AudioFlinger.cpp revision 612bbb57c59397a540e96f06bdd16e437a583af5
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 != NULL) { 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 ALOG_ASSERT((NULL != mPrimaryHardwareDev) && 727 (NULL != mPrimaryHardwareDev->get_master_volume), 728 "can't get master volume"); 729 730 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 731 mHardwareStatus = AUDIO_HW_IDLE; 732 return ret_val; 733 } 734 735 return mMasterVolume; 736} 737 738status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 739 audio_io_handle_t output) 740{ 741 // check calling permissions 742 if (!settingsAllowed()) { 743 return PERMISSION_DENIED; 744 } 745 746 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 747 ALOGE("setStreamVolume() invalid stream %d", stream); 748 return BAD_VALUE; 749 } 750 751 AutoMutex lock(mLock); 752 PlaybackThread *thread = NULL; 753 if (output) { 754 thread = checkPlaybackThread_l(output); 755 if (thread == NULL) { 756 return BAD_VALUE; 757 } 758 } 759 760 mStreamTypes[stream].volume = value; 761 762 if (thread == NULL) { 763 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 764 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 765 } 766 } else { 767 thread->setStreamVolume(stream, value); 768 } 769 770 return NO_ERROR; 771} 772 773status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 774{ 775 // check calling permissions 776 if (!settingsAllowed()) { 777 return PERMISSION_DENIED; 778 } 779 780 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 781 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 782 ALOGE("setStreamMute() invalid stream %d", stream); 783 return BAD_VALUE; 784 } 785 786 AutoMutex lock(mLock); 787 mStreamTypes[stream].mute = muted; 788 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 789 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 790 791 return NO_ERROR; 792} 793 794float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 795{ 796 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 797 return 0.0f; 798 } 799 800 AutoMutex lock(mLock); 801 float volume; 802 if (output) { 803 PlaybackThread *thread = checkPlaybackThread_l(output); 804 if (thread == NULL) { 805 return 0.0f; 806 } 807 volume = thread->streamVolume(stream); 808 } else { 809 volume = streamVolume_l(stream); 810 } 811 812 return volume; 813} 814 815bool AudioFlinger::streamMute(audio_stream_type_t stream) const 816{ 817 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 818 return true; 819 } 820 821 AutoMutex lock(mLock); 822 return streamMute_l(stream); 823} 824 825status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 826{ 827 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 828 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 829 // check calling permissions 830 if (!settingsAllowed()) { 831 return PERMISSION_DENIED; 832 } 833 834 // ioHandle == 0 means the parameters are global to the audio hardware interface 835 if (ioHandle == 0) { 836 status_t final_result = NO_ERROR; 837 { 838 AutoMutex lock(mHardwareLock); 839 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 840 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 841 audio_hw_device_t *dev = mAudioHwDevs[i]; 842 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 843 final_result = result ?: final_result; 844 } 845 mHardwareStatus = AUDIO_HW_IDLE; 846 } 847 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 848 AudioParameter param = AudioParameter(keyValuePairs); 849 String8 value; 850 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 851 Mutex::Autolock _l(mLock); 852 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 853 if (mBtNrecIsOff != btNrecIsOff) { 854 for (size_t i = 0; i < mRecordThreads.size(); i++) { 855 sp<RecordThread> thread = mRecordThreads.valueAt(i); 856 RecordThread::RecordTrack *track = thread->track(); 857 if (track != NULL) { 858 audio_devices_t device = (audio_devices_t)( 859 thread->device() & AUDIO_DEVICE_IN_ALL); 860 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 861 thread->setEffectSuspended(FX_IID_AEC, 862 suspend, 863 track->sessionId()); 864 thread->setEffectSuspended(FX_IID_NS, 865 suspend, 866 track->sessionId()); 867 } 868 } 869 mBtNrecIsOff = btNrecIsOff; 870 } 871 } 872 return final_result; 873 } 874 875 // hold a strong ref on thread in case closeOutput() or closeInput() is called 876 // and the thread is exited once the lock is released 877 sp<ThreadBase> thread; 878 { 879 Mutex::Autolock _l(mLock); 880 thread = checkPlaybackThread_l(ioHandle); 881 if (thread == NULL) { 882 thread = checkRecordThread_l(ioHandle); 883 } else if (thread == primaryPlaybackThread_l()) { 884 // indicate output device change to all input threads for pre processing 885 AudioParameter param = AudioParameter(keyValuePairs); 886 int value; 887 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 888 for (size_t i = 0; i < mRecordThreads.size(); i++) { 889 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 890 } 891 } 892 } 893 } 894 if (thread != 0) { 895 return thread->setParameters(keyValuePairs); 896 } 897 return BAD_VALUE; 898} 899 900String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 901{ 902// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 903// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 904 905 if (ioHandle == 0) { 906 String8 out_s8; 907 908 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 909 char *s; 910 { 911 AutoMutex lock(mHardwareLock); 912 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 913 audio_hw_device_t *dev = mAudioHwDevs[i]; 914 s = dev->get_parameters(dev, keys.string()); 915 mHardwareStatus = AUDIO_HW_IDLE; 916 } 917 out_s8 += String8(s ? s : ""); 918 free(s); 919 } 920 return out_s8; 921 } 922 923 Mutex::Autolock _l(mLock); 924 925 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 926 if (playbackThread != NULL) { 927 return playbackThread->getParameters(keys); 928 } 929 RecordThread *recordThread = checkRecordThread_l(ioHandle); 930 if (recordThread != NULL) { 931 return recordThread->getParameters(keys); 932 } 933 return String8(""); 934} 935 936size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 937{ 938 status_t ret = initCheck(); 939 if (ret != NO_ERROR) { 940 return 0; 941 } 942 943 AutoMutex lock(mHardwareLock); 944 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 945 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 946 mHardwareStatus = AUDIO_HW_IDLE; 947 return size; 948} 949 950unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 951{ 952 if (ioHandle == 0) { 953 return 0; 954 } 955 956 Mutex::Autolock _l(mLock); 957 958 RecordThread *recordThread = checkRecordThread_l(ioHandle); 959 if (recordThread != NULL) { 960 return recordThread->getInputFramesLost(); 961 } 962 return 0; 963} 964 965status_t AudioFlinger::setVoiceVolume(float value) 966{ 967 status_t ret = initCheck(); 968 if (ret != NO_ERROR) { 969 return ret; 970 } 971 972 // check calling permissions 973 if (!settingsAllowed()) { 974 return PERMISSION_DENIED; 975 } 976 977 AutoMutex lock(mHardwareLock); 978 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 979 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 980 mHardwareStatus = AUDIO_HW_IDLE; 981 982 return ret; 983} 984 985status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 986 audio_io_handle_t output) const 987{ 988 status_t status; 989 990 Mutex::Autolock _l(mLock); 991 992 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 993 if (playbackThread != NULL) { 994 return playbackThread->getRenderPosition(halFrames, dspFrames); 995 } 996 997 return BAD_VALUE; 998} 999 1000void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1001{ 1002 1003 Mutex::Autolock _l(mLock); 1004 1005 pid_t pid = IPCThreadState::self()->getCallingPid(); 1006 if (mNotificationClients.indexOfKey(pid) < 0) { 1007 sp<NotificationClient> notificationClient = new NotificationClient(this, 1008 client, 1009 pid); 1010 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1011 1012 mNotificationClients.add(pid, notificationClient); 1013 1014 sp<IBinder> binder = client->asBinder(); 1015 binder->linkToDeath(notificationClient); 1016 1017 // the config change is always sent from playback or record threads to avoid deadlock 1018 // with AudioSystem::gLock 1019 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1020 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1021 } 1022 1023 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1024 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1025 } 1026 } 1027} 1028 1029void AudioFlinger::removeNotificationClient(pid_t pid) 1030{ 1031 Mutex::Autolock _l(mLock); 1032 1033 mNotificationClients.removeItem(pid); 1034 1035 ALOGV("%d died, releasing its sessions", pid); 1036 size_t num = mAudioSessionRefs.size(); 1037 bool removed = false; 1038 for (size_t i = 0; i< num; ) { 1039 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1040 ALOGV(" pid %d @ %d", ref->mPid, i); 1041 if (ref->mPid == pid) { 1042 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1043 mAudioSessionRefs.removeAt(i); 1044 delete ref; 1045 removed = true; 1046 num--; 1047 } else { 1048 i++; 1049 } 1050 } 1051 if (removed) { 1052 purgeStaleEffects_l(); 1053 } 1054} 1055 1056// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1057void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1058{ 1059 size_t size = mNotificationClients.size(); 1060 for (size_t i = 0; i < size; i++) { 1061 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1062 param2); 1063 } 1064} 1065 1066// removeClient_l() must be called with AudioFlinger::mLock held 1067void AudioFlinger::removeClient_l(pid_t pid) 1068{ 1069 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1070 mClients.removeItem(pid); 1071} 1072 1073 1074// ---------------------------------------------------------------------------- 1075 1076AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1077 uint32_t device, type_t type) 1078 : Thread(false), 1079 mType(type), 1080 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1081 // mChannelMask 1082 mChannelCount(0), 1083 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1084 mParamStatus(NO_ERROR), 1085 mStandby(false), mId(id), 1086 mDevice(device), 1087 mDeathRecipient(new PMDeathRecipient(this)) 1088{ 1089} 1090 1091AudioFlinger::ThreadBase::~ThreadBase() 1092{ 1093 mParamCond.broadcast(); 1094 // do not lock the mutex in destructor 1095 releaseWakeLock_l(); 1096 if (mPowerManager != 0) { 1097 sp<IBinder> binder = mPowerManager->asBinder(); 1098 binder->unlinkToDeath(mDeathRecipient); 1099 } 1100} 1101 1102void AudioFlinger::ThreadBase::exit() 1103{ 1104 ALOGV("ThreadBase::exit"); 1105 { 1106 // This lock prevents the following race in thread (uniprocessor for illustration): 1107 // if (!exitPending()) { 1108 // // context switch from here to exit() 1109 // // exit() calls requestExit(), what exitPending() observes 1110 // // exit() calls signal(), which is dropped since no waiters 1111 // // context switch back from exit() to here 1112 // mWaitWorkCV.wait(...); 1113 // // now thread is hung 1114 // } 1115 AutoMutex lock(mLock); 1116 requestExit(); 1117 mWaitWorkCV.signal(); 1118 } 1119 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1120 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1121 requestExitAndWait(); 1122} 1123 1124status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1125{ 1126 status_t status; 1127 1128 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1129 Mutex::Autolock _l(mLock); 1130 1131 mNewParameters.add(keyValuePairs); 1132 mWaitWorkCV.signal(); 1133 // wait condition with timeout in case the thread loop has exited 1134 // before the request could be processed 1135 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1136 status = mParamStatus; 1137 mWaitWorkCV.signal(); 1138 } else { 1139 status = TIMED_OUT; 1140 } 1141 return status; 1142} 1143 1144void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1145{ 1146 Mutex::Autolock _l(mLock); 1147 sendConfigEvent_l(event, param); 1148} 1149 1150// sendConfigEvent_l() must be called with ThreadBase::mLock held 1151void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1152{ 1153 ConfigEvent configEvent; 1154 configEvent.mEvent = event; 1155 configEvent.mParam = param; 1156 mConfigEvents.add(configEvent); 1157 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1158 mWaitWorkCV.signal(); 1159} 1160 1161void AudioFlinger::ThreadBase::processConfigEvents() 1162{ 1163 mLock.lock(); 1164 while (!mConfigEvents.isEmpty()) { 1165 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1166 ConfigEvent configEvent = mConfigEvents[0]; 1167 mConfigEvents.removeAt(0); 1168 // release mLock before locking AudioFlinger mLock: lock order is always 1169 // AudioFlinger then ThreadBase to avoid cross deadlock 1170 mLock.unlock(); 1171 mAudioFlinger->mLock.lock(); 1172 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1173 mAudioFlinger->mLock.unlock(); 1174 mLock.lock(); 1175 } 1176 mLock.unlock(); 1177} 1178 1179status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1180{ 1181 const size_t SIZE = 256; 1182 char buffer[SIZE]; 1183 String8 result; 1184 1185 bool locked = tryLock(mLock); 1186 if (!locked) { 1187 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1188 write(fd, buffer, strlen(buffer)); 1189 } 1190 1191 snprintf(buffer, SIZE, "io handle: %d\n", mId); 1192 result.append(buffer); 1193 snprintf(buffer, SIZE, "TID: %d\n", getTid()); 1194 result.append(buffer); 1195 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1196 result.append(buffer); 1197 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1198 result.append(buffer); 1199 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1200 result.append(buffer); 1201 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1202 result.append(buffer); 1203 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1204 result.append(buffer); 1205 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1206 result.append(buffer); 1207 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1208 result.append(buffer); 1209 1210 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1211 result.append(buffer); 1212 result.append(" Index Command"); 1213 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1214 snprintf(buffer, SIZE, "\n %02d ", i); 1215 result.append(buffer); 1216 result.append(mNewParameters[i]); 1217 } 1218 1219 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1220 result.append(buffer); 1221 snprintf(buffer, SIZE, " Index event param\n"); 1222 result.append(buffer); 1223 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1224 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1225 result.append(buffer); 1226 } 1227 result.append("\n"); 1228 1229 write(fd, result.string(), result.size()); 1230 1231 if (locked) { 1232 mLock.unlock(); 1233 } 1234 return NO_ERROR; 1235} 1236 1237status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1238{ 1239 const size_t SIZE = 256; 1240 char buffer[SIZE]; 1241 String8 result; 1242 1243 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1244 write(fd, buffer, strlen(buffer)); 1245 1246 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1247 sp<EffectChain> chain = mEffectChains[i]; 1248 if (chain != 0) { 1249 chain->dump(fd, args); 1250 } 1251 } 1252 return NO_ERROR; 1253} 1254 1255void AudioFlinger::ThreadBase::acquireWakeLock() 1256{ 1257 Mutex::Autolock _l(mLock); 1258 acquireWakeLock_l(); 1259} 1260 1261void AudioFlinger::ThreadBase::acquireWakeLock_l() 1262{ 1263 if (mPowerManager == 0) { 1264 // use checkService() to avoid blocking if power service is not up yet 1265 sp<IBinder> binder = 1266 defaultServiceManager()->checkService(String16("power")); 1267 if (binder == 0) { 1268 ALOGW("Thread %s cannot connect to the power manager service", mName); 1269 } else { 1270 mPowerManager = interface_cast<IPowerManager>(binder); 1271 binder->linkToDeath(mDeathRecipient); 1272 } 1273 } 1274 if (mPowerManager != 0) { 1275 sp<IBinder> binder = new BBinder(); 1276 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1277 binder, 1278 String16(mName)); 1279 if (status == NO_ERROR) { 1280 mWakeLockToken = binder; 1281 } 1282 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1283 } 1284} 1285 1286void AudioFlinger::ThreadBase::releaseWakeLock() 1287{ 1288 Mutex::Autolock _l(mLock); 1289 releaseWakeLock_l(); 1290} 1291 1292void AudioFlinger::ThreadBase::releaseWakeLock_l() 1293{ 1294 if (mWakeLockToken != 0) { 1295 ALOGV("releaseWakeLock_l() %s", mName); 1296 if (mPowerManager != 0) { 1297 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1298 } 1299 mWakeLockToken.clear(); 1300 } 1301} 1302 1303void AudioFlinger::ThreadBase::clearPowerManager() 1304{ 1305 Mutex::Autolock _l(mLock); 1306 releaseWakeLock_l(); 1307 mPowerManager.clear(); 1308} 1309 1310void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1311{ 1312 sp<ThreadBase> thread = mThread.promote(); 1313 if (thread != 0) { 1314 thread->clearPowerManager(); 1315 } 1316 ALOGW("power manager service died !!!"); 1317} 1318 1319void AudioFlinger::ThreadBase::setEffectSuspended( 1320 const effect_uuid_t *type, bool suspend, int sessionId) 1321{ 1322 Mutex::Autolock _l(mLock); 1323 setEffectSuspended_l(type, suspend, sessionId); 1324} 1325 1326void AudioFlinger::ThreadBase::setEffectSuspended_l( 1327 const effect_uuid_t *type, bool suspend, int sessionId) 1328{ 1329 sp<EffectChain> chain = getEffectChain_l(sessionId); 1330 if (chain != 0) { 1331 if (type != NULL) { 1332 chain->setEffectSuspended_l(type, suspend); 1333 } else { 1334 chain->setEffectSuspendedAll_l(suspend); 1335 } 1336 } 1337 1338 updateSuspendedSessions_l(type, suspend, sessionId); 1339} 1340 1341void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1342{ 1343 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1344 if (index < 0) { 1345 return; 1346 } 1347 1348 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1349 mSuspendedSessions.editValueAt(index); 1350 1351 for (size_t i = 0; i < sessionEffects.size(); i++) { 1352 sp<SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1353 for (int j = 0; j < desc->mRefCount; j++) { 1354 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1355 chain->setEffectSuspendedAll_l(true); 1356 } else { 1357 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1358 desc->mType.timeLow); 1359 chain->setEffectSuspended_l(&desc->mType, true); 1360 } 1361 } 1362 } 1363} 1364 1365void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1366 bool suspend, 1367 int sessionId) 1368{ 1369 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1370 1371 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1372 1373 if (suspend) { 1374 if (index >= 0) { 1375 sessionEffects = mSuspendedSessions.editValueAt(index); 1376 } else { 1377 mSuspendedSessions.add(sessionId, sessionEffects); 1378 } 1379 } else { 1380 if (index < 0) { 1381 return; 1382 } 1383 sessionEffects = mSuspendedSessions.editValueAt(index); 1384 } 1385 1386 1387 int key = EffectChain::kKeyForSuspendAll; 1388 if (type != NULL) { 1389 key = type->timeLow; 1390 } 1391 index = sessionEffects.indexOfKey(key); 1392 1393 sp<SuspendedSessionDesc> desc; 1394 if (suspend) { 1395 if (index >= 0) { 1396 desc = sessionEffects.valueAt(index); 1397 } else { 1398 desc = new SuspendedSessionDesc(); 1399 if (type != NULL) { 1400 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1401 } 1402 sessionEffects.add(key, desc); 1403 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1404 } 1405 desc->mRefCount++; 1406 } else { 1407 if (index < 0) { 1408 return; 1409 } 1410 desc = sessionEffects.valueAt(index); 1411 if (--desc->mRefCount == 0) { 1412 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1413 sessionEffects.removeItemsAt(index); 1414 if (sessionEffects.isEmpty()) { 1415 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1416 sessionId); 1417 mSuspendedSessions.removeItem(sessionId); 1418 } 1419 } 1420 } 1421 if (!sessionEffects.isEmpty()) { 1422 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1423 } 1424} 1425 1426void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1427 bool enabled, 1428 int sessionId) 1429{ 1430 Mutex::Autolock _l(mLock); 1431 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1432} 1433 1434void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1435 bool enabled, 1436 int sessionId) 1437{ 1438 if (mType != RECORD) { 1439 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1440 // another session. This gives the priority to well behaved effect control panels 1441 // and applications not using global effects. 1442 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1443 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1444 } 1445 } 1446 1447 sp<EffectChain> chain = getEffectChain_l(sessionId); 1448 if (chain != 0) { 1449 chain->checkSuspendOnEffectEnabled(effect, enabled); 1450 } 1451} 1452 1453// ---------------------------------------------------------------------------- 1454 1455AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1456 AudioStreamOut* output, 1457 audio_io_handle_t id, 1458 uint32_t device, 1459 type_t type) 1460 : ThreadBase(audioFlinger, id, device, type), 1461 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1462 // Assumes constructor is called by AudioFlinger with it's mLock held, 1463 // but it would be safer to explicitly pass initial masterMute as parameter 1464 mMasterMute(audioFlinger->masterMute_l()), 1465 // mStreamTypes[] initialized in constructor body 1466 mOutput(output), 1467 // Assumes constructor is called by AudioFlinger with it's mLock held, 1468 // but it would be safer to explicitly pass initial masterVolume as parameter 1469 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1470 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false), 1471 mMixerStatus(MIXER_IDLE), 1472 mPrevMixerStatus(MIXER_IDLE), 1473 standbyDelay(AudioFlinger::mStandbyTimeInNsecs) 1474{ 1475 snprintf(mName, kNameLength, "AudioOut_%X", id); 1476 1477 readOutputParameters(); 1478 1479 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1480 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1481 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1482 stream = (audio_stream_type_t) (stream + 1)) { 1483 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1484 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1485 // initialized by stream_type_t default constructor 1486 // mStreamTypes[stream].valid = true; 1487 } 1488 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1489 // because mAudioFlinger doesn't have one to copy from 1490} 1491 1492AudioFlinger::PlaybackThread::~PlaybackThread() 1493{ 1494 delete [] mMixBuffer; 1495} 1496 1497status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1498{ 1499 dumpInternals(fd, args); 1500 dumpTracks(fd, args); 1501 dumpEffectChains(fd, args); 1502 return NO_ERROR; 1503} 1504 1505status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1506{ 1507 const size_t SIZE = 256; 1508 char buffer[SIZE]; 1509 String8 result; 1510 1511 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1512 result.append(buffer); 1513 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1514 for (size_t i = 0; i < mTracks.size(); ++i) { 1515 sp<Track> track = mTracks[i]; 1516 if (track != 0) { 1517 track->dump(buffer, SIZE); 1518 result.append(buffer); 1519 } 1520 } 1521 1522 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1523 result.append(buffer); 1524 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1525 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1526 sp<Track> track = mActiveTracks[i].promote(); 1527 if (track != 0) { 1528 track->dump(buffer, SIZE); 1529 result.append(buffer); 1530 } 1531 } 1532 write(fd, result.string(), result.size()); 1533 return NO_ERROR; 1534} 1535 1536status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1537{ 1538 const size_t SIZE = 256; 1539 char buffer[SIZE]; 1540 String8 result; 1541 1542 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1545 result.append(buffer); 1546 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1547 result.append(buffer); 1548 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1549 result.append(buffer); 1550 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1551 result.append(buffer); 1552 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1553 result.append(buffer); 1554 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1555 result.append(buffer); 1556 write(fd, result.string(), result.size()); 1557 1558 dumpBase(fd, args); 1559 1560 return NO_ERROR; 1561} 1562 1563// Thread virtuals 1564status_t AudioFlinger::PlaybackThread::readyToRun() 1565{ 1566 status_t status = initCheck(); 1567 if (status == NO_ERROR) { 1568 ALOGI("AudioFlinger's thread %p ready to run", this); 1569 } else { 1570 ALOGE("No working audio driver found."); 1571 } 1572 return status; 1573} 1574 1575void AudioFlinger::PlaybackThread::onFirstRef() 1576{ 1577 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1578} 1579 1580// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1581sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1582 const sp<AudioFlinger::Client>& client, 1583 audio_stream_type_t streamType, 1584 uint32_t sampleRate, 1585 audio_format_t format, 1586 uint32_t channelMask, 1587 int frameCount, 1588 const sp<IMemory>& sharedBuffer, 1589 int sessionId, 1590 bool isTimed, 1591 status_t *status) 1592{ 1593 sp<Track> track; 1594 status_t lStatus; 1595 1596 if (mType == DIRECT) { 1597 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1598 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1599 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1600 "for output %p with format %d", 1601 sampleRate, format, channelMask, mOutput, mFormat); 1602 lStatus = BAD_VALUE; 1603 goto Exit; 1604 } 1605 } 1606 } else { 1607 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1608 if (sampleRate > mSampleRate*2) { 1609 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1610 lStatus = BAD_VALUE; 1611 goto Exit; 1612 } 1613 } 1614 1615 lStatus = initCheck(); 1616 if (lStatus != NO_ERROR) { 1617 ALOGE("Audio driver not initialized."); 1618 goto Exit; 1619 } 1620 1621 { // scope for mLock 1622 Mutex::Autolock _l(mLock); 1623 1624 // all tracks in same audio session must share the same routing strategy otherwise 1625 // conflicts will happen when tracks are moved from one output to another by audio policy 1626 // manager 1627 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1628 for (size_t i = 0; i < mTracks.size(); ++i) { 1629 sp<Track> t = mTracks[i]; 1630 if (t != 0 && !t->isOutputTrack()) { 1631 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1632 if (sessionId == t->sessionId() && strategy != actual) { 1633 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1634 strategy, actual); 1635 lStatus = BAD_VALUE; 1636 goto Exit; 1637 } 1638 } 1639 } 1640 1641 if (!isTimed) { 1642 track = new Track(this, client, streamType, sampleRate, format, 1643 channelMask, frameCount, sharedBuffer, sessionId); 1644 } else { 1645 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1646 channelMask, frameCount, sharedBuffer, sessionId); 1647 } 1648 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1649 lStatus = NO_MEMORY; 1650 goto Exit; 1651 } 1652 mTracks.add(track); 1653 1654 sp<EffectChain> chain = getEffectChain_l(sessionId); 1655 if (chain != 0) { 1656 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1657 track->setMainBuffer(chain->inBuffer()); 1658 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1659 chain->incTrackCnt(); 1660 } 1661 1662 // invalidate track immediately if the stream type was moved to another thread since 1663 // createTrack() was called by the client process. 