&mSuspendedEffects.valueAt(i)->mType); 7954 } 7955 for (size_t i = 0; i < types.size(); i++) { 7956 setEffectSuspended_l(types[i], false); 7957 } 7958 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7959 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7960 } 7961 } 7962} 7963 7964 7965// The volume effect is used for automated tests only 7966#ifndef OPENSL_ES_H_ 7967static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7968 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7969const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7970#endif //OPENSL_ES_H_ 7971 7972bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7973{ 7974 // auxiliary effects and visualizer are never suspended on output mix 7975 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7976 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7977 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7978 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7979 return false; 7980 } 7981 return true; 7982} 7983 7984void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 7985{ 7986 effects.clear(); 7987 for (size_t i = 0; i < mEffects.size(); i++) { 7988 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 7989 effects.add(mEffects[i]); 7990 } 7991 } 7992} 7993 7994sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7995 const effect_uuid_t *type) 7996{ 7997 sp<EffectModule> effect = getEffectFromType_l(type); 7998 return effect != 0 && effect->isEnabled() ? effect : 0; 7999} 8000 8001void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8002 bool enabled) 8003{ 8004 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8005 if (enabled) { 8006 if (index < 0) { 8007 // if the effect is not suspend check if all effects are suspended 8008 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8009 if (index < 0) { 8010 return; 8011 } 8012 if (!isEffectEligibleForSuspend(effect->desc())) { 8013 return; 8014 } 8015 setEffectSuspended_l(&effect->desc().type, enabled); 8016 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8017 if (index < 0) { 8018 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8019 return; 8020 } 8021 } 8022 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8023 effect->desc().type.timeLow); 8024 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8025 // if effect is requested to suspended but was not yet enabled, supend it now. 8026 if (desc->mEffect == 0) { 8027 desc->mEffect = effect; 8028 effect->setEnabled(false); 8029 effect->setSuspended(true); 8030 } 8031 } else { 8032 if (index < 0) { 8033 return; 8034 } 8035 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8036 effect->desc().type.timeLow); 8037 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8038 desc->mEffect.clear(); 8039 effect->setSuspended(false); 8040 } 8041} 8042 8043#undef LOG_TAG 8044#define LOG_TAG "AudioFlinger" 8045 8046// ---------------------------------------------------------------------------- 8047 8048status_t AudioFlinger::onTransact( 8049 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8050{ 8051 return BnAudioFlinger::onTransact(code, data, reply, flags); 8052} 8053 8054}; // namespace android 8055