1664 if (!mStreamTypes[streamType].valid) { 1665 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1666 this, streamType); 1667 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1668 } 1669 } 1670 lStatus = NO_ERROR; 1671 1672Exit: 1673 if (status) { 1674 *status = lStatus; 1675 } 1676 return track; 1677} 1678 1679uint32_t AudioFlinger::PlaybackThread::latency() const 1680{ 1681 Mutex::Autolock _l(mLock); 1682 if (initCheck() == NO_ERROR) { 1683 return mOutput->stream->get_latency(mOutput->stream); 1684 } else { 1685 return 0; 1686 } 1687} 1688 1689void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1690{ 1691 Mutex::Autolock _l(mLock); 1692 mMasterVolume = value; 1693} 1694 1695void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1696{ 1697 Mutex::Autolock _l(mLock); 1698 setMasterMute_l(muted); 1699} 1700 1701void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1702{ 1703 Mutex::Autolock _l(mLock); 1704 mStreamTypes[stream].volume = value; 1705} 1706 1707void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1708{ 1709 Mutex::Autolock _l(mLock); 1710 mStreamTypes[stream].mute = muted; 1711} 1712 1713float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1714{ 1715 Mutex::Autolock _l(mLock); 1716 return mStreamTypes[stream].volume; 1717} 1718 1719// addTrack_l() must be called with ThreadBase::mLock held 1720status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1721{ 1722 status_t status = ALREADY_EXISTS; 1723 1724 // set retry count for buffer fill 1725 track->mRetryCount = kMaxTrackStartupRetries; 1726 if (mActiveTracks.indexOf(track) < 0) { 1727 // the track is newly added, make sure it fills up all its 1728 // buffers before playing. This is to ensure the client will 1729 // effectively get the latency it requested. 1730 track->mFillingUpStatus = Track::FS_FILLING; 1731 track->mResetDone = false; 1732 mActiveTracks.add(track); 1733 if (track->mainBuffer() != mMixBuffer) { 1734 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1735 if (chain != 0) { 1736 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1737 chain->incActiveTrackCnt(); 1738 } 1739 } 1740 1741 status = NO_ERROR; 1742 } 1743 1744 ALOGV("mWaitWorkCV.broadcast"); 1745 mWaitWorkCV.broadcast(); 1746 1747 return status; 1748} 1749 1750// destroyTrack_l() must be called with ThreadBase::mLock held 1751void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1752{ 1753 track->mState = TrackBase::TERMINATED; 1754 if (mActiveTracks.indexOf(track) < 0) { 1755 removeTrack_l(track); 1756 } 1757} 1758 1759void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1760{ 1761 mTracks.remove(track); 1762 deleteTrackName_l(track->name()); 1763 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1764 if (chain != 0) { 1765 chain->decTrackCnt(); 1766 } 1767} 1768 1769String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1770{ 1771 String8 out_s8 = String8(""); 1772 char *s; 1773 1774 Mutex::Autolock _l(mLock); 1775 if (initCheck() != NO_ERROR) { 1776 return out_s8; 1777 } 1778 1779 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1780 out_s8 = String8(s); 1781 free(s); 1782 return out_s8; 1783} 1784 1785// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1786void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1787 AudioSystem::OutputDescriptor desc; 1788 void *param2 = NULL; 1789 1790 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1791 1792 switch (event) { 1793 case AudioSystem::OUTPUT_OPENED: 1794 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1795 desc.channels = mChannelMask; 1796 desc.samplingRate = mSampleRate; 1797 desc.format = mFormat; 1798 desc.frameCount = mFrameCount; 1799 desc.latency = latency(); 1800 param2 = &desc; 1801 break; 1802 1803 case AudioSystem::STREAM_CONFIG_CHANGED: 1804 param2 = ¶m; 1805 case AudioSystem::OUTPUT_CLOSED: 1806 default: 1807 break; 1808 } 1809 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1810} 1811 1812void AudioFlinger::PlaybackThread::readOutputParameters() 1813{ 1814 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1815 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1816 mChannelCount = (uint16_t)popcount(mChannelMask); 1817 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1818 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1819 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1820 1821 // FIXME - Current mixer implementation only supports stereo output: Always 1822 // Allocate a stereo buffer even if HW output is mono. 1823 delete[] mMixBuffer; 1824 mMixBuffer = new int16_t[mFrameCount * 2]; 1825 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1826 1827 // force reconfiguration of effect chains and engines to take new buffer size and audio 1828 // parameters into account 1829 // Note that mLock is not held when readOutputParameters() is called from the constructor 1830 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1831 // matter. 1832 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1833 Vector< sp<EffectChain> > effectChains = mEffectChains; 1834 for (size_t i = 0; i < effectChains.size(); i ++) { 1835 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1836 } 1837} 1838 1839status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1840{ 1841 if (halFrames == NULL || dspFrames == NULL) { 1842 return BAD_VALUE; 1843 } 1844 Mutex::Autolock _l(mLock); 1845 if (initCheck() != NO_ERROR) { 1846 return INVALID_OPERATION; 1847 } 1848 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1849 1850 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1851} 1852 1853uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1854{ 1855 Mutex::Autolock _l(mLock); 1856 uint32_t result = 0; 1857 if (getEffectChain_l(sessionId) != 0) { 1858 result = EFFECT_SESSION; 1859 } 1860 1861 for (size_t i = 0; i < mTracks.size(); ++i) { 1862 sp<Track> track = mTracks[i]; 1863 if (sessionId == track->sessionId() && 1864 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1865 result |= TRACK_SESSION; 1866 break; 1867 } 1868 } 1869 1870 return result; 1871} 1872 1873uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1874{ 1875 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1876 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1877 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1878 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1879 } 1880 for (size_t i = 0; i < mTracks.size(); i++) { 1881 sp<Track> track = mTracks[i]; 1882 if (sessionId == track->sessionId() && 1883 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1884 return AudioSystem::getStrategyForStream(track->streamType()); 1885 } 1886 } 1887 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1888} 1889 1890 1891AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1892{ 1893 Mutex::Autolock _l(mLock); 1894 return mOutput; 1895} 1896 1897AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1898{ 1899 Mutex::Autolock _l(mLock); 1900 AudioStreamOut *output = mOutput; 1901 mOutput = NULL; 1902 return output; 1903} 1904 1905// this method must always be called either with ThreadBase mLock held or inside the thread loop 1906audio_stream_t* AudioFlinger::PlaybackThread::stream() 1907{ 1908 if (mOutput == NULL) { 1909 return NULL; 1910 } 1911 return &mOutput->stream->common; 1912} 1913 1914uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1915{ 1916 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1917 // decoding and transfer time. So sleeping for half of the latency would likely cause 1918 // underruns 1919 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1920 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1921 } else { 1922 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1923 } 1924} 1925 1926// ---------------------------------------------------------------------------- 1927 1928AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1929 audio_io_handle_t id, uint32_t device, type_t type) 1930 : PlaybackThread(audioFlinger, output, id, device, type) 1931{ 1932 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1933 // FIXME - Current mixer implementation only supports stereo output 1934 if (mChannelCount == 1) { 1935 ALOGE("Invalid audio hardware channel count"); 1936 } 1937} 1938 1939AudioFlinger::MixerThread::~MixerThread() 1940{ 1941 delete mAudioMixer; 1942} 1943 1944class CpuStats { 1945public: 1946 void sample(); 1947#ifdef DEBUG_CPU_USAGE 1948private: 1949 ThreadCpuUsage mCpu; 1950#endif 1951}; 1952 1953void CpuStats::sample() { 1954#ifdef DEBUG_CPU_USAGE 1955 const CentralTendencyStatistics& stats = mCpu.statistics(); 1956 mCpu.sampleAndEnable(); 1957 unsigned n = stats.n(); 1958 // mCpu.elapsed() is expensive, so don't call it every loop 1959 if ((n & 127) == 1) { 1960 long long elapsed = mCpu.elapsed(); 1961 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1962 double perLoop = elapsed / (double) n; 1963 double perLoop100 = perLoop * 0.01; 1964 double mean = stats.mean(); 1965 double stddev = stats.stddev(); 1966 double minimum = stats.minimum(); 1967 double maximum = stats.maximum(); 1968 mCpu.resetStatistics(); 1969 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", 1970 elapsed * .000000001, n, perLoop * .000001, 1971 mean * .001, 1972 stddev * .001, 1973 minimum * .001, 1974 maximum * .001, 1975 mean / perLoop100, 1976 stddev / perLoop100, 1977 minimum / perLoop100, 1978 maximum / perLoop100); 1979 } 1980 } 1981#endif 1982}; 1983 1984void AudioFlinger::PlaybackThread::checkSilentMode_l() 1985{ 1986 if (!mMasterMute) { 1987 char value[PROPERTY_VALUE_MAX]; 1988 if (property_get("ro.audio.silent", value, "0") > 0) { 1989 char *endptr; 1990 unsigned long ul = strtoul(value, &endptr, 0); 1991 if (*endptr == '\0' && ul != 0) { 1992 ALOGD("Silence is golden"); 1993 // The setprop command will not allow a property to be changed after 1994 // the first time it is set, so we don't have to worry about un-muting. 1995 setMasterMute_l(true); 1996 } 1997 } 1998 } 1999} 2000 2001bool AudioFlinger::PlaybackThread::threadLoop() 2002{ 2003 Vector< sp<Track> > tracksToRemove; 2004 2005 standbyTime = systemTime(); 2006 2007 // MIXER 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 cacheParameters_l(); 2018 sleepTime = idleSleepTime; 2019 2020if (mType == MIXER) { 2021 sleepTimeShift = 0; 2022} 2023 2024 // MIXER 2025 CpuStats cpuStats; 2026 2027 acquireWakeLock(); 2028 2029 while (!exitPending()) 2030 { 2031if (mType == MIXER) { 2032 cpuStats.sample(); 2033} 2034 2035 Vector< sp<EffectChain> > effectChains; 2036 2037 processConfigEvents(); 2038 2039 { // scope for mLock 2040 2041 Mutex::Autolock _l(mLock); 2042 2043 if (checkForNewParameters_l()) { 2044 cacheParameters_l(); 2045 } 2046 2047 saveOutputTracks(); 2048 2049 // put audio hardware into standby after short delay 2050 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2051 mSuspended > 0)) { 2052 if (!mStandby) { 2053 2054 threadLoop_standby(); 2055 2056 mStandby = true; 2057 mBytesWritten = 0; 2058 } 2059 2060 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2061 // we're about to wait, flush the binder command buffer 2062 IPCThreadState::self()->flushCommands(); 2063 2064 clearOutputTracks(); 2065 2066 if (exitPending()) break; 2067 2068 releaseWakeLock_l(); 2069 // wait until we have something to do... 2070 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2071 mWaitWorkCV.wait(mLock); 2072 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2073 acquireWakeLock_l(); 2074 2075 mPrevMixerStatus = MIXER_IDLE; 2076 2077 checkSilentMode_l(); 2078 2079 standbyTime = systemTime() + standbyDelay; 2080 sleepTime = idleSleepTime; 2081 if (mType == MIXER) { 2082 sleepTimeShift = 0; 2083 } 2084 2085 continue; 2086 } 2087 } 2088 2089 mixer_state newMixerStatus = prepareTracks_l(&tracksToRemove); 2090 // Shift in the new status; this could be a queue if it's 2091 // useful to filter the mixer status over several cycles. 2092 mPrevMixerStatus = mMixerStatus; 2093 mMixerStatus = newMixerStatus; 2094 2095 // prevent any changes in effect chain list and in each effect chain 2096 // during mixing and effect process as the audio buffers could be deleted 2097 // or modified if an effect is created or deleted 2098 lockEffectChains_l(effectChains); 2099 } 2100 2101 if (CC_LIKELY(mMixerStatus == MIXER_TRACKS_READY)) { 2102 threadLoop_mix(); 2103 } else { 2104 threadLoop_sleepTime(); 2105 } 2106 2107 if (mSuspended > 0) { 2108 sleepTime = suspendSleepTimeUs(); 2109 } 2110 2111 // only process effects if we're going to write 2112 if (sleepTime == 0) { 2113 for (size_t i = 0; i < effectChains.size(); i ++) { 2114 effectChains[i]->process_l(); 2115 } 2116 } 2117 2118 // enable changes in effect chain 2119 unlockEffectChains(effectChains); 2120 2121 // sleepTime == 0 means we must write to audio hardware 2122 if (sleepTime == 0) { 2123 2124 threadLoop_write(); 2125 2126if (mType == MIXER) { 2127 // write blocked detection 2128 nsecs_t now = systemTime(); 2129 nsecs_t delta = now - mLastWriteTime; 2130 if (!mStandby && delta > maxPeriod) { 2131 mNumDelayedWrites++; 2132 if ((now - lastWarning) > kWarningThrottleNs) { 2133 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2134 ns2ms(delta), mNumDelayedWrites, this); 2135 lastWarning = now; 2136 } 2137 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2138 // a different threshold. Or completely removed for what it is worth anyway... 2139 if (mStandby) { 2140 longStandbyExit = true; 2141 } 2142 } 2143} 2144 2145 mStandby = false; 2146 } else { 2147 usleep(sleepTime); 2148 } 2149 2150 // finally let go of removed track(s), without the lock held 2151 // since we can't guarantee the destructors won't acquire that 2152 // same lock. 2153 tracksToRemove.clear(); 2154 2155 // FIXME I don't understand the need for this here; 2156 // it was in the original code but maybe the 2157 // assignment in saveOutputTracks() makes this unnecessary? 2158 clearOutputTracks(); 2159 2160 // Effect chains will be actually deleted here if they were removed from 2161 // mEffectChains list during mixing or effects processing 2162 effectChains.clear(); 2163 2164 // FIXME Note that the above .clear() is no longer necessary since effectChains 2165 // is now local to this block, but will keep it for now (at least until merge done). 2166 } 2167 2168if (mType == MIXER || mType == DIRECT) { 2169 // put output stream into standby mode 2170 if (!mStandby) { 2171 mOutput->stream->common.standby(&mOutput->stream->common); 2172 } 2173} 2174if (mType == DUPLICATING) { 2175 // for DuplicatingThread, standby mode is handled by the outputTracks 2176} 2177 2178 releaseWakeLock(); 2179 2180 ALOGV("Thread %p type %d exiting", this, mType); 2181 return false; 2182} 2183 2184// shared by MIXER and DIRECT, overridden by DUPLICATING 2185void AudioFlinger::PlaybackThread::threadLoop_write() 2186{ 2187 // FIXME rewrite to reduce number of system calls 2188 mLastWriteTime = systemTime(); 2189 mInWrite = true; 2190 mBytesWritten += mixBufferSize; 2191 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2192 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2193 mNumWrites++; 2194 mInWrite = false; 2195} 2196 2197// shared by MIXER and DIRECT, overridden by DUPLICATING 2198void AudioFlinger::PlaybackThread::threadLoop_standby() 2199{ 2200 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2201 mOutput->stream->common.standby(&mOutput->stream->common); 2202} 2203 2204void AudioFlinger::MixerThread::threadLoop_mix() 2205{ 2206 // obtain the presentation timestamp of the next output buffer 2207 int64_t pts; 2208 status_t status = INVALID_OPERATION; 2209 2210 if (NULL != mOutput->stream->get_next_write_timestamp) { 2211 status = mOutput->stream->get_next_write_timestamp( 2212 mOutput->stream, &pts); 2213 } 2214 2215 if (status != NO_ERROR) { 2216 pts = AudioBufferProvider::kInvalidPTS; 2217 } 2218 2219 // mix buffers... 2220 mAudioMixer->process(pts); 2221 // increase sleep time progressively when application underrun condition clears. 2222 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2223 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2224 // such that we would underrun the audio HAL. 2225 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2226 sleepTimeShift--; 2227 } 2228 sleepTime = 0; 2229 standbyTime = systemTime() + standbyDelay; 2230 //TODO: delay standby when effects have a tail 2231} 2232 2233void AudioFlinger::MixerThread::threadLoop_sleepTime() 2234{ 2235 // If no tracks are ready, sleep once for the duration of an output 2236 // buffer size, then write 0s to the output 2237 if (sleepTime == 0) { 2238 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 2239 sleepTime = activeSleepTime >> sleepTimeShift; 2240 if (sleepTime < kMinThreadSleepTimeUs) { 2241 sleepTime = kMinThreadSleepTimeUs; 2242 } 2243 // reduce sleep time in case of consecutive application underruns to avoid 2244 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2245 // duration we would end up writing less data than needed by the audio HAL if 2246 // the condition persists. 2247 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2248 sleepTimeShift++; 2249 } 2250 } else { 2251 sleepTime = idleSleepTime; 2252 } 2253 } else if (mBytesWritten != 0 || 2254 (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2255 memset (mMixBuffer, 0, mixBufferSize); 2256 sleepTime = 0; 2257 ALOGV_IF((mBytesWritten == 0 && (mMixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2258 } 2259 // TODO add standby time extension fct of effect tail 2260} 2261 2262// prepareTracks_l() must be called with ThreadBase::mLock held 2263AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2264 Vector< sp<Track> > *tracksToRemove) 2265{ 2266 2267 mixer_state mixerStatus = MIXER_IDLE; 2268 // find out which tracks need to be processed 2269 size_t count = mActiveTracks.size(); 2270 size_t mixedTracks = 0; 2271 size_t tracksWithEffect = 0; 2272 2273 float masterVolume = mMasterVolume; 2274 bool masterMute = mMasterMute; 2275 2276 if (masterMute) { 2277 masterVolume = 0; 2278 } 2279 // Delegate master volume control to effect in output mix effect chain if needed 2280 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2281 if (chain != 0) { 2282 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2283 chain->setVolume_l(&v, &v); 2284 masterVolume = (float)((v + (1 << 23)) >> 24); 2285 chain.clear(); 2286 } 2287 2288 for (size_t i=0 ; i<count ; i++) { 2289 sp<Track> t = mActiveTracks[i].promote(); 2290 if (t == 0) continue; 2291 2292 // this const just means the local variable doesn't change 2293 Track* const track = t.get(); 2294 audio_track_cblk_t* cblk = track->cblk(); 2295 2296 // The first time a track is added we wait 2297 // for all its buffers to be filled before processing it 2298 int name = track->name(); 2299 // make sure that we have enough frames to mix one full buffer. 2300 // enforce this condition only once to enable draining the buffer in case the client 2301 // app does not call stop() and relies on underrun to stop: 2302 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2303 // during last round 2304 uint32_t minFrames = 1; 2305 if (!track->isStopped() && !track->isPausing() && 2306 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2307 if (t->sampleRate() == (int)mSampleRate) { 2308 minFrames = mFrameCount; 2309 } else { 2310 // +1 for rounding and +1 for additional sample needed for interpolation 2311 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2312 // add frames already consumed but not yet released by the resampler 2313 // because cblk->framesReady() will include these frames 2314 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2315 // the minimum track buffer size is normally twice the number of frames necessary 2316 // to fill one buffer and the resampler should not leave more than one buffer worth 2317 // of unreleased frames after each pass, but just in case... 2318 ALOG_ASSERT(minFrames <= cblk->frameCount); 2319 } 2320 } 2321 if ((track->framesReady() >= minFrames) && track->isReady() && 2322 !track->isPaused() && !track->isTerminated()) 2323 { 2324 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2325 2326 mixedTracks++; 2327 2328 // track->mainBuffer() != mMixBuffer means there is an effect chain 2329 // connected to the track 2330 chain.clear(); 2331 if (track->mainBuffer() != mMixBuffer) { 2332 chain = getEffectChain_l(track->sessionId()); 2333 // Delegate volume control to effect in track effect chain if needed 2334 if (chain != 0) { 2335 tracksWithEffect++; 2336 } else { 2337 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2338 name, track->sessionId()); 2339 } 2340 } 2341 2342 2343 int param = AudioMixer::VOLUME; 2344 if (track->mFillingUpStatus == Track::FS_FILLED) { 2345 // no ramp for the first volume setting 2346 track->mFillingUpStatus = Track::FS_ACTIVE; 2347 if (track->mState == TrackBase::RESUMING) { 2348 track->mState = TrackBase::ACTIVE; 2349 param = AudioMixer::RAMP_VOLUME; 2350 } 2351 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2352 } else if (cblk->server != 0) { 2353 // If the track is stopped before the first frame was mixed, 2354 // do not apply ramp 2355 param = AudioMixer::RAMP_VOLUME; 2356 } 2357 2358 // compute volume for this track 2359 uint32_t vl, vr, va; 2360 if (track->isMuted() || track->isPausing() || 2361 mStreamTypes[track->streamType()].mute) { 2362 vl = vr = va = 0; 2363 if (track->isPausing()) { 2364 track->setPaused(); 2365 } 2366 } else { 2367 2368 // read original volumes with volume control 2369 float typeVolume = mStreamTypes[track->streamType()].volume; 2370 float v = masterVolume * typeVolume; 2371 uint32_t vlr = cblk->getVolumeLR(); 2372 vl = vlr & 0xFFFF; 2373 vr = vlr >> 16; 2374 // track volumes come from shared memory, so can't be trusted and must be clamped 2375 if (vl > MAX_GAIN_INT) { 2376 ALOGV("Track left volume out of range: %04X", vl); 2377 vl = MAX_GAIN_INT; 2378 } 2379 if (vr > MAX_GAIN_INT) { 2380 ALOGV("Track right volume out of range: %04X", vr); 2381 vr = MAX_GAIN_INT; 2382 } 2383 // now apply the master volume and stream type volume 2384 vl = (uint32_t)(v * vl) << 12; 2385 vr = (uint32_t)(v * vr) << 12; 2386 // assuming master volume and stream type volume each go up to 1.0, 2387 // vl and vr are now in 8.24 format 2388 2389 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2390 // send level comes from shared memory and so may be corrupt 2391 if (sendLevel > MAX_GAIN_INT) { 2392 ALOGV("Track send level out of range: %04X", sendLevel); 2393 sendLevel = MAX_GAIN_INT; 2394 } 2395 va = (uint32_t)(v * sendLevel); 2396 } 2397 // Delegate volume control to effect in track effect chain if needed 2398 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2399 // Do not ramp volume if volume is controlled by effect 2400 param = AudioMixer::VOLUME; 2401 track->mHasVolumeController = true; 2402 } else { 2403 // force no volume ramp when volume controller was just disabled or removed 2404 // from effect chain to avoid volume spike 2405 if (track->mHasVolumeController) { 2406 param = AudioMixer::VOLUME; 2407 } 2408 track->mHasVolumeController = false; 2409 } 2410 2411 // Convert volumes from 8.24 to 4.12 format 2412 // This additional clamping is needed in case chain->setVolume_l() overshot 2413 vl = (vl + (1 << 11)) >> 12; 2414 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2415 vr = (vr + (1 << 11)) >> 12; 2416 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2417 2418 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2419 2420 // XXX: these things DON'T need to be done each time 2421 mAudioMixer->setBufferProvider(name, track); 2422 mAudioMixer->enable(name); 2423 2424 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2425 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2426 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2427 mAudioMixer->setParameter( 2428 name, 2429 AudioMixer::TRACK, 2430 AudioMixer::FORMAT, (void *)track->format()); 2431 mAudioMixer->setParameter( 2432 name, 2433 AudioMixer::TRACK, 2434 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2435 mAudioMixer->setParameter( 2436 name, 2437 AudioMixer::RESAMPLE, 2438 AudioMixer::SAMPLE_RATE, 2439 (void *)(cblk->sampleRate)); 2440 mAudioMixer->setParameter( 2441 name, 2442 AudioMixer::TRACK, 2443 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2444 mAudioMixer->setParameter( 2445 name, 2446 AudioMixer::TRACK, 2447 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2448 2449 // reset retry count 2450 track->mRetryCount = kMaxTrackRetries; 2451 // If one track is ready, set the mixer ready if: 2452 // - the mixer was not ready during previous round OR 2453 // - no other track is not ready 2454 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2455 mixerStatus != MIXER_TRACKS_ENABLED) { 2456 mixerStatus = MIXER_TRACKS_READY; 2457 } 2458 } else { 2459 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2460 if (track->isStopped()) { 2461 track->reset(); 2462 } 2463 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2464 // We have consumed all the buffers of this track. 2465 // Remove it from the list of active tracks. 2466 tracksToRemove->add(track); 2467 } else { 2468 // No buffers for this track. Give it a few chances to 2469 // fill a buffer, then remove it from active list. 2470 if (--(track->mRetryCount) <= 0) { 2471 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2472 tracksToRemove->add(track); 2473 // indicate to client process that the track was disabled because of underrun 2474 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2475 // If one track is not ready, mark the mixer also not ready if: 2476 // - the mixer was ready during previous round OR 2477 // - no other track is ready 2478 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2479 mixerStatus != MIXER_TRACKS_READY) { 2480 mixerStatus = MIXER_TRACKS_ENABLED; 2481 } 2482 } 2483 mAudioMixer->disable(name); 2484 } 2485 } 2486 2487 // remove all the tracks that need to be... 2488 count = tracksToRemove->size(); 2489 if (CC_UNLIKELY(count)) { 2490 for (size_t i=0 ; i<count ; i++) { 2491 const sp<Track>& track = tracksToRemove->itemAt(i); 2492 mActiveTracks.remove(track); 2493 if (track->mainBuffer() != mMixBuffer) { 2494 chain = getEffectChain_l(track->sessionId()); 2495 if (chain != 0) { 2496 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2497 chain->decActiveTrackCnt(); 2498 } 2499 } 2500 if (track->isTerminated()) { 2501 removeTrack_l(track); 2502 } 2503 } 2504 } 2505 2506 // mix buffer must be cleared if all tracks are connected to an 2507 // effect chain as in this case the mixer will not write to 2508 // mix buffer and track effects will accumulate into it 2509 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2510 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2511 } 2512 2513 return mixerStatus; 2514} 2515 2516/* 2517The derived values that are cached: 2518 - mixBufferSize from frame count * frame size 2519 - activeSleepTime from activeSleepTimeUs() 2520 - idleSleepTime from idleSleepTimeUs() 2521 - standbyDelay from mActiveSleepTimeUs (DIRECT only) 2522 - maxPeriod from frame count and sample rate (MIXER only) 2523 2524The parameters that affect these derived values are: 2525 - frame count 2526 - frame size 2527 - sample rate 2528 - device type: A2DP or not 2529 - device latency 2530 - format: PCM or not 2531 - active sleep time 2532 - idle sleep time 2533*/ 2534 2535void AudioFlinger::PlaybackThread::cacheParameters_l() 2536{ 2537 mixBufferSize = mFrameCount * mFrameSize; 2538 activeSleepTime = activeSleepTimeUs(); 2539 idleSleepTime = idleSleepTimeUs(); 2540} 2541 2542void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2543{ 2544 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2545 this, streamType, mTracks.size()); 2546 Mutex::Autolock _l(mLock); 2547 2548 size_t size = mTracks.size(); 2549 for (size_t i = 0; i < size; i++) { 2550 sp<Track> t = mTracks[i]; 2551 if (t->streamType() == streamType) { 2552 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2553 t->mCblk->cv.signal(); 2554 } 2555 } 2556} 2557 2558void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2559{ 2560 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2561 this, streamType, valid); 2562 Mutex::Autolock _l(mLock); 2563 2564 mStreamTypes[streamType].valid = valid; 2565} 2566 2567// getTrackName_l() must be called with ThreadBase::mLock held 2568int AudioFlinger::MixerThread::getTrackName_l() 2569{ 2570 return mAudioMixer->getTrackName(); 2571} 2572 2573// deleteTrackName_l() must be called with ThreadBase::mLock held 2574void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2575{ 2576 ALOGV("remove track (%d) and delete from mixer", name); 2577 mAudioMixer->deleteTrackName(name); 2578} 2579 2580// checkForNewParameters_l() must be called with ThreadBase::mLock held 2581bool AudioFlinger::MixerThread::checkForNewParameters_l() 2582{ 2583 bool reconfig = false; 2584 2585 while (!mNewParameters.isEmpty()) { 2586 status_t status = NO_ERROR; 2587 String8 keyValuePair = mNewParameters[0]; 2588 AudioParameter param = AudioParameter(keyValuePair); 2589 int value; 2590 2591 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2592 reconfig = true; 2593 } 2594 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2595 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2596 status = BAD_VALUE; 2597 } else { 2598 reconfig = true; 2599 } 2600 } 2601 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2602 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2603 status = BAD_VALUE; 2604 } else { 2605 reconfig = true; 2606 } 2607 } 2608 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2609 // do not accept frame count changes if tracks are open as the track buffer 2610 // size depends on frame count and correct behavior would not be guaranteed 2611 // if frame count is changed after track creation 2612 if (!mTracks.isEmpty()) { 2613 status = INVALID_OPERATION; 2614 } else { 2615 reconfig = true; 2616 } 2617 } 2618 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2619 // when changing the audio output device, call addBatteryData to notify 2620 // the change 2621 if ((int)mDevice != value) { 2622 uint32_t params = 0; 2623 // check whether speaker is on 2624 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2625 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2626 } 2627 2628 int deviceWithoutSpeaker 2629 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2630 // check if any other device (except speaker) is on 2631 if (value & deviceWithoutSpeaker ) { 2632 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2633 } 2634 2635 if (params != 0) { 2636 addBatteryData(params); 2637 } 2638 } 2639 2640 // forward device change to effects that have requested to be 2641 // aware of attached audio device. 2642 mDevice = (uint32_t)value; 2643 for (size_t i = 0; i < mEffectChains.size(); i++) { 2644 mEffectChains[i]->setDevice_l(mDevice); 2645 } 2646 } 2647 2648 if (status == NO_ERROR) { 2649 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2650 keyValuePair.string()); 2651 if (!mStandby && status == INVALID_OPERATION) { 2652 mOutput->stream->common.standby(&mOutput->stream->common); 2653 mStandby = true; 2654 mBytesWritten = 0; 2655 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2656 keyValuePair.string()); 2657 } 2658 if (status == NO_ERROR && reconfig) { 2659 delete mAudioMixer; 2660 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2661 mAudioMixer = NULL; 2662 readOutputParameters(); 2663 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2664 for (size_t i = 0; i < mTracks.size() ; i++) { 2665 int name = getTrackName_l(); 2666 if (name < 0) break; 2667 mTracks[i]->mName = name; 2668 // limit track sample rate to 2 x new output sample rate 2669 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2670 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2671 } 2672 } 2673 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2674 } 2675 } 2676 2677 mNewParameters.removeAt(0); 2678 2679 mParamStatus = status; 2680 mParamCond.signal(); 2681 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2682 // already timed out waiting for the status and will never signal the condition. 2683 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2684 } 2685 return reconfig; 2686} 2687 2688status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2689{ 2690 const size_t SIZE = 256; 2691 char buffer[SIZE]; 2692 String8 result; 2693 2694 PlaybackThread::dumpInternals(fd, args); 2695 2696 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2697 result.append(buffer); 2698 write(fd, result.string(), result.size()); 2699 return NO_ERROR; 2700} 2701 2702uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2703{ 2704 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2705} 2706 2707uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2708{ 2709 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2710} 2711 2712void AudioFlinger::MixerThread::cacheParameters_l() 2713{ 2714 PlaybackThread::cacheParameters_l(); 2715 2716 // FIXME: Relaxed timing because of a certain device that can't meet latency 2717 // Should be reduced to 2x after the vendor fixes the driver issue 2718 // increase threshold again due to low power audio mode. The way this warning 2719 // threshold is calculated and its usefulness should be reconsidered anyway. 2720 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2721} 2722 2723// ---------------------------------------------------------------------------- 2724AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2725 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2726 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2727 // mLeftVolFloat, mRightVolFloat 2728 // mLeftVolShort, mRightVolShort 2729{ 2730} 2731 2732AudioFlinger::DirectOutputThread::~DirectOutputThread() 2733{ 2734} 2735 2736AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2737 Vector< sp<Track> > *tracksToRemove 2738) 2739{ 2740 sp<Track> trackToRemove; 2741 2742 mixer_state mixerStatus = MIXER_IDLE; 2743 2744 // find out which tracks need to be processed 2745 if (mActiveTracks.size() != 0) { 2746 sp<Track> t = mActiveTracks[0].promote(); 2747 // The track died recently 2748 if (t == 0) return MIXER_IDLE; 2749 2750 Track* const track = t.get(); 2751 audio_track_cblk_t* cblk = track->cblk(); 2752 2753 // The first time a track is added we wait 2754 // for all its buffers to be filled before processing it 2755 if (cblk->framesReady() && track->isReady() && 2756 !track->isPaused() && !track->isTerminated()) 2757 { 2758 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2759 2760 if (track->mFillingUpStatus == Track::FS_FILLED) { 2761 track->mFillingUpStatus = Track::FS_ACTIVE; 2762 mLeftVolFloat = mRightVolFloat = 0; 2763 mLeftVolShort = mRightVolShort = 0; 2764 if (track->mState == TrackBase::RESUMING) { 2765 track->mState = TrackBase::ACTIVE; 2766 rampVolume = true; 2767 } 2768 } else if (cblk->server != 0) { 2769 // If the track is stopped before the first frame was mixed, 2770 // do not apply ramp 2771 rampVolume = true; 2772 } 2773 // compute volume for this track 2774 float left, right; 2775 if (track->isMuted() || mMasterMute || track->isPausing() || 2776 mStreamTypes[track->streamType()].mute) { 2777 left = right = 0; 2778 if (track->isPausing()) { 2779 track->setPaused(); 2780 } 2781 } else { 2782 float typeVolume = mStreamTypes[track->streamType()].volume; 2783 float v = mMasterVolume * typeVolume; 2784 uint32_t vlr = cblk->getVolumeLR(); 2785 float v_clamped = v * (vlr & 0xFFFF); 2786 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2787 left = v_clamped/MAX_GAIN; 2788 v_clamped = v * (vlr >> 16); 2789 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2790 right = v_clamped/MAX_GAIN; 2791 } 2792 2793 if (left != mLeftVolFloat || right != mRightVolFloat) { 2794 mLeftVolFloat = left; 2795 mRightVolFloat = right; 2796 2797 // If audio HAL implements volume control, 2798 // force software volume to nominal value 2799 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2800 left = 1.0f; 2801 right = 1.0f; 2802 } 2803 2804 // Convert volumes from float to 8.24 2805 uint32_t vl = (uint32_t)(left * (1 << 24)); 2806 uint32_t vr = (uint32_t)(right * (1 << 24)); 2807 2808 // Delegate volume control to effect in track effect chain if needed 2809 // only one effect chain can be present on DirectOutputThread, so if 2810 // there is one, the track is connected to it 2811 if (!mEffectChains.isEmpty()) { 2812 // Do not ramp volume if volume is controlled by effect 2813 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2814 rampVolume = false; 2815 } 2816 } 2817 2818 // Convert volumes from 8.24 to 4.12 format 2819 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2820 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2821 leftVol = (uint16_t)v_clamped; 2822 v_clamped = (vr + (1 << 11)) >> 12; 2823 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2824 rightVol = (uint16_t)v_clamped; 2825 } else { 2826 leftVol = mLeftVolShort; 2827 rightVol = mRightVolShort; 2828 rampVolume = false; 2829 } 2830 2831 // reset retry count 2832 track->mRetryCount = kMaxTrackRetriesDirect; 2833 mActiveTrack = t; 2834 mixerStatus = MIXER_TRACKS_READY; 2835 } else { 2836 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2837 if (track->isStopped()) { 2838 track->reset(); 2839 } 2840 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2841 // We have consumed all the buffers of this track. 2842 // Remove it from the list of active tracks. 2843 trackToRemove = track; 2844 } else { 2845 // No buffers for this track. Give it a few chances to 2846 // fill a buffer, then remove it from active list. 2847 if (--(track->mRetryCount) <= 0) { 2848 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2849 trackToRemove = track; 2850 } else { 2851 mixerStatus = MIXER_TRACKS_ENABLED; 2852 } 2853 } 2854 } 2855 } 2856 2857 // FIXME merge this with similar code for removing multiple tracks 2858 // remove all the tracks that need to be... 2859 if (CC_UNLIKELY(trackToRemove != 0)) { 2860 tracksToRemove->add(trackToRemove); 2861 mActiveTracks.remove(trackToRemove); 2862 if (!mEffectChains.isEmpty()) { 2863 ALOGV("stopping track on chain %p for session Id: %d", mEffectChains[0].get(), 2864 trackToRemove->sessionId()); 2865 mEffectChains[0]->decActiveTrackCnt(); 2866 } 2867 if (trackToRemove->isTerminated()) { 2868 removeTrack_l(trackToRemove); 2869 } 2870 } 2871 2872 return mixerStatus; 2873} 2874 2875void AudioFlinger::DirectOutputThread::threadLoop_mix() 2876{ 2877 AudioBufferProvider::Buffer buffer; 2878 size_t frameCount = mFrameCount; 2879 int8_t *curBuf = (int8_t *)mMixBuffer; 2880 // output audio to hardware 2881 while (frameCount) { 2882 buffer.frameCount = frameCount; 2883 mActiveTrack->getNextBuffer(&buffer); 2884 if (CC_UNLIKELY(buffer.raw == NULL)) { 2885 memset(curBuf, 0, frameCount * mFrameSize); 2886 break; 2887 } 2888 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2889 frameCount -= buffer.frameCount; 2890 curBuf += buffer.frameCount * mFrameSize; 2891 mActiveTrack->releaseBuffer(&buffer); 2892 } 2893 sleepTime = 0; 2894 standbyTime = systemTime() + standbyDelay; 2895 mActiveTrack.clear(); 2896 2897 // apply volume 2898 2899 // Do not apply volume on compressed audio 2900 if (!audio_is_linear_pcm(mFormat)) { 2901 return; 2902 } 2903 2904 // convert to signed 16 bit before volume calculation 2905 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2906 size_t count = mFrameCount * mChannelCount; 2907 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2908 int16_t *dst = mMixBuffer + count-1; 2909 while (count--) { 2910 *dst-- = (int16_t)(*src--^0x80) << 8; 2911 } 2912 } 2913 2914 frameCount = mFrameCount; 2915 int16_t *out = mMixBuffer; 2916 if (rampVolume) { 2917 if (mChannelCount == 1) { 2918 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2919 int32_t vlInc = d / (int32_t)frameCount; 2920 int32_t vl = ((int32_t)mLeftVolShort << 16); 2921 do { 2922 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2923 out++; 2924 vl += vlInc; 2925 } while (--frameCount); 2926 2927 } else { 2928 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2929 int32_t vlInc = d / (int32_t)frameCount; 2930 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2931 int32_t vrInc = d / (int32_t)frameCount; 2932 int32_t vl = ((int32_t)mLeftVolShort << 16); 2933 int32_t vr = ((int32_t)mRightVolShort << 16); 2934 do { 2935 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2936 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2937 out += 2; 2938 vl += vlInc; 2939 vr += vrInc; 2940 } while (--frameCount); 2941 } 2942 } else { 2943 if (mChannelCount == 1) { 2944 do { 2945 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2946 out++; 2947 } while (--frameCount); 2948 } else { 2949 do { 2950 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2951 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2952 out += 2; 2953 } while (--frameCount); 2954 } 2955 } 2956 2957 // convert back to unsigned 8 bit after volume calculation 2958 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2959 size_t count = mFrameCount * mChannelCount; 2960 int16_t *src = mMixBuffer; 2961 uint8_t *dst = (uint8_t *)mMixBuffer; 2962 while (count--) { 2963 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2964 } 2965 } 2966 2967 mLeftVolShort = leftVol; 2968 mRightVolShort = rightVol; 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 3077void AudioFlinger::DirectOutputThread::cacheParameters_l() 3078{ 3079 PlaybackThread::cacheParameters_l(); 3080 3081 // use shorter standby delay as on normal output to release 3082 // hardware resources as soon as possible 3083 standbyDelay = microseconds(activeSleepTime*2); 3084} 3085 3086// ---------------------------------------------------------------------------- 3087 3088AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3089 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3090 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3091 mWaitTimeMs(UINT_MAX) 3092{ 3093 addOutputTrack(mainThread); 3094} 3095 3096AudioFlinger::DuplicatingThread::~DuplicatingThread() 3097{ 3098 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3099 mOutputTracks[i]->destroy(); 3100 } 3101} 3102 3103void AudioFlinger::DuplicatingThread::threadLoop_mix() 3104{ 3105 // mix buffers... 3106 if (outputsReady(outputTracks)) { 3107 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3108 } else { 3109 memset(mMixBuffer, 0, mixBufferSize); 3110 } 3111 sleepTime = 0; 3112 writeFrames = mFrameCount; 3113} 3114 3115void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3116{ 3117 if (sleepTime == 0) { 3118 if (mMixerStatus == MIXER_TRACKS_ENABLED) { 3119 sleepTime = activeSleepTime; 3120 } else { 3121 sleepTime = idleSleepTime; 3122 } 3123 } else if (mBytesWritten != 0) { 3124 // flush remaining overflow buffers in output tracks 3125 for (size_t i = 0; i < outputTracks.size(); i++) { 3126 if (outputTracks[i]->isActive()) { 3127 sleepTime = 0; 3128 writeFrames = 0; 3129 memset(mMixBuffer, 0, mixBufferSize); 3130 break; 3131 } 3132 } 3133 } 3134} 3135 3136void AudioFlinger::DuplicatingThread::threadLoop_write() 3137{ 3138 standbyTime = systemTime() + standbyDelay; 3139 for (size_t i = 0; i < outputTracks.size(); i++) { 3140 outputTracks[i]->write(mMixBuffer, writeFrames); 3141 } 3142 mBytesWritten += mixBufferSize; 3143} 3144 3145void AudioFlinger::DuplicatingThread::threadLoop_standby() 3146{ 3147 // DuplicatingThread implements standby by stopping all tracks 3148 for (size_t i = 0; i < outputTracks.size(); i++) { 3149 outputTracks[i]->stop(); 3150 } 3151} 3152 3153void AudioFlinger::DuplicatingThread::saveOutputTracks() 3154{ 3155 outputTracks = mOutputTracks; 3156} 3157 3158void AudioFlinger::DuplicatingThread::clearOutputTracks() 3159{ 3160 outputTracks.clear(); 3161} 3162 3163void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3164{ 3165 Mutex::Autolock _l(mLock); 3166 // FIXME explain this formula 3167 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3168 OutputTrack *outputTrack = new OutputTrack(thread, 3169 this, 3170 mSampleRate, 3171 mFormat, 3172 mChannelMask, 3173 frameCount); 3174 if (outputTrack->cblk() != NULL) { 3175 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3176 mOutputTracks.add(outputTrack); 3177 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3178 updateWaitTime_l(); 3179 } 3180} 3181 3182void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3183{ 3184 Mutex::Autolock _l(mLock); 3185 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3186 if (mOutputTracks[i]->thread() == thread) { 3187 mOutputTracks[i]->destroy(); 3188 mOutputTracks.removeAt(i); 3189 updateWaitTime_l(); 3190 return; 3191 } 3192 } 3193 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3194} 3195 3196// caller must hold mLock 3197void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3198{ 3199 mWaitTimeMs = UINT_MAX; 3200 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3201 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3202 if (strong != 0) { 3203 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3204 if (waitTimeMs < mWaitTimeMs) { 3205 mWaitTimeMs = waitTimeMs; 3206 } 3207 } 3208 } 3209} 3210 3211 3212bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3213{ 3214 for (size_t i = 0; i < outputTracks.size(); i++) { 3215 sp<ThreadBase> thread = outputTracks[i]->thread().promote(); 3216 if (thread == 0) { 3217 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3218 return false; 3219 } 3220 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3221 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3222 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3223 return false; 3224 } 3225 } 3226 return true; 3227} 3228 3229uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3230{ 3231 return (mWaitTimeMs * 1000) / 2; 3232} 3233 3234void AudioFlinger::DuplicatingThread::cacheParameters_l() 3235{ 3236 // updateWaitTime_l() sets mWaitTimeMs, which affects activeSleepTimeUs(), so call it first 3237 updateWaitTime_l(); 3238 3239 MixerThread::cacheParameters_l(); 3240} 3241 3242// ---------------------------------------------------------------------------- 3243 3244// TrackBase constructor must be called with AudioFlinger::mLock held 3245AudioFlinger::ThreadBase::TrackBase::TrackBase( 3246 ThreadBase *thread, 3247 const sp<Client>& client, 3248 uint32_t sampleRate, 3249 audio_format_t format, 3250 uint32_t channelMask, 3251 int frameCount, 3252 const sp<IMemory>& sharedBuffer, 3253 int sessionId) 3254 : RefBase(), 3255 mThread(thread), 3256 mClient(client), 3257 mCblk(NULL), 3258 // mBuffer 3259 // mBufferEnd 3260 mFrameCount(0), 3261 mState(IDLE), 3262 mFormat(format), 3263 mStepServerFailed(false), 3264 mSessionId(sessionId) 3265 // mChannelCount 3266 // mChannelMask 3267{ 3268 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3269 3270 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3271 size_t size = sizeof(audio_track_cblk_t); 3272 uint8_t channelCount = popcount(channelMask); 3273 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3274 if (sharedBuffer == 0) { 3275 size += bufferSize; 3276 } 3277 3278 if (client != NULL) { 3279 mCblkMemory = client->heap()->allocate(size); 3280 if (mCblkMemory != 0) { 3281 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3282 if (mCblk != NULL) { // construct the shared structure in-place. 3283 new(mCblk) audio_track_cblk_t(); 3284 // clear all buffers 3285 mCblk->frameCount = frameCount; 3286 mCblk->sampleRate = sampleRate; 3287 mChannelCount = channelCount; 3288 mChannelMask = channelMask; 3289 if (sharedBuffer == 0) { 3290 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3291 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3292 // Force underrun condition to avoid false underrun callback until first data is 3293 // written to buffer (other flags are cleared) 3294 mCblk->flags = CBLK_UNDERRUN_ON; 3295 } else { 3296 mBuffer = sharedBuffer->pointer(); 3297 } 3298 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3299 } 3300 } else { 3301 ALOGE("not enough memory for AudioTrack size=%u", size); 3302 client->heap()->dump("AudioTrack"); 3303 return; 3304 } 3305 } else { 3306 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3307 // construct the shared structure in-place. 3308 new(mCblk) audio_track_cblk_t(); 3309 // clear all buffers 3310 mCblk->frameCount = frameCount; 3311 mCblk->sampleRate = sampleRate; 3312 mChannelCount = channelCount; 3313 mChannelMask = channelMask; 3314 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3315 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3316 // Force underrun condition to avoid false underrun callback until first data is 3317 // written to buffer (other flags are cleared) 3318 mCblk->flags = CBLK_UNDERRUN_ON; 3319 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3320 } 3321} 3322 3323AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3324{ 3325 if (mCblk != NULL) { 3326 if (mClient == 0) { 3327 delete mCblk; 3328 } else { 3329 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3330 } 3331 } 3332 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3333 if (mClient != 0) { 3334 // Client destructor must run with AudioFlinger mutex locked 3335 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3336 // If the client's reference count drops to zero, the associated destructor 3337 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3338 // relying on the automatic clear() at end of scope. 3339 mClient.clear(); 3340 } 3341} 3342 3343// AudioBufferProvider interface 3344// getNextBuffer() = 0; 3345// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3346void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3347{ 3348 buffer->raw = NULL; 3349 mFrameCount = buffer->frameCount; 3350 (void) step(); // ignore return value of step() 3351 buffer->frameCount = 0; 3352} 3353 3354bool AudioFlinger::ThreadBase::TrackBase::step() { 3355 bool result; 3356 audio_track_cblk_t* cblk = this->cblk(); 3357 3358 result = cblk->stepServer(mFrameCount); 3359 if (!result) { 3360 ALOGV("stepServer failed acquiring cblk mutex"); 3361 mStepServerFailed = true; 3362 } 3363 return result; 3364} 3365 3366void AudioFlinger::ThreadBase::TrackBase::reset() { 3367 audio_track_cblk_t* cblk = this->cblk(); 3368 3369 cblk->user = 0; 3370 cblk->server = 0; 3371 cblk->userBase = 0; 3372 cblk->serverBase = 0; 3373 mStepServerFailed = false; 3374 ALOGV("TrackBase::reset"); 3375} 3376 3377int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3378 return (int)mCblk->sampleRate; 3379} 3380 3381void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3382 audio_track_cblk_t* cblk = this->cblk(); 3383 size_t frameSize = cblk->frameSize; 3384 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3385 int8_t *bufferEnd = bufferStart + frames * frameSize; 3386 3387 // Check validity of returned pointer in case the track control block would have been corrupted. 3388 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3389 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3390 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3391 server %d, serverBase %d, user %d, userBase %d", 3392 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3393 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3394 return NULL; 3395 } 3396 3397 return bufferStart; 3398} 3399 3400// ---------------------------------------------------------------------------- 3401 3402// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3403AudioFlinger::PlaybackThread::Track::Track( 3404 PlaybackThread *thread, 3405 const sp<Client>& client, 3406 audio_stream_type_t streamType, 3407 uint32_t sampleRate, 3408 audio_format_t format, 3409 uint32_t channelMask, 3410 int frameCount, 3411 const sp<IMemory>& sharedBuffer, 3412 int sessionId) 3413 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3414 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3415 mAuxEffectId(0), mHasVolumeController(false) 3416{ 3417 if (mCblk != NULL) { 3418 if (thread != NULL) { 3419 mName = thread->getTrackName_l(); 3420 mMainBuffer = thread->mixBuffer(); 3421 } 3422 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3423 if (mName < 0) { 3424 ALOGE("no more track names available"); 3425 } 3426 mStreamType = streamType; 3427 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3428 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3429 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3430 } 3431} 3432 3433AudioFlinger::PlaybackThread::Track::~Track() 3434{ 3435 ALOGV("PlaybackThread::Track destructor"); 3436 sp<ThreadBase> thread = mThread.promote(); 3437 if (thread != 0) { 3438 Mutex::Autolock _l(thread->mLock); 3439 mState = TERMINATED; 3440 } 3441} 3442 3443void AudioFlinger::PlaybackThread::Track::destroy() 3444{ 3445 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3446 // by removing it from mTracks vector, so there is a risk that this Tracks's 3447 // destructor is called. As the destructor needs to lock mLock, 3448 // we must acquire a strong reference on this Track before locking mLock 3449 // here so that the destructor is called only when exiting this function. 3450 // On the other hand, as long as Track::destroy() is only called by 3451 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3452 // this Track with its member mTrack. 3453 sp<Track> keep(this); 3454 { // scope for mLock 3455 sp<ThreadBase> thread = mThread.promote(); 3456 if (thread != 0) { 3457 if (!isOutputTrack()) { 3458 if (mState == ACTIVE || mState == RESUMING) { 3459 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3460 3461 // to track the speaker usage 3462 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3463 } 3464 AudioSystem::releaseOutput(thread->id()); 3465 } 3466 Mutex::Autolock _l(thread->mLock); 3467 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3468 playbackThread->destroyTrack_l(this); 3469 } 3470 } 3471} 3472 3473void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3474{ 3475 uint32_t vlr = mCblk->getVolumeLR(); 3476 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", 3477 mName - AudioMixer::TRACK0, 3478 (mClient == 0) ? getpid_cached : mClient->pid(), 3479 mStreamType, 3480 mFormat, 3481 mChannelMask, 3482 mSessionId, 3483 mFrameCount, 3484 mState, 3485 mMute, 3486 mFillingUpStatus, 3487 mCblk->sampleRate, 3488 vlr & 0xFFFF, 3489 vlr >> 16, 3490 mCblk->server, 3491 mCblk->user, 3492 (int)mMainBuffer, 3493 (int)mAuxBuffer); 3494} 3495 3496// AudioBufferProvider interface 3497status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3498 AudioBufferProvider::Buffer* buffer, int64_t pts) 3499{ 3500 audio_track_cblk_t* cblk = this->cblk(); 3501 uint32_t framesReady; 3502 uint32_t framesReq = buffer->frameCount; 3503 3504 // Check if last stepServer failed, try to step now 3505 if (mStepServerFailed) { 3506 if (!step()) goto getNextBuffer_exit; 3507 ALOGV("stepServer recovered"); 3508 mStepServerFailed = false; 3509 } 3510 3511 framesReady = cblk->framesReady(); 3512 3513 if (CC_LIKELY(framesReady)) { 3514 uint32_t s = cblk->server; 3515 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3516 3517 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3518 if (framesReq > framesReady) { 3519 framesReq = framesReady; 3520 } 3521 if (s + framesReq > bufferEnd) { 3522 framesReq = bufferEnd - s; 3523 } 3524 3525 buffer->raw = getBuffer(s, framesReq); 3526 if (buffer->raw == NULL) goto getNextBuffer_exit; 3527 3528 buffer->frameCount = framesReq; 3529 return NO_ERROR; 3530 } 3531 3532getNextBuffer_exit: 3533 buffer->raw = NULL; 3534 buffer->frameCount = 0; 3535 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3536 return NOT_ENOUGH_DATA; 3537} 3538 3539uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const { 3540 return mCblk->framesReady(); 3541} 3542 3543bool AudioFlinger::PlaybackThread::Track::isReady() const { 3544 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3545 3546 if (framesReady() >= mCblk->frameCount || 3547 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3548 mFillingUpStatus = FS_FILLED; 3549 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3550 return true; 3551 } 3552 return false; 3553} 3554 3555status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3556{ 3557 status_t status = NO_ERROR; 3558 ALOGV("start(%d), calling pid %d session %d tid %d", 3559 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3560 sp<ThreadBase> thread = mThread.promote(); 3561 if (thread != 0) { 3562 Mutex::Autolock _l(thread->mLock); 3563 track_state state = mState; 3564 // here the track could be either new, or restarted 3565 // in both cases "unstop" the track 3566 if (mState == PAUSED) { 3567 mState = TrackBase::RESUMING; 3568 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3569 } else { 3570 mState = TrackBase::ACTIVE; 3571 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3572 } 3573 3574 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3575 thread->mLock.unlock(); 3576 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3577 thread->mLock.lock(); 3578 3579 // to track the speaker usage 3580 if (status == NO_ERROR) { 3581 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3582 } 3583 } 3584 if (status == NO_ERROR) { 3585 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3586 playbackThread->addTrack_l(this); 3587 } else { 3588 mState = state; 3589 } 3590 } else { 3591 status = BAD_VALUE; 3592 } 3593 return status; 3594} 3595 3596void AudioFlinger::PlaybackThread::Track::stop() 3597{ 3598 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3599 sp<ThreadBase> thread = mThread.promote(); 3600 if (thread != 0) { 3601 Mutex::Autolock _l(thread->mLock); 3602 track_state state = mState; 3603 if (mState > STOPPED) { 3604 mState = STOPPED; 3605 // If the track is not active (PAUSED and buffers full), flush buffers 3606 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3607 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3608 reset(); 3609 } 3610 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3611 } 3612 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3613 thread->mLock.unlock(); 3614 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3615 thread->mLock.lock(); 3616 3617 // to track the speaker usage 3618 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3619 } 3620 } 3621} 3622 3623void AudioFlinger::PlaybackThread::Track::pause() 3624{ 3625 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3626 sp<ThreadBase> thread = mThread.promote(); 3627 if (thread != 0) { 3628 Mutex::Autolock _l(thread->mLock); 3629 if (mState == ACTIVE || mState == RESUMING) { 3630 mState = PAUSING; 3631 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3632 if (!isOutputTrack()) { 3633 thread->mLock.unlock(); 3634 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3635 thread->mLock.lock(); 3636 3637 // to track the speaker usage 3638 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3639 } 3640 } 3641 } 3642} 3643 3644void AudioFlinger::PlaybackThread::Track::flush() 3645{ 3646 ALOGV("flush(%d)", mName); 3647 sp<ThreadBase> thread = mThread.promote(); 3648 if (thread != 0) { 3649 Mutex::Autolock _l(thread->mLock); 3650 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3651 return; 3652 } 3653 // No point remaining in PAUSED state after a flush => go to 3654 // STOPPED state 3655 mState = STOPPED; 3656 3657 // do not reset the track if it is still in the process of being stopped or paused. 3658 // this will be done by prepareTracks_l() when the track is stopped. 3659 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3660 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3661 reset(); 3662 } 3663 } 3664} 3665 3666void AudioFlinger::PlaybackThread::Track::reset() 3667{ 3668 // Do not reset twice to avoid discarding data written just after a flush and before 3669 // the audioflinger thread detects the track is stopped. 3670 if (!mResetDone) { 3671 TrackBase::reset(); 3672 // Force underrun condition to avoid false underrun callback until first data is 3673 // written to buffer 3674 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3675 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3676 mFillingUpStatus = FS_FILLING; 3677 mResetDone = true; 3678 } 3679} 3680 3681void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3682{ 3683 mMute = muted; 3684} 3685 3686status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3687{ 3688 status_t status = DEAD_OBJECT; 3689 sp<ThreadBase> thread = mThread.promote(); 3690 if (thread != 0) { 3691 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3692 status = playbackThread->attachAuxEffect(this, EffectId); 3693 } 3694 return status; 3695} 3696 3697void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3698{ 3699 mAuxEffectId = EffectId; 3700 mAuxBuffer = buffer; 3701} 3702 3703// timed audio tracks 3704 3705sp<AudioFlinger::PlaybackThread::TimedTrack> 3706AudioFlinger::PlaybackThread::TimedTrack::create( 3707 PlaybackThread *thread, 3708 const sp<Client>& client, 3709 audio_stream_type_t streamType, 3710 uint32_t sampleRate, 3711 audio_format_t format, 3712 uint32_t channelMask, 3713 int frameCount, 3714 const sp<IMemory>& sharedBuffer, 3715 int sessionId) { 3716 if (!client->reserveTimedTrack()) 3717 return NULL; 3718 3719 sp<TimedTrack> track = new TimedTrack( 3720 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3721 sharedBuffer, sessionId); 3722 3723 if (track == NULL) { 3724 client->releaseTimedTrack(); 3725 return NULL; 3726 } 3727 3728 return track; 3729} 3730 3731AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3732 PlaybackThread *thread, 3733 const sp<Client>& client, 3734 audio_stream_type_t streamType, 3735 uint32_t sampleRate, 3736 audio_format_t format, 3737 uint32_t channelMask, 3738 int frameCount, 3739 const sp<IMemory>& sharedBuffer, 3740 int sessionId) 3741 : Track(thread, client, streamType, sampleRate, format, channelMask, 3742 frameCount, sharedBuffer, sessionId), 3743 mTimedSilenceBuffer(NULL), 3744 mTimedSilenceBufferSize(0), 3745 mTimedAudioOutputOnTime(false), 3746 mMediaTimeTransformValid(false) 3747{ 3748 LocalClock lc; 3749 mLocalTimeFreq = lc.getLocalFreq(); 3750 3751 mLocalTimeToSampleTransform.a_zero = 0; 3752 mLocalTimeToSampleTransform.b_zero = 0; 3753 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3754 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3755 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3756 &mLocalTimeToSampleTransform.a_to_b_denom); 3757} 3758 3759AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3760 mClient->releaseTimedTrack(); 3761 delete [] mTimedSilenceBuffer; 3762} 3763 3764status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3765 size_t size, sp<IMemory>* buffer) { 3766 3767 Mutex::Autolock _l(mTimedBufferQueueLock); 3768 3769 trimTimedBufferQueue_l(); 3770 3771 // lazily initialize the shared memory heap for timed buffers 3772 if (mTimedMemoryDealer == NULL) { 3773 const int kTimedBufferHeapSize = 512 << 10; 3774 3775 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3776 "AudioFlingerTimed"); 3777 if (mTimedMemoryDealer == NULL) 3778 return NO_MEMORY; 3779 } 3780 3781 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3782 if (newBuffer == NULL) { 3783 newBuffer = mTimedMemoryDealer->allocate(size); 3784 if (newBuffer == NULL) 3785 return NO_MEMORY; 3786 } 3787 3788 *buffer = newBuffer; 3789 return NO_ERROR; 3790} 3791 3792// caller must hold mTimedBufferQueueLock 3793void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3794 int64_t mediaTimeNow; 3795 { 3796 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3797 if (!mMediaTimeTransformValid) 3798 return; 3799 3800 int64_t targetTimeNow; 3801 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3802 ? mCCHelper.getCommonTime(&targetTimeNow) 3803 : mCCHelper.getLocalTime(&targetTimeNow); 3804 3805 if (OK != res) 3806 return; 3807 3808 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3809 &mediaTimeNow)) { 3810 return; 3811 } 3812 } 3813 3814 size_t trimIndex; 3815 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3816 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3817 break; 3818 } 3819 3820 if (trimIndex) { 3821 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3822 } 3823} 3824 3825status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3826 const sp<IMemory>& buffer, int64_t pts) { 3827 3828 { 3829 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3830 if (!mMediaTimeTransformValid) 3831 return INVALID_OPERATION; 3832 } 3833 3834 Mutex::Autolock _l(mTimedBufferQueueLock); 3835 3836 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3837 3838 return NO_ERROR; 3839} 3840 3841status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3842 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3843 3844 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3845 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3846 target); 3847 3848 if (!(target == TimedAudioTrack::LOCAL_TIME || 3849 target == TimedAudioTrack::COMMON_TIME)) { 3850 return BAD_VALUE; 3851 } 3852 3853 Mutex::Autolock lock(mMediaTimeTransformLock); 3854 mMediaTimeTransform = xform; 3855 mMediaTimeTransformTarget = target; 3856 mMediaTimeTransformValid = true; 3857 3858 return NO_ERROR; 3859} 3860 3861#define min(a, b) ((a) < (b) ? (a) : (b)) 3862 3863// implementation of getNextBuffer for tracks whose buffers have timestamps 3864status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3865 AudioBufferProvider::Buffer* buffer, int64_t pts) 3866{ 3867 if (pts == AudioBufferProvider::kInvalidPTS) { 3868 buffer->raw = 0; 3869 buffer->frameCount = 0; 3870 return INVALID_OPERATION; 3871 } 3872 3873 Mutex::Autolock _l(mTimedBufferQueueLock); 3874 3875 while (true) { 3876 3877 // if we have no timed buffers, then fail 3878 if (mTimedBufferQueue.isEmpty()) { 3879 buffer->raw = 0; 3880 buffer->frameCount = 0; 3881 return NOT_ENOUGH_DATA; 3882 } 3883 3884 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3885 3886 // calculate the PTS of the head of the timed buffer queue expressed in 3887 // local time 3888 int64_t headLocalPTS; 3889 { 3890 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3891 3892 ALOG_ASSERT(mMediaTimeTransformValid, "media time transform invalid"); 3893 3894 if (mMediaTimeTransform.a_to_b_denom == 0) { 3895 // the transform represents a pause, so yield silence 3896 timedYieldSilence(buffer->frameCount, buffer); 3897 return NO_ERROR; 3898 } 3899 3900 int64_t transformedPTS; 3901 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3902 &transformedPTS)) { 3903 // the transform failed. this shouldn't happen, but if it does 3904 // then just drop this buffer 3905 ALOGW("timedGetNextBuffer transform failed"); 3906 buffer->raw = 0; 3907 buffer->frameCount = 0; 3908 mTimedBufferQueue.removeAt(0); 3909 return NO_ERROR; 3910 } 3911 3912 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3913 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3914 &headLocalPTS)) { 3915 buffer->raw = 0; 3916 buffer->frameCount = 0; 3917 return INVALID_OPERATION; 3918 } 3919 } else { 3920 headLocalPTS = transformedPTS; 3921 } 3922 } 3923 3924 // adjust the head buffer's PTS to reflect the portion of the head buffer 3925 // that has already been consumed 3926 int64_t effectivePTS = headLocalPTS + 3927 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3928 3929 // Calculate the delta in samples between the head of the input buffer 3930 // queue and the start of the next output buffer that will be written. 3931 // If the transformation fails because of over or underflow, it means 3932 // that the sample's position in the output stream is so far out of 3933 // whack that it should just be dropped. 3934 int64_t sampleDelta; 3935 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3936 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3937 mTimedBufferQueue.removeAt(0); 3938 continue; 3939 } 3940 if (!mLocalTimeToSampleTransform.doForwardTransform( 3941 (effectivePTS - pts) << 32, &sampleDelta)) { 3942 ALOGV("*** too late during sample rate transform: dropped buffer"); 3943 mTimedBufferQueue.removeAt(0); 3944 continue; 3945 } 3946 3947 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 3948 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 3949 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 3950 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 3951 3952 // if the delta between the ideal placement for the next input sample and 3953 // the current output position is within this threshold, then we will 3954 // concatenate the next input samples to the previous output 3955 const int64_t kSampleContinuityThreshold = 3956 (static_cast<int64_t>(sampleRate()) << 32) / 10; 3957 3958 // if this is the first buffer of audio that we're emitting from this track 3959 // then it should be almost exactly on time. 3960 const int64_t kSampleStartupThreshold = 1LL << 32; 3961 3962 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 3963 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 3964 // the next input is close enough to being on time, so concatenate it 3965 // with the last output 3966 timedYieldSamples(buffer); 3967 3968 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3969 return NO_ERROR; 3970 } else if (sampleDelta > 0) { 3971 // the gap between the current output position and the proper start of 3972 // the next input sample is too big, so fill it with silence 3973 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 3974 3975 timedYieldSilence(framesUntilNextInput, buffer); 3976 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 3977 return NO_ERROR; 3978 } else { 3979 // the next input sample is late 3980 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 3981 size_t onTimeSamplePosition = 3982 head.position() + lateFrames * mCblk->frameSize; 3983 3984 if (onTimeSamplePosition > head.buffer()->size()) { 3985 // all the remaining samples in the head are too late, so 3986 // drop it and move on 3987 ALOGV("*** too late: dropped buffer"); 3988 mTimedBufferQueue.removeAt(0); 3989 continue; 3990 } else { 3991 // skip over the late samples 3992 head.setPosition(onTimeSamplePosition); 3993 3994 // yield the available samples 3995 timedYieldSamples(buffer); 3996 3997 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3998 return NO_ERROR; 3999 } 4000 } 4001 } 4002} 4003 4004// Yield samples from the timed buffer queue head up to the given output 4005// buffer's capacity. 4006// 4007// Caller must hold mTimedBufferQueueLock 4008void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 4009 AudioBufferProvider::Buffer* buffer) { 4010 4011 const TimedBuffer& head = mTimedBufferQueue[0]; 4012 4013 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4014 head.position()); 4015 4016 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4017 mCblk->frameSize); 4018 size_t framesRequested = buffer->frameCount; 4019 buffer->frameCount = min(framesLeftInHead, framesRequested); 4020 4021 mTimedAudioOutputOnTime = true; 4022} 4023 4024// Yield samples of silence up to the given output buffer's capacity 4025// 4026// Caller must hold mTimedBufferQueueLock 4027void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4028 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4029 4030 // lazily allocate a buffer filled with silence 4031 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4032 delete [] mTimedSilenceBuffer; 4033 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4034 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4035 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4036 } 4037 4038 buffer->raw = mTimedSilenceBuffer; 4039 size_t framesRequested = buffer->frameCount; 4040 buffer->frameCount = min(numFrames, framesRequested); 4041 4042 mTimedAudioOutputOnTime = false; 4043} 4044 4045// AudioBufferProvider interface 4046void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4047 AudioBufferProvider::Buffer* buffer) { 4048 4049 Mutex::Autolock _l(mTimedBufferQueueLock); 4050 4051 // If the buffer which was just released is part of the buffer at the head 4052 // of the queue, be sure to update the amt of the buffer which has been 4053 // consumed. If the buffer being returned is not part of the head of the 4054 // queue, its either because the buffer is part of the silence buffer, or 4055 // because the head of the timed queue was trimmed after the mixer called 4056 // getNextBuffer but before the mixer called releaseBuffer. 4057 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4058 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4059 4060 void* start = head.buffer()->pointer(); 4061 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4062 4063 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4064 head.setPosition(head.position() + 4065 (buffer->frameCount * mCblk->frameSize)); 4066 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4067 mTimedBufferQueue.removeAt(0); 4068 } 4069 } 4070 } 4071 4072 buffer->raw = 0; 4073 buffer->frameCount = 0; 4074} 4075 4076uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4077 Mutex::Autolock _l(mTimedBufferQueueLock); 4078 4079 uint32_t frames = 0; 4080 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4081 const TimedBuffer& tb = mTimedBufferQueue[i]; 4082 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4083 } 4084 4085 return frames; 4086} 4087 4088AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4089 : mPTS(0), mPosition(0) {} 4090 4091AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4092 const sp<IMemory>& buffer, int64_t pts) 4093 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4094 4095// ---------------------------------------------------------------------------- 4096 4097// RecordTrack constructor must be called with AudioFlinger::mLock held 4098AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4099 RecordThread *thread, 4100 const sp<Client>& client, 4101 uint32_t sampleRate, 4102 audio_format_t format, 4103 uint32_t channelMask, 4104 int frameCount, 4105 int sessionId) 4106 : TrackBase(thread, client, sampleRate, format, 4107 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4108 mOverflow(false) 4109{ 4110 if (mCblk != NULL) { 4111 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4112 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4113 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4114 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4115 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4116 } else { 4117 mCblk->frameSize = sizeof(int8_t); 4118 } 4119 } 4120} 4121 4122AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4123{ 4124 sp<ThreadBase> thread = mThread.promote(); 4125 if (thread != 0) { 4126 AudioSystem::releaseInput(thread->id()); 4127 } 4128} 4129 4130// AudioBufferProvider interface 4131status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4132{ 4133 audio_track_cblk_t* cblk = this->cblk(); 4134 uint32_t framesAvail; 4135 uint32_t framesReq = buffer->frameCount; 4136 4137 // Check if last stepServer failed, try to step now 4138 if (mStepServerFailed) { 4139 if (!step()) goto getNextBuffer_exit; 4140 ALOGV("stepServer recovered"); 4141 mStepServerFailed = false; 4142 } 4143 4144 framesAvail = cblk->framesAvailable_l(); 4145 4146 if (CC_LIKELY(framesAvail)) { 4147 uint32_t s = cblk->server; 4148 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4149 4150 if (framesReq > framesAvail) { 4151 framesReq = framesAvail; 4152 } 4153 if (s + framesReq > bufferEnd) { 4154 framesReq = bufferEnd - s; 4155 } 4156 4157 buffer->raw = getBuffer(s, framesReq); 4158 if (buffer->raw == NULL) goto getNextBuffer_exit; 4159 4160 buffer->frameCount = framesReq; 4161 return NO_ERROR; 4162 } 4163 4164getNextBuffer_exit: 4165 buffer->raw = NULL; 4166 buffer->frameCount = 0; 4167 return NOT_ENOUGH_DATA; 4168} 4169 4170status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4171{ 4172 sp<ThreadBase> thread = mThread.promote(); 4173 if (thread != 0) { 4174 RecordThread *recordThread = (RecordThread *)thread.get(); 4175 return recordThread->start(this, tid); 4176 } else { 4177 return BAD_VALUE; 4178 } 4179} 4180 4181void AudioFlinger::RecordThread::RecordTrack::stop() 4182{ 4183 sp<ThreadBase> thread = mThread.promote(); 4184 if (thread != 0) { 4185 RecordThread *recordThread = (RecordThread *)thread.get(); 4186 recordThread->stop(this); 4187 TrackBase::reset(); 4188 // Force overerrun condition to avoid false overrun callback until first data is 4189 // read from buffer 4190 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4191 } 4192} 4193 4194void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4195{ 4196 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4197 (mClient == 0) ? getpid_cached : mClient->pid(), 4198 mFormat, 4199 mChannelMask, 4200 mSessionId, 4201 mFrameCount, 4202 mState, 4203 mCblk->sampleRate, 4204 mCblk->server, 4205 mCblk->user); 4206} 4207 4208 4209// ---------------------------------------------------------------------------- 4210 4211AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4212 PlaybackThread *playbackThread, 4213 DuplicatingThread *sourceThread, 4214 uint32_t sampleRate, 4215 audio_format_t format, 4216 uint32_t channelMask, 4217 int frameCount) 4218 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4219 mActive(false), mSourceThread(sourceThread) 4220{ 4221 4222 if (mCblk != NULL) { 4223 mCblk->flags |= CBLK_DIRECTION_OUT; 4224 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4225 mOutBuffer.frameCount = 0; 4226 playbackThread->mTracks.add(this); 4227 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4228 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4229 mCblk, mBuffer, mCblk->buffers, 4230 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4231 } else { 4232 ALOGW("Error creating output track on thread %p", playbackThread); 4233 } 4234} 4235 4236AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4237{ 4238 clearBufferQueue(); 4239} 4240 4241status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4242{ 4243 status_t status = Track::start(tid); 4244 if (status != NO_ERROR) { 4245 return status; 4246 } 4247 4248 mActive = true; 4249 mRetryCount = 127; 4250 return status; 4251} 4252 4253void AudioFlinger::PlaybackThread::OutputTrack::stop() 4254{ 4255 Track::stop(); 4256 clearBufferQueue(); 4257 mOutBuffer.frameCount = 0; 4258 mActive = false; 4259} 4260 4261bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4262{ 4263 Buffer *pInBuffer; 4264 Buffer inBuffer; 4265 uint32_t channelCount = mChannelCount; 4266 bool outputBufferFull = false; 4267 inBuffer.frameCount = frames; 4268 inBuffer.i16 = data; 4269 4270 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4271 4272 if (!mActive && frames != 0) { 4273 start(0); 4274 sp<ThreadBase> thread = mThread.promote(); 4275 if (thread != 0) { 4276 MixerThread *mixerThread = (MixerThread *)thread.get(); 4277 if (mCblk->frameCount > frames){ 4278 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4279 uint32_t startFrames = (mCblk->frameCount - frames); 4280 pInBuffer = new Buffer; 4281 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4282 pInBuffer->frameCount = startFrames; 4283 pInBuffer->i16 = pInBuffer->mBuffer; 4284 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4285 mBufferQueue.add(pInBuffer); 4286 } else { 4287 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4288 } 4289 } 4290 } 4291 } 4292 4293 while (waitTimeLeftMs) { 4294 // First write pending buffers, then new data 4295 if (mBufferQueue.size()) { 4296 pInBuffer = mBufferQueue.itemAt(0); 4297 } else { 4298 pInBuffer = &inBuffer; 4299 } 4300 4301 if (pInBuffer->frameCount == 0) { 4302 break; 4303 } 4304 4305 if (mOutBuffer.frameCount == 0) { 4306 mOutBuffer.frameCount = pInBuffer->frameCount; 4307 nsecs_t startTime = systemTime(); 4308 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4309 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4310 outputBufferFull = true; 4311 break; 4312 } 4313 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4314 if (waitTimeLeftMs >= waitTimeMs) { 4315 waitTimeLeftMs -= waitTimeMs; 4316 } else { 4317 waitTimeLeftMs = 0; 4318 } 4319 } 4320 4321 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4322 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4323 mCblk->stepUser(outFrames); 4324 pInBuffer->frameCount -= outFrames; 4325 pInBuffer->i16 += outFrames * channelCount; 4326 mOutBuffer.frameCount -= outFrames; 4327 mOutBuffer.i16 += outFrames * channelCount; 4328 4329 if (pInBuffer->frameCount == 0) { 4330 if (mBufferQueue.size()) { 4331 mBufferQueue.removeAt(0); 4332 delete [] pInBuffer->mBuffer; 4333 delete pInBuffer; 4334 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4335 } else { 4336 break; 4337 } 4338 } 4339 } 4340 4341 // If we could not write all frames, allocate a buffer and queue it for next time. 4342 if (inBuffer.frameCount) { 4343 sp<ThreadBase> thread = mThread.promote(); 4344 if (thread != 0 && !thread->standby()) { 4345 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4346 pInBuffer = new Buffer; 4347 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4348 pInBuffer->frameCount = inBuffer.frameCount; 4349 pInBuffer->i16 = pInBuffer->mBuffer; 4350 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4351 mBufferQueue.add(pInBuffer); 4352 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4353 } else { 4354 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4355 } 4356 } 4357 } 4358 4359 // Calling write() with a 0 length buffer, means that no more data will be written: 4360 // If no more buffers are pending, fill output track buffer to make sure it is started 4361 // by output mixer. 4362 if (frames == 0 && mBufferQueue.size() == 0) { 4363 if (mCblk->user < mCblk->frameCount) { 4364 frames = mCblk->frameCount - mCblk->user; 4365 pInBuffer = new Buffer; 4366 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4367 pInBuffer->frameCount = frames; 4368 pInBuffer->i16 = pInBuffer->mBuffer; 4369 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4370 mBufferQueue.add(pInBuffer); 4371 } else if (mActive) { 4372 stop(); 4373 } 4374 } 4375 4376 return outputBufferFull; 4377} 4378 4379status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4380{ 4381 int active; 4382 status_t result; 4383 audio_track_cblk_t* cblk = mCblk; 4384 uint32_t framesReq = buffer->frameCount; 4385 4386// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4387 buffer->frameCount = 0; 4388 4389 uint32_t framesAvail = cblk->framesAvailable(); 4390 4391 4392 if (framesAvail == 0) { 4393 Mutex::Autolock _l(cblk->lock); 4394 goto start_loop_here; 4395 while (framesAvail == 0) { 4396 active = mActive; 4397 if (CC_UNLIKELY(!active)) { 4398 ALOGV("Not active and NO_MORE_BUFFERS"); 4399 return NO_MORE_BUFFERS; 4400 } 4401 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4402 if (result != NO_ERROR) { 4403 return NO_MORE_BUFFERS; 4404 } 4405 // read the server count again 4406 start_loop_here: 4407 framesAvail = cblk->framesAvailable_l(); 4408 } 4409 } 4410 4411// if (framesAvail < framesReq) { 4412// return NO_MORE_BUFFERS; 4413// } 4414 4415 if (framesReq > framesAvail) { 4416 framesReq = framesAvail; 4417 } 4418 4419 uint32_t u = cblk->user; 4420 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4421 4422 if (u + framesReq > bufferEnd) { 4423 framesReq = bufferEnd - u; 4424 } 4425 4426 buffer->frameCount = framesReq; 4427 buffer->raw = (void *)cblk->buffer(u); 4428 return NO_ERROR; 4429} 4430 4431 4432void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4433{ 4434 size_t size = mBufferQueue.size(); 4435 4436 for (size_t i = 0; i < size; i++) { 4437 Buffer *pBuffer = mBufferQueue.itemAt(i); 4438 delete [] pBuffer->mBuffer; 4439 delete pBuffer; 4440 } 4441 mBufferQueue.clear(); 4442} 4443 4444// ---------------------------------------------------------------------------- 4445 4446AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4447 : RefBase(), 4448 mAudioFlinger(audioFlinger), 4449 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4450 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4451 mPid(pid), 4452 mTimedTrackCount(0) 4453{ 4454 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4455} 4456 4457// Client destructor must be called with AudioFlinger::mLock held 4458AudioFlinger::Client::~Client() 4459{ 4460 mAudioFlinger->removeClient_l(mPid); 4461} 4462 4463sp<MemoryDealer> AudioFlinger::Client::heap() const 4464{ 4465 return mMemoryDealer; 4466} 4467 4468// Reserve one of the limited slots for a timed audio track associated 4469// with this client 4470bool AudioFlinger::Client::reserveTimedTrack() 4471{ 4472 const int kMaxTimedTracksPerClient = 4; 4473 4474 Mutex::Autolock _l(mTimedTrackLock); 4475 4476 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4477 ALOGW("can not create timed track - pid %d has exceeded the limit", 4478 mPid); 4479 return false; 4480 } 4481 4482 mTimedTrackCount++; 4483 return true; 4484} 4485 4486// Release a slot for a timed audio track 4487void AudioFlinger::Client::releaseTimedTrack() 4488{ 4489 Mutex::Autolock _l(mTimedTrackLock); 4490 mTimedTrackCount--; 4491} 4492 4493// ---------------------------------------------------------------------------- 4494 4495AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4496 const sp<IAudioFlingerClient>& client, 4497 pid_t pid) 4498 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4499{ 4500} 4501 4502AudioFlinger::NotificationClient::~NotificationClient() 4503{ 4504} 4505 4506void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4507{ 4508 sp<NotificationClient> keep(this); 4509 mAudioFlinger->removeNotificationClient(mPid); 4510} 4511 4512// ---------------------------------------------------------------------------- 4513 4514AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4515 : BnAudioTrack(), 4516 mTrack(track) 4517{ 4518} 4519 4520AudioFlinger::TrackHandle::~TrackHandle() { 4521 // just stop the track on deletion, associated resources 4522 // will be freed from the main thread once all pending buffers have 4523 // been played. Unless it's not in the active track list, in which 4524 // case we free everything now... 4525 mTrack->destroy(); 4526} 4527 4528sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4529 return mTrack->getCblk(); 4530} 4531 4532status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4533 return mTrack->start(tid); 4534} 4535 4536void AudioFlinger::TrackHandle::stop() { 4537 mTrack->stop(); 4538} 4539 4540void AudioFlinger::TrackHandle::flush() { 4541 mTrack->flush(); 4542} 4543 4544void AudioFlinger::TrackHandle::mute(bool e) { 4545 mTrack->mute(e); 4546} 4547 4548void AudioFlinger::TrackHandle::pause() { 4549 mTrack->pause(); 4550} 4551 4552status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4553{ 4554 return mTrack->attachAuxEffect(EffectId); 4555} 4556 4557status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4558 sp<IMemory>* buffer) { 4559 if (!mTrack->isTimedTrack()) 4560 return INVALID_OPERATION; 4561 4562 PlaybackThread::TimedTrack* tt = 4563 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4564 return tt->allocateTimedBuffer(size, buffer); 4565} 4566 4567status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4568 int64_t pts) { 4569 if (!mTrack->isTimedTrack()) 4570 return INVALID_OPERATION; 4571 4572 PlaybackThread::TimedTrack* tt = 4573 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4574 return tt->queueTimedBuffer(buffer, pts); 4575} 4576 4577status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4578 const LinearTransform& xform, int target) { 4579 4580 if (!mTrack->isTimedTrack()) 4581 return INVALID_OPERATION; 4582 4583 PlaybackThread::TimedTrack* tt = 4584 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4585 return tt->setMediaTimeTransform( 4586 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4587} 4588 4589status_t AudioFlinger::TrackHandle::onTransact( 4590 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4591{ 4592 return BnAudioTrack::onTransact(code, data, reply, flags); 4593} 4594 4595// ---------------------------------------------------------------------------- 4596 4597sp<IAudioRecord> AudioFlinger::openRecord( 4598 pid_t pid, 4599 audio_io_handle_t input, 4600 uint32_t sampleRate, 4601 audio_format_t format, 4602 uint32_t channelMask, 4603 int frameCount, 4604 // FIXME dead, remove from IAudioFlinger 4605 uint32_t flags, 4606 int *sessionId, 4607 status_t *status) 4608{ 4609 sp<RecordThread::RecordTrack> recordTrack; 4610 sp<RecordHandle> recordHandle; 4611 sp<Client> client; 4612 status_t lStatus; 4613 RecordThread *thread; 4614 size_t inFrameCount; 4615 int lSessionId; 4616 4617 // check calling permissions 4618 if (!recordingAllowed()) { 4619 lStatus = PERMISSION_DENIED; 4620 goto Exit; 4621 } 4622 4623 // add client to list 4624 { // scope for mLock 4625 Mutex::Autolock _l(mLock); 4626 thread = checkRecordThread_l(input); 4627 if (thread == NULL) { 4628 lStatus = BAD_VALUE; 4629 goto Exit; 4630 } 4631 4632 client = registerPid_l(pid); 4633 4634 // If no audio session id is provided, create one here 4635 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4636 lSessionId = *sessionId; 4637 } else { 4638 lSessionId = nextUniqueId(); 4639 if (sessionId != NULL) { 4640 *sessionId = lSessionId; 4641 } 4642 } 4643 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4644 recordTrack = thread->createRecordTrack_l(client, 4645 sampleRate, 4646 format, 4647 channelMask, 4648 frameCount, 4649 lSessionId, 4650 &lStatus); 4651 } 4652 if (lStatus != NO_ERROR) { 4653 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4654 // destructor is called by the TrackBase destructor with mLock held 4655 client.clear(); 4656 recordTrack.clear(); 4657 goto Exit; 4658 } 4659 4660 // return to handle to client 4661 recordHandle = new RecordHandle(recordTrack); 4662 lStatus = NO_ERROR; 4663 4664Exit: 4665 if (status) { 4666 *status = lStatus; 4667 } 4668 return recordHandle; 4669} 4670 4671// ---------------------------------------------------------------------------- 4672 4673AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4674 : BnAudioRecord(), 4675 mRecordTrack(recordTrack) 4676{ 4677} 4678 4679AudioFlinger::RecordHandle::~RecordHandle() { 4680 stop(); 4681} 4682 4683sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4684 return mRecordTrack->getCblk(); 4685} 4686 4687status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4688 ALOGV("RecordHandle::start()"); 4689 return mRecordTrack->start(tid); 4690} 4691 4692void AudioFlinger::RecordHandle::stop() { 4693 ALOGV("RecordHandle::stop()"); 4694 mRecordTrack->stop(); 4695} 4696 4697status_t AudioFlinger::RecordHandle::onTransact( 4698 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4699{ 4700 return BnAudioRecord::onTransact(code, data, reply, flags); 4701} 4702 4703// ---------------------------------------------------------------------------- 4704 4705AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4706 AudioStreamIn *input, 4707 uint32_t sampleRate, 4708 uint32_t channels, 4709 audio_io_handle_t id, 4710 uint32_t device) : 4711 ThreadBase(audioFlinger, id, device, RECORD), 4712 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4713 // mRsmpInIndex and mInputBytes set by readInputParameters() 4714 mReqChannelCount(popcount(channels)), 4715 mReqSampleRate(sampleRate) 4716 // mBytesRead is only meaningful while active, and so is cleared in start() 4717 // (but might be better to also clear here for dump?) 4718{ 4719 snprintf(mName, kNameLength, "AudioIn_%X", id); 4720 4721 readInputParameters(); 4722} 4723 4724 4725AudioFlinger::RecordThread::~RecordThread() 4726{ 4727 delete[] mRsmpInBuffer; 4728 delete mResampler; 4729 delete[] mRsmpOutBuffer; 4730} 4731 4732void AudioFlinger::RecordThread::onFirstRef() 4733{ 4734 run(mName, PRIORITY_URGENT_AUDIO); 4735} 4736 4737status_t AudioFlinger::RecordThread::readyToRun() 4738{ 4739 status_t status = initCheck(); 4740 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4741 return status; 4742} 4743 4744bool AudioFlinger::RecordThread::threadLoop() 4745{ 4746 AudioBufferProvider::Buffer buffer; 4747 sp<RecordTrack> activeTrack; 4748 Vector< sp<EffectChain> > effectChains; 4749 4750 nsecs_t lastWarning = 0; 4751 4752 acquireWakeLock(); 4753 4754 // start recording 4755 while (!exitPending()) { 4756 4757 processConfigEvents(); 4758 4759 { // scope for mLock 4760 Mutex::Autolock _l(mLock); 4761 checkForNewParameters_l(); 4762 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4763 if (!mStandby) { 4764 mInput->stream->common.standby(&mInput->stream->common); 4765 mStandby = true; 4766 } 4767 4768 if (exitPending()) break; 4769 4770 releaseWakeLock_l(); 4771 ALOGV("RecordThread: loop stopping"); 4772 // go to sleep 4773 mWaitWorkCV.wait(mLock); 4774 ALOGV("RecordThread: loop starting"); 4775 acquireWakeLock_l(); 4776 continue; 4777 } 4778 if (mActiveTrack != 0) { 4779 if (mActiveTrack->mState == TrackBase::PAUSING) { 4780 if (!mStandby) { 4781 mInput->stream->common.standby(&mInput->stream->common); 4782 mStandby = true; 4783 } 4784 mActiveTrack.clear(); 4785 mStartStopCond.broadcast(); 4786 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4787 if (mReqChannelCount != mActiveTrack->channelCount()) { 4788 mActiveTrack.clear(); 4789 mStartStopCond.broadcast(); 4790 } else if (mBytesRead != 0) { 4791 // record start succeeds only if first read from audio input 4792 // succeeds 4793 if (mBytesRead > 0) { 4794 mActiveTrack->mState = TrackBase::ACTIVE; 4795 } else { 4796 mActiveTrack.clear(); 4797 } 4798 mStartStopCond.broadcast(); 4799 } 4800 mStandby = false; 4801 } 4802 } 4803 lockEffectChains_l(effectChains); 4804 } 4805 4806 if (mActiveTrack != 0) { 4807 if (mActiveTrack->mState != TrackBase::ACTIVE && 4808 mActiveTrack->mState != TrackBase::RESUMING) { 4809 unlockEffectChains(effectChains); 4810 usleep(kRecordThreadSleepUs); 4811 continue; 4812 } 4813 for (size_t i = 0; i < effectChains.size(); i ++) { 4814 effectChains[i]->process_l(); 4815 } 4816 4817 buffer.frameCount = mFrameCount; 4818 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4819 size_t framesOut = buffer.frameCount; 4820 if (mResampler == NULL) { 4821 // no resampling 4822 while (framesOut) { 4823 size_t framesIn = mFrameCount - mRsmpInIndex; 4824 if (framesIn) { 4825 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4826 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4827 if (framesIn > framesOut) 4828 framesIn = framesOut; 4829 mRsmpInIndex += framesIn; 4830 framesOut -= framesIn; 4831 if ((int)mChannelCount == mReqChannelCount || 4832 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4833 memcpy(dst, src, framesIn * mFrameSize); 4834 } else { 4835 int16_t *src16 = (int16_t *)src; 4836 int16_t *dst16 = (int16_t *)dst; 4837 if (mChannelCount == 1) { 4838 while (framesIn--) { 4839 *dst16++ = *src16; 4840 *dst16++ = *src16++; 4841 } 4842 } else { 4843 while (framesIn--) { 4844 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4845 src16 += 2; 4846 } 4847 } 4848 } 4849 } 4850 if (framesOut && mFrameCount == mRsmpInIndex) { 4851 if (framesOut == mFrameCount && 4852 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4853 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4854 framesOut = 0; 4855 } else { 4856 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4857 mRsmpInIndex = 0; 4858 } 4859 if (mBytesRead < 0) { 4860 ALOGE("Error reading audio input"); 4861 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4862 // Force input into standby so that it tries to 4863 // recover at next read attempt 4864 mInput->stream->common.standby(&mInput->stream->common); 4865 usleep(kRecordThreadSleepUs); 4866 } 4867 mRsmpInIndex = mFrameCount; 4868 framesOut = 0; 4869 buffer.frameCount = 0; 4870 } 4871 } 4872 } 4873 } else { 4874 // resampling 4875 4876 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4877 // alter output frame count as if we were expecting stereo samples 4878 if (mChannelCount == 1 && mReqChannelCount == 1) { 4879 framesOut >>= 1; 4880 } 4881 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4882 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4883 // are 32 bit aligned which should be always true. 4884 if (mChannelCount == 2 && mReqChannelCount == 1) { 4885 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4886 // the resampler always outputs stereo samples: do post stereo to mono conversion 4887 int16_t *src = (int16_t *)mRsmpOutBuffer; 4888 int16_t *dst = buffer.i16; 4889 while (framesOut--) { 4890 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4891 src += 2; 4892 } 4893 } else { 4894 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4895 } 4896 4897 } 4898 mActiveTrack->releaseBuffer(&buffer); 4899 mActiveTrack->overflow(); 4900 } 4901 // client isn't retrieving buffers fast enough 4902 else { 4903 if (!mActiveTrack->setOverflow()) { 4904 nsecs_t now = systemTime(); 4905 if ((now - lastWarning) > kWarningThrottleNs) { 4906 ALOGW("RecordThread: buffer overflow"); 4907 lastWarning = now; 4908 } 4909 } 4910 // Release the processor for a while before asking for a new buffer. 4911 // This will give the application more chance to read from the buffer and 4912 // clear the overflow. 4913 usleep(kRecordThreadSleepUs); 4914 } 4915 } 4916 // enable changes in effect chain 4917 unlockEffectChains(effectChains); 4918 effectChains.clear(); 4919 } 4920 4921 if (!mStandby) { 4922 mInput->stream->common.standby(&mInput->stream->common); 4923 } 4924 mActiveTrack.clear(); 4925 4926 mStartStopCond.broadcast(); 4927 4928 releaseWakeLock(); 4929 4930 ALOGV("RecordThread %p exiting", this); 4931 return false; 4932} 4933 4934 4935sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4936 const sp<AudioFlinger::Client>& client, 4937 uint32_t sampleRate, 4938 audio_format_t format, 4939 int channelMask, 4940 int frameCount, 4941 int sessionId, 4942 status_t *status) 4943{ 4944 sp<RecordTrack> track; 4945 status_t lStatus; 4946 4947 lStatus = initCheck(); 4948 if (lStatus != NO_ERROR) { 4949 ALOGE("Audio driver not initialized."); 4950 goto Exit; 4951 } 4952 4953 { // scope for mLock 4954 Mutex::Autolock _l(mLock); 4955 4956 track = new RecordTrack(this, client, sampleRate, 4957 format, channelMask, frameCount, sessionId); 4958 4959 if (track->getCblk() == 0) { 4960 lStatus = NO_MEMORY; 4961 goto Exit; 4962 } 4963 4964 mTrack = track.get(); 4965 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4966 bool suspend = audio_is_bluetooth_sco_device( 4967 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4968 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4969 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4970 } 4971 lStatus = NO_ERROR; 4972 4973Exit: 4974 if (status) { 4975 *status = lStatus; 4976 } 4977 return track; 4978} 4979 4980status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 4981{ 4982 ALOGV("RecordThread::start tid=%d", tid); 4983 sp<ThreadBase> strongMe = this; 4984 status_t status = NO_ERROR; 4985 { 4986 AutoMutex lock(mLock); 4987 if (mActiveTrack != 0) { 4988 if (recordTrack != mActiveTrack.get()) { 4989 status = -EBUSY; 4990 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4991 mActiveTrack->mState = TrackBase::ACTIVE; 4992 } 4993 return status; 4994 } 4995 4996 recordTrack->mState = TrackBase::IDLE; 4997 mActiveTrack = recordTrack; 4998 mLock.unlock(); 4999 status_t status = AudioSystem::startInput(mId); 5000 mLock.lock(); 5001 if (status != NO_ERROR) { 5002 mActiveTrack.clear(); 5003 return status; 5004 } 5005 mRsmpInIndex = mFrameCount; 5006 mBytesRead = 0; 5007 if (mResampler != NULL) { 5008 mResampler->reset(); 5009 } 5010 mActiveTrack->mState = TrackBase::RESUMING; 5011 // signal thread to start 5012 ALOGV("Signal record thread"); 5013 mWaitWorkCV.signal(); 5014 // do not wait for mStartStopCond if exiting 5015 if (exitPending()) { 5016 mActiveTrack.clear(); 5017 status = INVALID_OPERATION; 5018 goto startError; 5019 } 5020 mStartStopCond.wait(mLock); 5021 if (mActiveTrack == 0) { 5022 ALOGV("Record failed to start"); 5023 status = BAD_VALUE; 5024 goto startError; 5025 } 5026 ALOGV("Record started OK"); 5027 return status; 5028 } 5029startError: 5030 AudioSystem::stopInput(mId); 5031 return status; 5032} 5033 5034void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5035 ALOGV("RecordThread::stop"); 5036 sp<ThreadBase> strongMe = this; 5037 { 5038 AutoMutex lock(mLock); 5039 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5040 mActiveTrack->mState = TrackBase::PAUSING; 5041 // do not wait for mStartStopCond if exiting 5042 if (exitPending()) { 5043 return; 5044 } 5045 mStartStopCond.wait(mLock); 5046 // if we have been restarted, recordTrack == mActiveTrack.get() here 5047 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5048 mLock.unlock(); 5049 AudioSystem::stopInput(mId); 5050 mLock.lock(); 5051 ALOGV("Record stopped OK"); 5052 } 5053 } 5054 } 5055} 5056 5057status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5058{ 5059 const size_t SIZE = 256; 5060 char buffer[SIZE]; 5061 String8 result; 5062 5063 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5064 result.append(buffer); 5065 5066 if (mActiveTrack != 0) { 5067 result.append("Active Track:\n"); 5068 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5069 mActiveTrack->dump(buffer, SIZE); 5070 result.append(buffer); 5071 5072 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5073 result.append(buffer); 5074 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5075 result.append(buffer); 5076 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5077 result.append(buffer); 5078 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5079 result.append(buffer); 5080 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5081 result.append(buffer); 5082 5083 5084 } else { 5085 result.append("No record client\n"); 5086 } 5087 write(fd, result.string(), result.size()); 5088 5089 dumpBase(fd, args); 5090 dumpEffectChains(fd, args); 5091 5092 return NO_ERROR; 5093} 5094 5095// AudioBufferProvider interface 5096status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5097{ 5098 size_t framesReq = buffer->frameCount; 5099 size_t framesReady = mFrameCount - mRsmpInIndex; 5100 int channelCount; 5101 5102 if (framesReady == 0) { 5103 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5104 if (mBytesRead < 0) { 5105 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5106 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5107 // Force input into standby so that it tries to 5108 // recover at next read attempt 5109 mInput->stream->common.standby(&mInput->stream->common); 5110 usleep(kRecordThreadSleepUs); 5111 } 5112 buffer->raw = NULL; 5113 buffer->frameCount = 0; 5114 return NOT_ENOUGH_DATA; 5115 } 5116 mRsmpInIndex = 0; 5117 framesReady = mFrameCount; 5118 } 5119 5120 if (framesReq > framesReady) { 5121 framesReq = framesReady; 5122 } 5123 5124 if (mChannelCount == 1 && mReqChannelCount == 2) { 5125 channelCount = 1; 5126 } else { 5127 channelCount = 2; 5128 } 5129 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5130 buffer->frameCount = framesReq; 5131 return NO_ERROR; 5132} 5133 5134// AudioBufferProvider interface 5135void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5136{ 5137 mRsmpInIndex += buffer->frameCount; 5138 buffer->frameCount = 0; 5139} 5140 5141bool AudioFlinger::RecordThread::checkForNewParameters_l() 5142{ 5143 bool reconfig = false; 5144 5145 while (!mNewParameters.isEmpty()) { 5146 status_t status = NO_ERROR; 5147 String8 keyValuePair = mNewParameters[0]; 5148 AudioParameter param = AudioParameter(keyValuePair); 5149 int value; 5150 audio_format_t reqFormat = mFormat; 5151 int reqSamplingRate = mReqSampleRate; 5152 int reqChannelCount = mReqChannelCount; 5153 5154 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5155 reqSamplingRate = value; 5156 reconfig = true; 5157 } 5158 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5159 reqFormat = (audio_format_t) value; 5160 reconfig = true; 5161 } 5162 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5163 reqChannelCount = popcount(value); 5164 reconfig = true; 5165 } 5166 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5167 // do not accept frame count changes if tracks are open as the track buffer 5168 // size depends on frame count and correct behavior would not be guaranteed 5169 // if frame count is changed after track creation 5170 if (mActiveTrack != 0) { 5171 status = INVALID_OPERATION; 5172 } else { 5173 reconfig = true; 5174 } 5175 } 5176 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5177 // forward device change to effects that have requested to be 5178 // aware of attached audio device. 5179 for (size_t i = 0; i < mEffectChains.size(); i++) { 5180 mEffectChains[i]->setDevice_l(value); 5181 } 5182 // store input device and output device but do not forward output device to audio HAL. 5183 // Note that status is ignored by the caller for output device 5184 // (see AudioFlinger::setParameters() 5185 if (value & AUDIO_DEVICE_OUT_ALL) { 5186 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5187 status = BAD_VALUE; 5188 } else { 5189 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5190 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5191 if (mTrack != NULL) { 5192 bool suspend = audio_is_bluetooth_sco_device( 5193 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5194 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5195 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5196 } 5197 } 5198 mDevice |= (uint32_t)value; 5199 } 5200 if (status == NO_ERROR) { 5201 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5202 if (status == INVALID_OPERATION) { 5203 mInput->stream->common.standby(&mInput->stream->common); 5204 status = mInput->stream->common.set_parameters(&mInput->stream->common, 5205 keyValuePair.string()); 5206 } 5207 if (reconfig) { 5208 if (status == BAD_VALUE && 5209 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5210 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5211 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5212 popcount(mInput->stream->common.get_channels(&mInput->stream->common)) <= FCC_2 && 5213 (reqChannelCount <= FCC_2)) { 5214 status = NO_ERROR; 5215 } 5216 if (status == NO_ERROR) { 5217 readInputParameters(); 5218 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5219 } 5220 } 5221 } 5222 5223 mNewParameters.removeAt(0); 5224 5225 mParamStatus = status; 5226 mParamCond.signal(); 5227 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5228 // already timed out waiting for the status and will never signal the condition. 5229 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5230 } 5231 return reconfig; 5232} 5233 5234String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5235{ 5236 char *s; 5237 String8 out_s8 = String8(); 5238 5239 Mutex::Autolock _l(mLock); 5240 if (initCheck() != NO_ERROR) { 5241 return out_s8; 5242 } 5243 5244 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5245 out_s8 = String8(s); 5246 free(s); 5247 return out_s8; 5248} 5249 5250void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5251 AudioSystem::OutputDescriptor desc; 5252 void *param2 = NULL; 5253 5254 switch (event) { 5255 case AudioSystem::INPUT_OPENED: 5256 case AudioSystem::INPUT_CONFIG_CHANGED: 5257 desc.channels = mChannelMask; 5258 desc.samplingRate = mSampleRate; 5259 desc.format = mFormat; 5260 desc.frameCount = mFrameCount; 5261 desc.latency = 0; 5262 param2 = &desc; 5263 break; 5264 5265 case AudioSystem::INPUT_CLOSED: 5266 default: 5267 break; 5268 } 5269 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5270} 5271 5272void AudioFlinger::RecordThread::readInputParameters() 5273{ 5274 delete mRsmpInBuffer; 5275 // mRsmpInBuffer is always assigned a new[] below 5276 delete mRsmpOutBuffer; 5277 mRsmpOutBuffer = NULL; 5278 delete mResampler; 5279 mResampler = NULL; 5280 5281 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5282 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5283 mChannelCount = (uint16_t)popcount(mChannelMask); 5284 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5285 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5286 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5287 mFrameCount = mInputBytes / mFrameSize; 5288 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5289 5290 if (mSampleRate != mReqSampleRate && mChannelCount <= FCC_2 && mReqChannelCount <= FCC_2) 5291 { 5292 int channelCount; 5293 // optimization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5294 // stereo to mono post process as the resampler always outputs stereo. 5295 if (mChannelCount == 1 && mReqChannelCount == 2) { 5296 channelCount = 1; 5297 } else { 5298 channelCount = 2; 5299 } 5300 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5301 mResampler->setSampleRate(mSampleRate); 5302 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5303 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5304 5305 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5306 if (mChannelCount == 1 && mReqChannelCount == 1) { 5307 mFrameCount >>= 1; 5308 } 5309 5310 } 5311 mRsmpInIndex = mFrameCount; 5312} 5313 5314unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5315{ 5316 Mutex::Autolock _l(mLock); 5317 if (initCheck() != NO_ERROR) { 5318 return 0; 5319 } 5320 5321 return mInput->stream->get_input_frames_lost(mInput->stream); 5322} 5323 5324uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5325{ 5326 Mutex::Autolock _l(mLock); 5327 uint32_t result = 0; 5328 if (getEffectChain_l(sessionId) != 0) { 5329 result = EFFECT_SESSION; 5330 } 5331 5332 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5333 result |= TRACK_SESSION; 5334 } 5335 5336 return result; 5337} 5338 5339AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5340{ 5341 Mutex::Autolock _l(mLock); 5342 return mTrack; 5343} 5344 5345AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5346{ 5347 Mutex::Autolock _l(mLock); 5348 return mInput; 5349} 5350 5351AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5352{ 5353 Mutex::Autolock _l(mLock); 5354 AudioStreamIn *input = mInput; 5355 mInput = NULL; 5356 return input; 5357} 5358 5359// this method must always be called either with ThreadBase mLock held or inside the thread loop 5360audio_stream_t* AudioFlinger::RecordThread::stream() 5361{ 5362 if (mInput == NULL) { 5363 return NULL; 5364 } 5365 return &mInput->stream->common; 5366} 5367 5368 5369// ---------------------------------------------------------------------------- 5370 5371audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5372 uint32_t *pSamplingRate, 5373 audio_format_t *pFormat, 5374 uint32_t *pChannels, 5375 uint32_t *pLatencyMs, 5376 audio_policy_output_flags_t flags) 5377{ 5378 status_t status; 5379 PlaybackThread *thread = NULL; 5380 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5381 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5382 uint32_t channels = pChannels ? *pChannels : 0; 5383 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5384 audio_stream_out_t *outStream; 5385 audio_hw_device_t *outHwDev; 5386 5387 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5388 pDevices ? *pDevices : 0, 5389 samplingRate, 5390 format, 5391 channels, 5392 flags); 5393 5394 if (pDevices == NULL || *pDevices == 0) { 5395 return 0; 5396 } 5397 5398 Mutex::Autolock _l(mLock); 5399 5400 outHwDev = findSuitableHwDev_l(*pDevices); 5401 if (outHwDev == NULL) 5402 return 0; 5403 5404 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5405 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5406 &channels, &samplingRate, &outStream); 5407 mHardwareStatus = AUDIO_HW_IDLE; 5408 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5409 outStream, 5410 samplingRate, 5411 format, 5412 channels, 5413 status); 5414 5415 if (outStream != NULL) { 5416 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5417 audio_io_handle_t id = nextUniqueId(); 5418 5419 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5420 (format != AUDIO_FORMAT_PCM_16_BIT) || 5421 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5422 thread = new DirectOutputThread(this, output, id, *pDevices); 5423 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5424 } else { 5425 thread = new MixerThread(this, output, id, *pDevices); 5426 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5427 } 5428 mPlaybackThreads.add(id, thread); 5429 5430 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5431 if (pFormat != NULL) *pFormat = format; 5432 if (pChannels != NULL) *pChannels = channels; 5433 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5434 5435 // notify client processes of the new output creation 5436 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5437 return id; 5438 } 5439 5440 return 0; 5441} 5442 5443audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5444 audio_io_handle_t output2) 5445{ 5446 Mutex::Autolock _l(mLock); 5447 MixerThread *thread1 = checkMixerThread_l(output1); 5448 MixerThread *thread2 = checkMixerThread_l(output2); 5449 5450 if (thread1 == NULL || thread2 == NULL) { 5451 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5452 return 0; 5453 } 5454 5455 audio_io_handle_t id = nextUniqueId(); 5456 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5457 thread->addOutputTrack(thread2); 5458 mPlaybackThreads.add(id, thread); 5459 // notify client processes of the new output creation 5460 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5461 return id; 5462} 5463 5464status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5465{ 5466 // keep strong reference on the playback thread so that 5467 // it is not destroyed while exit() is executed 5468 sp<PlaybackThread> thread; 5469 { 5470 Mutex::Autolock _l(mLock); 5471 thread = checkPlaybackThread_l(output); 5472 if (thread == NULL) { 5473 return BAD_VALUE; 5474 } 5475 5476 ALOGV("closeOutput() %d", output); 5477 5478 if (thread->type() == ThreadBase::MIXER) { 5479 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5480 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5481 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5482 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5483 } 5484 } 5485 } 5486 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5487 mPlaybackThreads.removeItem(output); 5488 } 5489 thread->exit(); 5490 // The thread entity (active unit of execution) is no longer running here, 5491 // but the ThreadBase container still exists. 5492 5493 if (thread->type() != ThreadBase::DUPLICATING) { 5494 AudioStreamOut *out = thread->clearOutput(); 5495 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 5496 // from now on thread->mOutput is NULL 5497 out->hwDev->close_output_stream(out->hwDev, out->stream); 5498 delete out; 5499 } 5500 return NO_ERROR; 5501} 5502 5503status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5504{ 5505 Mutex::Autolock _l(mLock); 5506 PlaybackThread *thread = checkPlaybackThread_l(output); 5507 5508 if (thread == NULL) { 5509 return BAD_VALUE; 5510 } 5511 5512 ALOGV("suspendOutput() %d", output); 5513 thread->suspend(); 5514 5515 return NO_ERROR; 5516} 5517 5518status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5519{ 5520 Mutex::Autolock _l(mLock); 5521 PlaybackThread *thread = checkPlaybackThread_l(output); 5522 5523 if (thread == NULL) { 5524 return BAD_VALUE; 5525 } 5526 5527 ALOGV("restoreOutput() %d", output); 5528 5529 thread->restore(); 5530 5531 return NO_ERROR; 5532} 5533 5534audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5535 uint32_t *pSamplingRate, 5536 audio_format_t *pFormat, 5537 uint32_t *pChannels, 5538 audio_in_acoustics_t acoustics) 5539{ 5540 status_t status; 5541 RecordThread *thread = NULL; 5542 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5543 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5544 uint32_t channels = pChannels ? *pChannels : 0; 5545 uint32_t reqSamplingRate = samplingRate; 5546 audio_format_t reqFormat = format; 5547 uint32_t reqChannels = channels; 5548 audio_stream_in_t *inStream; 5549 audio_hw_device_t *inHwDev; 5550 5551 if (pDevices == NULL || *pDevices == 0) { 5552 return 0; 5553 } 5554 5555 Mutex::Autolock _l(mLock); 5556 5557 inHwDev = findSuitableHwDev_l(*pDevices); 5558 if (inHwDev == NULL) 5559 return 0; 5560 5561 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5562 &channels, &samplingRate, 5563 acoustics, 5564 &inStream); 5565 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5566 inStream, 5567 samplingRate, 5568 format, 5569 channels, 5570 acoustics, 5571 status); 5572 5573 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5574 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5575 // or stereo to mono conversions on 16 bit PCM inputs. 5576 if (inStream == NULL && status == BAD_VALUE && 5577 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5578 (samplingRate <= 2 * reqSamplingRate) && 5579 (popcount(channels) <= FCC_2) && (popcount(reqChannels) <= FCC_2)) { 5580 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5581 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5582 &channels, &samplingRate, 5583 acoustics, 5584 &inStream); 5585 } 5586 5587 if (inStream != NULL) { 5588 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5589 5590 audio_io_handle_t id = nextUniqueId(); 5591 // Start record thread 5592 // RecorThread require both input and output device indication to forward to audio 5593 // pre processing modules 5594 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5595 thread = new RecordThread(this, 5596 input, 5597 reqSamplingRate, 5598 reqChannels, 5599 id, 5600 device); 5601 mRecordThreads.add(id, thread); 5602 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5603 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5604 if (pFormat != NULL) *pFormat = format; 5605 if (pChannels != NULL) *pChannels = reqChannels; 5606 5607 input->stream->common.standby(&input->stream->common); 5608 5609 // notify client processes of the new input creation 5610 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5611 return id; 5612 } 5613 5614 return 0; 5615} 5616 5617status_t AudioFlinger::closeInput(audio_io_handle_t input) 5618{ 5619 // keep strong reference on the record thread so that 5620 // it is not destroyed while exit() is executed 5621 sp<RecordThread> thread; 5622 { 5623 Mutex::Autolock _l(mLock); 5624 thread = checkRecordThread_l(input); 5625 if (thread == NULL) { 5626 return BAD_VALUE; 5627 } 5628 5629 ALOGV("closeInput() %d", input); 5630 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5631 mRecordThreads.removeItem(input); 5632 } 5633 thread->exit(); 5634 // The thread entity (active unit of execution) is no longer running here, 5635 // but the ThreadBase container still exists. 5636 5637 AudioStreamIn *in = thread->clearInput(); 5638 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 5639 // from now on thread->mInput is NULL 5640 in->hwDev->close_input_stream(in->hwDev, in->stream); 5641 delete in; 5642 5643 return NO_ERROR; 5644} 5645 5646status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5647{ 5648 Mutex::Autolock _l(mLock); 5649 MixerThread *dstThread = checkMixerThread_l(output); 5650 if (dstThread == NULL) { 5651 ALOGW("setStreamOutput() bad output id %d", output); 5652 return BAD_VALUE; 5653 } 5654 5655 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5656 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5657 5658 dstThread->setStreamValid(stream, true); 5659 5660 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5661 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5662 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5663 MixerThread *srcThread = (MixerThread *)thread; 5664 srcThread->setStreamValid(stream, false); 5665 srcThread->invalidateTracks(stream); 5666 } 5667 } 5668 5669 return NO_ERROR; 5670} 5671 5672 5673int AudioFlinger::newAudioSessionId() 5674{ 5675 return nextUniqueId(); 5676} 5677 5678void AudioFlinger::acquireAudioSessionId(int audioSession) 5679{ 5680 Mutex::Autolock _l(mLock); 5681 pid_t caller = IPCThreadState::self()->getCallingPid(); 5682 ALOGV("acquiring %d from %d", audioSession, caller); 5683 size_t num = mAudioSessionRefs.size(); 5684 for (size_t i = 0; i< num; i++) { 5685 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5686 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5687 ref->mCnt++; 5688 ALOGV(" incremented refcount to %d", ref->mCnt); 5689 return; 5690 } 5691 } 5692 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5693 ALOGV(" added new entry for %d", audioSession); 5694} 5695 5696void AudioFlinger::releaseAudioSessionId(int audioSession) 5697{ 5698 Mutex::Autolock _l(mLock); 5699 pid_t caller = IPCThreadState::self()->getCallingPid(); 5700 ALOGV("releasing %d from %d", audioSession, caller); 5701 size_t num = mAudioSessionRefs.size(); 5702 for (size_t i = 0; i< num; i++) { 5703 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5704 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5705 ref->mCnt--; 5706 ALOGV(" decremented refcount to %d", ref->mCnt); 5707 if (ref->mCnt == 0) { 5708 mAudioSessionRefs.removeAt(i); 5709 delete ref; 5710 purgeStaleEffects_l(); 5711 } 5712 return; 5713 } 5714 } 5715 ALOGW("session id %d not found for pid %d", audioSession, caller); 5716} 5717 5718void AudioFlinger::purgeStaleEffects_l() { 5719 5720 ALOGV("purging stale effects"); 5721 5722 Vector< sp<EffectChain> > chains; 5723 5724 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5725 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5726 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5727 sp<EffectChain> ec = t->mEffectChains[j]; 5728 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5729 chains.push(ec); 5730 } 5731 } 5732 } 5733 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5734 sp<RecordThread> t = mRecordThreads.valueAt(i); 5735 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5736 sp<EffectChain> ec = t->mEffectChains[j]; 5737 chains.push(ec); 5738 } 5739 } 5740 5741 for (size_t i = 0; i < chains.size(); i++) { 5742 sp<EffectChain> ec = chains[i]; 5743 int sessionid = ec->sessionId(); 5744 sp<ThreadBase> t = ec->mThread.promote(); 5745 if (t == 0) { 5746 continue; 5747 } 5748 size_t numsessionrefs = mAudioSessionRefs.size(); 5749 bool found = false; 5750 for (size_t k = 0; k < numsessionrefs; k++) { 5751 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5752 if (ref->mSessionid == sessionid) { 5753 ALOGV(" session %d still exists for %d with %d refs", 5754 sessionid, ref->mPid, ref->mCnt); 5755 found = true; 5756 break; 5757 } 5758 } 5759 if (!found) { 5760 // remove all effects from the chain 5761 while (ec->mEffects.size()) { 5762 sp<EffectModule> effect = ec->mEffects[0]; 5763 effect->unPin(); 5764 Mutex::Autolock _l (t->mLock); 5765 t->removeEffect_l(effect); 5766 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5767 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5768 if (handle != 0) { 5769 handle->mEffect.clear(); 5770 if (handle->mHasControl && handle->mEnabled) { 5771 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5772 } 5773 } 5774 } 5775 AudioSystem::unregisterEffect(effect->id()); 5776 } 5777 } 5778 } 5779 return; 5780} 5781 5782// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5783AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5784{ 5785 return mPlaybackThreads.valueFor(output).get(); 5786} 5787 5788// checkMixerThread_l() must be called with AudioFlinger::mLock held 5789AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5790{ 5791 PlaybackThread *thread = checkPlaybackThread_l(output); 5792 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5793} 5794 5795// checkRecordThread_l() must be called with AudioFlinger::mLock held 5796AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5797{ 5798 return mRecordThreads.valueFor(input).get(); 5799} 5800 5801uint32_t AudioFlinger::nextUniqueId() 5802{ 5803 return android_atomic_inc(&mNextUniqueId); 5804} 5805 5806AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5807{ 5808 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5809 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5810 AudioStreamOut *output = thread->getOutput(); 5811 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5812 return thread; 5813 } 5814 } 5815 return NULL; 5816} 5817 5818uint32_t AudioFlinger::primaryOutputDevice_l() const 5819{ 5820 PlaybackThread *thread = primaryPlaybackThread_l(); 5821 5822 if (thread == NULL) { 5823 return 0; 5824 } 5825 5826 return thread->device(); 5827} 5828 5829 5830// ---------------------------------------------------------------------------- 5831// Effect management 5832// ---------------------------------------------------------------------------- 5833 5834 5835status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5836{ 5837 Mutex::Autolock _l(mLock); 5838 return EffectQueryNumberEffects(numEffects); 5839} 5840 5841status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5842{ 5843 Mutex::Autolock _l(mLock); 5844 return EffectQueryEffect(index, descriptor); 5845} 5846 5847status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5848 effect_descriptor_t *descriptor) const 5849{ 5850 Mutex::Autolock _l(mLock); 5851 return EffectGetDescriptor(pUuid, descriptor); 5852} 5853 5854 5855sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5856 effect_descriptor_t *pDesc, 5857 const sp<IEffectClient>& effectClient, 5858 int32_t priority, 5859 audio_io_handle_t io, 5860 int sessionId, 5861 status_t *status, 5862 int *id, 5863 int *enabled) 5864{ 5865 status_t lStatus = NO_ERROR; 5866 sp<EffectHandle> handle; 5867 effect_descriptor_t desc; 5868 5869 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5870 pid, effectClient.get(), priority, sessionId, io); 5871 5872 if (pDesc == NULL) { 5873 lStatus = BAD_VALUE; 5874 goto Exit; 5875 } 5876 5877 // check audio settings permission for global effects 5878 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5879 lStatus = PERMISSION_DENIED; 5880 goto Exit; 5881 } 5882 5883 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5884 // that can only be created by audio policy manager (running in same process) 5885 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5886 lStatus = PERMISSION_DENIED; 5887 goto Exit; 5888 } 5889 5890 if (io == 0) { 5891 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5892 // output must be specified by AudioPolicyManager when using session 5893 // AUDIO_SESSION_OUTPUT_STAGE 5894 lStatus = BAD_VALUE; 5895 goto Exit; 5896 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5897 // if the output returned by getOutputForEffect() is removed before we lock the 5898 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5899 // and we will exit safely 5900 io = AudioSystem::getOutputForEffect(&desc); 5901 } 5902 } 5903 5904 { 5905 Mutex::Autolock _l(mLock); 5906 5907 5908 if (!EffectIsNullUuid(&pDesc->uuid)) { 5909 // if uuid is specified, request effect descriptor 5910 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5911 if (lStatus < 0) { 5912 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5913 goto Exit; 5914 } 5915 } else { 5916 // if uuid is not specified, look for an available implementation 5917 // of the required type in effect factory 5918 if (EffectIsNullUuid(&pDesc->type)) { 5919 ALOGW("createEffect() no effect type"); 5920 lStatus = BAD_VALUE; 5921 goto Exit; 5922 } 5923 uint32_t numEffects = 0; 5924 effect_descriptor_t d; 5925 d.flags = 0; // prevent compiler warning 5926 bool found = false; 5927 5928 lStatus = EffectQueryNumberEffects(&numEffects); 5929 if (lStatus < 0) { 5930 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5931 goto Exit; 5932 } 5933 for (uint32_t i = 0; i < numEffects; i++) { 5934 lStatus = EffectQueryEffect(i, &desc); 5935 if (lStatus < 0) { 5936 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5937 continue; 5938 } 5939 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5940 // If matching type found save effect descriptor. If the session is 5941 // 0 and the effect is not auxiliary, continue enumeration in case 5942 // an auxiliary version of this effect type is available 5943 found = true; 5944 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5945 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5946 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5947 break; 5948 } 5949 } 5950 } 5951 if (!found) { 5952 lStatus = BAD_VALUE; 5953 ALOGW("createEffect() effect not found"); 5954 goto Exit; 5955 } 5956 // For same effect type, chose auxiliary version over insert version if 5957 // connect to output mix (Compliance to OpenSL ES) 5958 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5959 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5960 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5961 } 5962 } 5963 5964 // Do not allow auxiliary effects on a session different from 0 (output mix) 5965 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5966 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5967 lStatus = INVALID_OPERATION; 5968 goto Exit; 5969 } 5970 5971 // check recording permission for visualizer 5972 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5973 !recordingAllowed()) { 5974 lStatus = PERMISSION_DENIED; 5975 goto Exit; 5976 } 5977 5978 // return effect descriptor 5979 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5980 5981 // If output is not specified try to find a matching audio session ID in one of the 5982 // output threads. 5983 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5984 // because of code checking output when entering the function. 5985 // Note: io is never 0 when creating an effect on an input 5986 if (io == 0) { 5987 // look for the thread where the specified audio session is present 5988 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5989 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5990 io = mPlaybackThreads.keyAt(i); 5991 break; 5992 } 5993 } 5994 if (io == 0) { 5995 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5996 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5997 io = mRecordThreads.keyAt(i); 5998 break; 5999 } 6000 } 6001 } 6002 // If no output thread contains the requested session ID, default to 6003 // first output. The effect chain will be moved to the correct output 6004 // thread when a track with the same session ID is created 6005 if (io == 0 && mPlaybackThreads.size()) { 6006 io = mPlaybackThreads.keyAt(0); 6007 } 6008 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6009 } 6010 ThreadBase *thread = checkRecordThread_l(io); 6011 if (thread == NULL) { 6012 thread = checkPlaybackThread_l(io); 6013 if (thread == NULL) { 6014 ALOGE("createEffect() unknown output thread"); 6015 lStatus = BAD_VALUE; 6016 goto Exit; 6017 } 6018 } 6019 6020 sp<Client> client = registerPid_l(pid); 6021 6022 // create effect on selected output thread 6023 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6024 &desc, enabled, &lStatus); 6025 if (handle != 0 && id != NULL) { 6026 *id = handle->id(); 6027 } 6028 } 6029 6030Exit: 6031 if (status != NULL) { 6032 *status = lStatus; 6033 } 6034 return handle; 6035} 6036 6037status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6038 audio_io_handle_t dstOutput) 6039{ 6040 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6041 sessionId, srcOutput, dstOutput); 6042 Mutex::Autolock _l(mLock); 6043 if (srcOutput == dstOutput) { 6044 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6045 return NO_ERROR; 6046 } 6047 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6048 if (srcThread == NULL) { 6049 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6050 return BAD_VALUE; 6051 } 6052 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6053 if (dstThread == NULL) { 6054 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6055 return BAD_VALUE; 6056 } 6057 6058 Mutex::Autolock _dl(dstThread->mLock); 6059 Mutex::Autolock _sl(srcThread->mLock); 6060 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6061 6062 return NO_ERROR; 6063} 6064 6065// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6066status_t AudioFlinger::moveEffectChain_l(int sessionId, 6067 AudioFlinger::PlaybackThread *srcThread, 6068 AudioFlinger::PlaybackThread *dstThread, 6069 bool reRegister) 6070{ 6071 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6072 sessionId, srcThread, dstThread); 6073 6074 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6075 if (chain == 0) { 6076 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6077 sessionId, srcThread); 6078 return INVALID_OPERATION; 6079 } 6080 6081 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6082 // so that a new chain is created with correct parameters when first effect is added. This is 6083 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6084 // removed. 6085 srcThread->removeEffectChain_l(chain); 6086 6087 // transfer all effects one by one so that new effect chain is created on new thread with 6088 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6089 audio_io_handle_t dstOutput = dstThread->id(); 6090 sp<EffectChain> dstChain; 6091 uint32_t strategy = 0; // prevent compiler warning 6092 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6093 while (effect != 0) { 6094 srcThread->removeEffect_l(effect); 6095 dstThread->addEffect_l(effect); 6096 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6097 if (effect->state() == EffectModule::ACTIVE || 6098 effect->state() == EffectModule::STOPPING) { 6099 effect->start(); 6100 } 6101 // if the move request is not received from audio policy manager, the effect must be 6102 // re-registered with the new strategy and output 6103 if (dstChain == 0) { 6104 dstChain = effect->chain().promote(); 6105 if (dstChain == 0) { 6106 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6107 srcThread->addEffect_l(effect); 6108 return NO_INIT; 6109 } 6110 strategy = dstChain->strategy(); 6111 } 6112 if (reRegister) { 6113 AudioSystem::unregisterEffect(effect->id()); 6114 AudioSystem::registerEffect(&effect->desc(), 6115 dstOutput, 6116 strategy, 6117 sessionId, 6118 effect->id()); 6119 } 6120 effect = chain->getEffectFromId_l(0); 6121 } 6122 6123 return NO_ERROR; 6124} 6125 6126 6127// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6128sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6129 const sp<AudioFlinger::Client>& client, 6130 const sp<IEffectClient>& effectClient, 6131 int32_t priority, 6132 int sessionId, 6133 effect_descriptor_t *desc, 6134 int *enabled, 6135 status_t *status 6136 ) 6137{ 6138 sp<EffectModule> effect; 6139 sp<EffectHandle> handle; 6140 status_t lStatus; 6141 sp<EffectChain> chain; 6142 bool chainCreated = false; 6143 bool effectCreated = false; 6144 bool effectRegistered = false; 6145 6146 lStatus = initCheck(); 6147 if (lStatus != NO_ERROR) { 6148 ALOGW("createEffect_l() Audio driver not initialized."); 6149 goto Exit; 6150 } 6151 6152 // Do not allow effects with session ID 0 on direct output or duplicating threads 6153 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6154 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6155 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6156 desc->name, sessionId); 6157 lStatus = BAD_VALUE; 6158 goto Exit; 6159 } 6160 // Only Pre processor effects are allowed on input threads and only on input threads 6161 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6162 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6163 desc->name, desc->flags, mType); 6164 lStatus = BAD_VALUE; 6165 goto Exit; 6166 } 6167 6168 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6169 6170 { // scope for mLock 6171 Mutex::Autolock _l(mLock); 6172 6173 // check for existing effect chain with the requested audio session 6174 chain = getEffectChain_l(sessionId); 6175 if (chain == 0) { 6176 // create a new chain for this session 6177 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6178 chain = new EffectChain(this, sessionId); 6179 addEffectChain_l(chain); 6180 chain->setStrategy(getStrategyForSession_l(sessionId)); 6181 chainCreated = true; 6182 } else { 6183 effect = chain->getEffectFromDesc_l(desc); 6184 } 6185 6186 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6187 6188 if (effect == 0) { 6189 int id = mAudioFlinger->nextUniqueId(); 6190 // Check CPU and memory usage 6191 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6192 if (lStatus != NO_ERROR) { 6193 goto Exit; 6194 } 6195 effectRegistered = true; 6196 // create a new effect module if none present in the chain 6197 effect = new EffectModule(this, chain, desc, id, sessionId); 6198 lStatus = effect->status(); 6199 if (lStatus != NO_ERROR) { 6200 goto Exit; 6201 } 6202 lStatus = chain->addEffect_l(effect); 6203 if (lStatus != NO_ERROR) { 6204 goto Exit; 6205 } 6206 effectCreated = true; 6207 6208 effect->setDevice(mDevice); 6209 effect->setMode(mAudioFlinger->getMode()); 6210 } 6211 // create effect handle and connect it to effect module 6212 handle = new EffectHandle(effect, client, effectClient, priority); 6213 lStatus = effect->addHandle(handle); 6214 if (enabled != NULL) { 6215 *enabled = (int)effect->isEnabled(); 6216 } 6217 } 6218 6219Exit: 6220 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6221 Mutex::Autolock _l(mLock); 6222 if (effectCreated) { 6223 chain->removeEffect_l(effect); 6224 } 6225 if (effectRegistered) { 6226 AudioSystem::unregisterEffect(effect->id()); 6227 } 6228 if (chainCreated) { 6229 removeEffectChain_l(chain); 6230 } 6231 handle.clear(); 6232 } 6233 6234 if (status != NULL) { 6235 *status = lStatus; 6236 } 6237 return handle; 6238} 6239 6240sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6241{ 6242 sp<EffectChain> chain = getEffectChain_l(sessionId); 6243 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6244} 6245 6246// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6247// PlaybackThread::mLock held 6248status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6249{ 6250 // check for existing effect chain with the requested audio session 6251 int sessionId = effect->sessionId(); 6252 sp<EffectChain> chain = getEffectChain_l(sessionId); 6253 bool chainCreated = false; 6254 6255 if (chain == 0) { 6256 // create a new chain for this session 6257 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6258 chain = new EffectChain(this, sessionId); 6259 addEffectChain_l(chain); 6260 chain->setStrategy(getStrategyForSession_l(sessionId)); 6261 chainCreated = true; 6262 } 6263 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6264 6265 if (chain->getEffectFromId_l(effect->id()) != 0) { 6266 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6267 this, effect->desc().name, chain.get()); 6268 return BAD_VALUE; 6269 } 6270 6271 status_t status = chain->addEffect_l(effect); 6272 if (status != NO_ERROR) { 6273 if (chainCreated) { 6274 removeEffectChain_l(chain); 6275 } 6276 return status; 6277 } 6278 6279 effect->setDevice(mDevice); 6280 effect->setMode(mAudioFlinger->getMode()); 6281 return NO_ERROR; 6282} 6283 6284void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6285 6286 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6287 effect_descriptor_t desc = effect->desc(); 6288 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6289 detachAuxEffect_l(effect->id()); 6290 } 6291 6292 sp<EffectChain> chain = effect->chain().promote(); 6293 if (chain != 0) { 6294 // remove effect chain if removing last effect 6295 if (chain->removeEffect_l(effect) == 0) { 6296 removeEffectChain_l(chain); 6297 } 6298 } else { 6299 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6300 } 6301} 6302 6303void AudioFlinger::ThreadBase::lockEffectChains_l( 6304 Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6305{ 6306 effectChains = mEffectChains; 6307 for (size_t i = 0; i < mEffectChains.size(); i++) { 6308 mEffectChains[i]->lock(); 6309 } 6310} 6311 6312void AudioFlinger::ThreadBase::unlockEffectChains( 6313 const Vector< sp<AudioFlinger::EffectChain> >& effectChains) 6314{ 6315 for (size_t i = 0; i < effectChains.size(); i++) { 6316 effectChains[i]->unlock(); 6317 } 6318} 6319 6320sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6321{ 6322 Mutex::Autolock _l(mLock); 6323 return getEffectChain_l(sessionId); 6324} 6325 6326sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6327{ 6328 size_t size = mEffectChains.size(); 6329 for (size_t i = 0; i < size; i++) { 6330 if (mEffectChains[i]->sessionId() == sessionId) { 6331 return mEffectChains[i]; 6332 } 6333 } 6334 return 0; 6335} 6336 6337void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6338{ 6339 Mutex::Autolock _l(mLock); 6340 size_t size = mEffectChains.size(); 6341 for (size_t i = 0; i < size; i++) { 6342 mEffectChains[i]->setMode_l(mode); 6343 } 6344} 6345 6346void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6347 const wp<EffectHandle>& handle, 6348 bool unpinIfLast) { 6349 6350 Mutex::Autolock _l(mLock); 6351 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6352 // delete the effect module if removing last handle on it 6353 if (effect->removeHandle(handle) == 0) { 6354 if (!effect->isPinned() || unpinIfLast) { 6355 removeEffect_l(effect); 6356 AudioSystem::unregisterEffect(effect->id()); 6357 } 6358 } 6359} 6360 6361status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6362{ 6363 int session = chain->sessionId(); 6364 int16_t *buffer = mMixBuffer; 6365 bool ownsBuffer = false; 6366 6367 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6368 if (session > 0) { 6369 // Only one effect chain can be present in direct output thread and it uses 6370 // the mix buffer as input 6371 if (mType != DIRECT) { 6372 size_t numSamples = mFrameCount * mChannelCount; 6373 buffer = new int16_t[numSamples]; 6374 memset(buffer, 0, numSamples * sizeof(int16_t)); 6375 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6376 ownsBuffer = true; 6377 } 6378 6379 // Attach all tracks with same session ID to this chain. 6380 for (size_t i = 0; i < mTracks.size(); ++i) { 6381 sp<Track> track = mTracks[i]; 6382 if (session == track->sessionId()) { 6383 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6384 track->setMainBuffer(buffer); 6385 chain->incTrackCnt(); 6386 } 6387 } 6388 6389 // indicate all active tracks in the chain 6390 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6391 sp<Track> track = mActiveTracks[i].promote(); 6392 if (track == 0) continue; 6393 if (session == track->sessionId()) { 6394 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6395 chain->incActiveTrackCnt(); 6396 } 6397 } 6398 } 6399 6400 chain->setInBuffer(buffer, ownsBuffer); 6401 chain->setOutBuffer(mMixBuffer); 6402 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6403 // chains list in order to be processed last as it contains output stage effects 6404 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6405 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6406 // after track specific effects and before output stage 6407 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6408 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6409 // Effect chain for other sessions are inserted at beginning of effect 6410 // chains list to be processed before output mix effects. Relative order between other 6411 // sessions is not important 6412 size_t size = mEffectChains.size(); 6413 size_t i = 0; 6414 for (i = 0; i < size; i++) { 6415 if (mEffectChains[i]->sessionId() < session) break; 6416 } 6417 mEffectChains.insertAt(chain, i); 6418 checkSuspendOnAddEffectChain_l(chain); 6419 6420 return NO_ERROR; 6421} 6422 6423size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6424{ 6425 int session = chain->sessionId(); 6426 6427 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6428 6429 for (size_t i = 0; i < mEffectChains.size(); i++) { 6430 if (chain == mEffectChains[i]) { 6431 mEffectChains.removeAt(i); 6432 // detach all active tracks from the chain 6433 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6434 sp<Track> track = mActiveTracks[i].promote(); 6435 if (track == 0) continue; 6436 if (session == track->sessionId()) { 6437 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6438 chain.get(), session); 6439 chain->decActiveTrackCnt(); 6440 } 6441 } 6442 6443 // detach all tracks with same session ID from this chain 6444 for (size_t i = 0; i < mTracks.size(); ++i) { 6445 sp<Track> track = mTracks[i]; 6446 if (session == track->sessionId()) { 6447 track->setMainBuffer(mMixBuffer); 6448 chain->decTrackCnt(); 6449 } 6450 } 6451 break; 6452 } 6453 } 6454 return mEffectChains.size(); 6455} 6456 6457status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6458 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6459{ 6460 Mutex::Autolock _l(mLock); 6461 return attachAuxEffect_l(track, EffectId); 6462} 6463 6464status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6465 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6466{ 6467 status_t status = NO_ERROR; 6468 6469 if (EffectId == 0) { 6470 track->setAuxBuffer(0, NULL); 6471 } else { 6472 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6473 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6474 if (effect != 0) { 6475 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6476 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6477 } else { 6478 status = INVALID_OPERATION; 6479 } 6480 } else { 6481 status = BAD_VALUE; 6482 } 6483 } 6484 return status; 6485} 6486 6487void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6488{ 6489 for (size_t i = 0; i < mTracks.size(); ++i) { 6490 sp<Track> track = mTracks[i]; 6491 if (track->auxEffectId() == effectId) { 6492 attachAuxEffect_l(track, 0); 6493 } 6494 } 6495} 6496 6497status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6498{ 6499 // only one chain per input thread 6500 if (mEffectChains.size() != 0) { 6501 return INVALID_OPERATION; 6502 } 6503 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6504 6505 chain->setInBuffer(NULL); 6506 chain->setOutBuffer(NULL); 6507 6508 checkSuspendOnAddEffectChain_l(chain); 6509 6510 mEffectChains.add(chain); 6511 6512 return NO_ERROR; 6513} 6514 6515size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6516{ 6517 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6518 ALOGW_IF(mEffectChains.size() != 1, 6519 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6520 chain.get(), mEffectChains.size(), this); 6521 if (mEffectChains.size() == 1) { 6522 mEffectChains.removeAt(0); 6523 } 6524 return 0; 6525} 6526 6527// ---------------------------------------------------------------------------- 6528// EffectModule implementation 6529// ---------------------------------------------------------------------------- 6530 6531#undef LOG_TAG 6532#define LOG_TAG "AudioFlinger::EffectModule" 6533 6534AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6535 const wp<AudioFlinger::EffectChain>& chain, 6536 effect_descriptor_t *desc, 6537 int id, 6538 int sessionId) 6539 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6540 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6541{ 6542 ALOGV("Constructor %p", this); 6543 int lStatus; 6544 if (thread == NULL) { 6545 return; 6546 } 6547 6548 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6549 6550 // create effect engine from effect factory 6551 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6552 6553 if (mStatus != NO_ERROR) { 6554 return; 6555 } 6556 lStatus = init(); 6557 if (lStatus < 0) { 6558 mStatus = lStatus; 6559 goto Error; 6560 } 6561 6562 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6563 mPinned = true; 6564 } 6565 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6566 return; 6567Error: 6568 EffectRelease(mEffectInterface); 6569 mEffectInterface = NULL; 6570 ALOGV("Constructor Error %d", mStatus); 6571} 6572 6573AudioFlinger::EffectModule::~EffectModule() 6574{ 6575 ALOGV("Destructor %p", this); 6576 if (mEffectInterface != NULL) { 6577 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6578 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6579 sp<ThreadBase> thread = mThread.promote(); 6580 if (thread != 0) { 6581 audio_stream_t *stream = thread->stream(); 6582 if (stream != NULL) { 6583 stream->remove_audio_effect(stream, mEffectInterface); 6584 } 6585 } 6586 } 6587 // release effect engine 6588 EffectRelease(mEffectInterface); 6589 } 6590} 6591 6592status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6593{ 6594 status_t status; 6595 6596 Mutex::Autolock _l(mLock); 6597 int priority = handle->priority(); 6598 size_t size = mHandles.size(); 6599 sp<EffectHandle> h; 6600 size_t i; 6601 for (i = 0; i < size; i++) { 6602 h = mHandles[i].promote(); 6603 if (h == 0) continue; 6604 if (h->priority() <= priority) break; 6605 } 6606 // if inserted in first place, move effect control from previous owner to this handle 6607 if (i == 0) { 6608 bool enabled = false; 6609 if (h != 0) { 6610 enabled = h->enabled(); 6611 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6612 } 6613 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6614 status = NO_ERROR; 6615 } else { 6616 status = ALREADY_EXISTS; 6617 } 6618 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6619 mHandles.insertAt(handle, i); 6620 return status; 6621} 6622 6623size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6624{ 6625 Mutex::Autolock _l(mLock); 6626 size_t size = mHandles.size(); 6627 size_t i; 6628 for (i = 0; i < size; i++) { 6629 if (mHandles[i] == handle) break; 6630 } 6631 if (i == size) { 6632 return size; 6633 } 6634 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6635 6636 bool enabled = false; 6637 EffectHandle *hdl = handle.unsafe_get(); 6638 if (hdl != NULL) { 6639 ALOGV("removeHandle() unsafe_get OK"); 6640 enabled = hdl->enabled(); 6641 } 6642 mHandles.removeAt(i); 6643 size = mHandles.size(); 6644 // if removed from first place, move effect control from this handle to next in line 6645 if (i == 0 && size != 0) { 6646 sp<EffectHandle> h = mHandles[0].promote(); 6647 if (h != 0) { 6648 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6649 } 6650 } 6651 6652 // Prevent calls to process() and other functions on effect interface from now on. 6653 // The effect engine will be released by the destructor when the last strong reference on 6654 // this object is released which can happen after next process is called. 6655 if (size == 0 && !mPinned) { 6656 mState = DESTROYED; 6657 } 6658 6659 return size; 6660} 6661 6662sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6663{ 6664 Mutex::Autolock _l(mLock); 6665 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6666} 6667 6668void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6669{ 6670 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6671 // keep a strong reference on this EffectModule to avoid calling the 6672 // destructor before we exit 6673 sp<EffectModule> keep(this); 6674 { 6675 sp<ThreadBase> thread = mThread.promote(); 6676 if (thread != 0) { 6677 thread->disconnectEffect(keep, handle, unpinIfLast); 6678 } 6679 } 6680} 6681 6682void AudioFlinger::EffectModule::updateState() { 6683 Mutex::Autolock _l(mLock); 6684 6685 switch (mState) { 6686 case RESTART: 6687 reset_l(); 6688 // FALL THROUGH 6689 6690 case STARTING: 6691 // clear auxiliary effect input buffer for next accumulation 6692 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6693 memset(mConfig.inputCfg.buffer.raw, 6694 0, 6695 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6696 } 6697 start_l(); 6698 mState = ACTIVE; 6699 break; 6700 case STOPPING: 6701 stop_l(); 6702 mDisableWaitCnt = mMaxDisableWaitCnt; 6703 mState = STOPPED; 6704 break; 6705 case STOPPED: 6706 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6707 // turn off sequence. 6708 if (--mDisableWaitCnt == 0) { 6709 reset_l(); 6710 mState = IDLE; 6711 } 6712 break; 6713 default: //IDLE , ACTIVE, DESTROYED 6714 break; 6715 } 6716} 6717 6718void AudioFlinger::EffectModule::process() 6719{ 6720 Mutex::Autolock _l(mLock); 6721 6722 if (mState == DESTROYED || mEffectInterface == NULL || 6723 mConfig.inputCfg.buffer.raw == NULL || 6724 mConfig.outputCfg.buffer.raw == NULL) { 6725 return; 6726 } 6727 6728 if (isProcessEnabled()) { 6729 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6730 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6731 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6732 mConfig.inputCfg.buffer.s32, 6733 mConfig.inputCfg.buffer.frameCount/2); 6734 } 6735 6736 // do the actual processing in the effect engine 6737 int ret = (*mEffectInterface)->process(mEffectInterface, 6738 &mConfig.inputCfg.buffer, 6739 &mConfig.outputCfg.buffer); 6740 6741 // force transition to IDLE state when engine is ready 6742 if (mState == STOPPED && ret == -ENODATA) { 6743 mDisableWaitCnt = 1; 6744 } 6745 6746 // clear auxiliary effect input buffer for next accumulation 6747 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6748 memset(mConfig.inputCfg.buffer.raw, 0, 6749 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6750 } 6751 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6752 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6753 // If an insert effect is idle and input buffer is different from output buffer, 6754 // accumulate input onto output 6755 sp<EffectChain> chain = mChain.promote(); 6756 if (chain != 0 && chain->activeTrackCnt() != 0) { 6757 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6758 int16_t *in = mConfig.inputCfg.buffer.s16; 6759 int16_t *out = mConfig.outputCfg.buffer.s16; 6760 for (size_t i = 0; i < frameCnt; i++) { 6761 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6762 } 6763 } 6764 } 6765} 6766 6767void AudioFlinger::EffectModule::reset_l() 6768{ 6769 if (mEffectInterface == NULL) { 6770 return; 6771 } 6772 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6773} 6774 6775status_t AudioFlinger::EffectModule::configure() 6776{ 6777 uint32_t channels; 6778 if (mEffectInterface == NULL) { 6779 return NO_INIT; 6780 } 6781 6782 sp<ThreadBase> thread = mThread.promote(); 6783 if (thread == 0) { 6784 return DEAD_OBJECT; 6785 } 6786 6787 // TODO: handle configuration of effects replacing track process 6788 if (thread->channelCount() == 1) { 6789 channels = AUDIO_CHANNEL_OUT_MONO; 6790 } else { 6791 channels = AUDIO_CHANNEL_OUT_STEREO; 6792 } 6793 6794 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6795 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6796 } else { 6797 mConfig.inputCfg.channels = channels; 6798 } 6799 mConfig.outputCfg.channels = channels; 6800 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6801 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6802 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6803 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6804 mConfig.inputCfg.bufferProvider.cookie = NULL; 6805 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6806 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6807 mConfig.outputCfg.bufferProvider.cookie = NULL; 6808 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6809 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6810 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6811 // Insert effect: 6812 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6813 // always overwrites output buffer: input buffer == output buffer 6814 // - in other sessions: 6815 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6816 // other effect: overwrites output buffer: input buffer == output buffer 6817 // Auxiliary effect: 6818 // accumulates in output buffer: input buffer != output buffer 6819 // Therefore: accumulate <=> input buffer != output buffer 6820 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6821 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6822 } else { 6823 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6824 } 6825 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6826 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6827 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6828 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6829 6830 ALOGV("configure() %p thread %p buffer %p framecount %d", 6831 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6832 6833 status_t cmdStatus; 6834 uint32_t size = sizeof(int); 6835 status_t status = (*mEffectInterface)->command(mEffectInterface, 6836 EFFECT_CMD_SET_CONFIG, 6837 sizeof(effect_config_t), 6838 &mConfig, 6839 &size, 6840 &cmdStatus); 6841 if (status == 0) { 6842 status = cmdStatus; 6843 } 6844 6845 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6846 (1000 * mConfig.outputCfg.buffer.frameCount); 6847 6848 return status; 6849} 6850 6851status_t AudioFlinger::EffectModule::init() 6852{ 6853 Mutex::Autolock _l(mLock); 6854 if (mEffectInterface == NULL) { 6855 return NO_INIT; 6856 } 6857 status_t cmdStatus; 6858 uint32_t size = sizeof(status_t); 6859 status_t status = (*mEffectInterface)->command(mEffectInterface, 6860 EFFECT_CMD_INIT, 6861 0, 6862 NULL, 6863 &size, 6864 &cmdStatus); 6865 if (status == 0) { 6866 status = cmdStatus; 6867 } 6868 return status; 6869} 6870 6871status_t AudioFlinger::EffectModule::start() 6872{ 6873 Mutex::Autolock _l(mLock); 6874 return start_l(); 6875} 6876 6877status_t AudioFlinger::EffectModule::start_l() 6878{ 6879 if (mEffectInterface == NULL) { 6880 return NO_INIT; 6881 } 6882 status_t cmdStatus; 6883 uint32_t size = sizeof(status_t); 6884 status_t status = (*mEffectInterface)->command(mEffectInterface, 6885 EFFECT_CMD_ENABLE, 6886 0, 6887 NULL, 6888 &size, 6889 &cmdStatus); 6890 if (status == 0) { 6891 status = cmdStatus; 6892 } 6893 if (status == 0 && 6894 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6895 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6896 sp<ThreadBase> thread = mThread.promote(); 6897 if (thread != 0) { 6898 audio_stream_t *stream = thread->stream(); 6899 if (stream != NULL) { 6900 stream->add_audio_effect(stream, mEffectInterface); 6901 } 6902 } 6903 } 6904 return status; 6905} 6906 6907status_t AudioFlinger::EffectModule::stop() 6908{ 6909 Mutex::Autolock _l(mLock); 6910 return stop_l(); 6911} 6912 6913status_t AudioFlinger::EffectModule::stop_l() 6914{ 6915 if (mEffectInterface == NULL) { 6916 return NO_INIT; 6917 } 6918 status_t cmdStatus; 6919 uint32_t size = sizeof(status_t); 6920 status_t status = (*mEffectInterface)->command(mEffectInterface, 6921 EFFECT_CMD_DISABLE, 6922 0, 6923 NULL, 6924 &size, 6925 &cmdStatus); 6926 if (status == 0) { 6927 status = cmdStatus; 6928 } 6929 if (status == 0 && 6930 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6931 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6932 sp<ThreadBase> thread = mThread.promote(); 6933 if (thread != 0) { 6934 audio_stream_t *stream = thread->stream(); 6935 if (stream != NULL) { 6936 stream->remove_audio_effect(stream, mEffectInterface); 6937 } 6938 } 6939 } 6940 return status; 6941} 6942 6943status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6944 uint32_t cmdSize, 6945 void *pCmdData, 6946 uint32_t *replySize, 6947 void *pReplyData) 6948{ 6949 Mutex::Autolock _l(mLock); 6950// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6951 6952 if (mState == DESTROYED || mEffectInterface == NULL) { 6953 return NO_INIT; 6954 } 6955 status_t status = (*mEffectInterface)->command(mEffectInterface, 6956 cmdCode, 6957 cmdSize, 6958 pCmdData, 6959 replySize, 6960 pReplyData); 6961 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6962 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6963 for (size_t i = 1; i < mHandles.size(); i++) { 6964 sp<EffectHandle> h = mHandles[i].promote(); 6965 if (h != 0) { 6966 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6967 } 6968 } 6969 } 6970 return status; 6971} 6972 6973status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6974{ 6975 6976 Mutex::Autolock _l(mLock); 6977 ALOGV("setEnabled %p enabled %d", this, enabled); 6978 6979 if (enabled != isEnabled()) { 6980 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6981 if (enabled && status != NO_ERROR) { 6982 return status; 6983 } 6984 6985 switch (mState) { 6986 // going from disabled to enabled 6987 case IDLE: 6988 mState = STARTING; 6989 break; 6990 case STOPPED: 6991 mState = RESTART; 6992 break; 6993 case STOPPING: 6994 mState = ACTIVE; 6995 break; 6996 6997 // going from enabled to disabled 6998 case RESTART: 6999 mState = STOPPED; 7000 break; 7001 case STARTING: 7002 mState = IDLE; 7003 break; 7004 case ACTIVE: 7005 mState = STOPPING; 7006 break; 7007 case DESTROYED: 7008 return NO_ERROR; // simply ignore as we are being destroyed 7009 } 7010 for (size_t i = 1; i < mHandles.size(); i++) { 7011 sp<EffectHandle> h = mHandles[i].promote(); 7012 if (h != 0) { 7013 h->setEnabled(enabled); 7014 } 7015 } 7016 } 7017 return NO_ERROR; 7018} 7019 7020bool AudioFlinger::EffectModule::isEnabled() const 7021{ 7022 switch (mState) { 7023 case RESTART: 7024 case STARTING: 7025 case ACTIVE: 7026 return true; 7027 case IDLE: 7028 case STOPPING: 7029 case STOPPED: 7030 case DESTROYED: 7031 default: 7032 return false; 7033 } 7034} 7035 7036bool AudioFlinger::EffectModule::isProcessEnabled() const 7037{ 7038 switch (mState) { 7039 case RESTART: 7040 case ACTIVE: 7041 case STOPPING: 7042 case STOPPED: 7043 return true; 7044 case IDLE: 7045 case STARTING: 7046 case DESTROYED: 7047 default: 7048 return false; 7049 } 7050} 7051 7052status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7053{ 7054 Mutex::Autolock _l(mLock); 7055 status_t status = NO_ERROR; 7056 7057 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7058 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7059 if (isProcessEnabled() && 7060 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7061 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7062 status_t cmdStatus; 7063 uint32_t volume[2]; 7064 uint32_t *pVolume = NULL; 7065 uint32_t size = sizeof(volume); 7066 volume[0] = *left; 7067 volume[1] = *right; 7068 if (controller) { 7069 pVolume = volume; 7070 } 7071 status = (*mEffectInterface)->command(mEffectInterface, 7072 EFFECT_CMD_SET_VOLUME, 7073 size, 7074 volume, 7075 &size, 7076 pVolume); 7077 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7078 *left = volume[0]; 7079 *right = volume[1]; 7080 } 7081 } 7082 return status; 7083} 7084 7085status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7086{ 7087 Mutex::Autolock _l(mLock); 7088 status_t status = NO_ERROR; 7089 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7090 // audio pre processing modules on RecordThread can receive both output and 7091 // input device indication in the same call 7092 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7093 if (dev) { 7094 status_t cmdStatus; 7095 uint32_t size = sizeof(status_t); 7096 7097 status = (*mEffectInterface)->command(mEffectInterface, 7098 EFFECT_CMD_SET_DEVICE, 7099 sizeof(uint32_t), 7100 &dev, 7101 &size, 7102 &cmdStatus); 7103 if (status == NO_ERROR) { 7104 status = cmdStatus; 7105 } 7106 } 7107 dev = device & AUDIO_DEVICE_IN_ALL; 7108 if (dev) { 7109 status_t cmdStatus; 7110 uint32_t size = sizeof(status_t); 7111 7112 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7113 EFFECT_CMD_SET_INPUT_DEVICE, 7114 sizeof(uint32_t), 7115 &dev, 7116 &size, 7117 &cmdStatus); 7118 if (status2 == NO_ERROR) { 7119 status2 = cmdStatus; 7120 } 7121 if (status == NO_ERROR) { 7122 status = status2; 7123 } 7124 } 7125 } 7126 return status; 7127} 7128 7129status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7130{ 7131 Mutex::Autolock _l(mLock); 7132 status_t status = NO_ERROR; 7133 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7134 status_t cmdStatus; 7135 uint32_t size = sizeof(status_t); 7136 status = (*mEffectInterface)->command(mEffectInterface, 7137 EFFECT_CMD_SET_AUDIO_MODE, 7138 sizeof(audio_mode_t), 7139 &mode, 7140 &size, 7141 &cmdStatus); 7142 if (status == NO_ERROR) { 7143 status = cmdStatus; 7144 } 7145 } 7146 return status; 7147} 7148 7149void AudioFlinger::EffectModule::setSuspended(bool suspended) 7150{ 7151 Mutex::Autolock _l(mLock); 7152 mSuspended = suspended; 7153} 7154 7155bool AudioFlinger::EffectModule::suspended() const 7156{ 7157 Mutex::Autolock _l(mLock); 7158 return mSuspended; 7159} 7160 7161status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7162{ 7163 const size_t SIZE = 256; 7164 char buffer[SIZE]; 7165 String8 result; 7166 7167 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7168 result.append(buffer); 7169 7170 bool locked = tryLock(mLock); 7171 // failed to lock - AudioFlinger is probably deadlocked 7172 if (!locked) { 7173 result.append("\t\tCould not lock Fx mutex:\n"); 7174 } 7175 7176 result.append("\t\tSession Status State Engine:\n"); 7177 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7178 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7179 result.append(buffer); 7180 7181 result.append("\t\tDescriptor:\n"); 7182 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7183 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7184 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7185 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7186 result.append(buffer); 7187 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7188 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7189 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7190 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7191 result.append(buffer); 7192 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7193 mDescriptor.apiVersion, 7194 mDescriptor.flags); 7195 result.append(buffer); 7196 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7197 mDescriptor.name); 7198 result.append(buffer); 7199 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7200 mDescriptor.implementor); 7201 result.append(buffer); 7202 7203 result.append("\t\t- Input configuration:\n"); 7204 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7205 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7206 (uint32_t)mConfig.inputCfg.buffer.raw, 7207 mConfig.inputCfg.buffer.frameCount, 7208 mConfig.inputCfg.samplingRate, 7209 mConfig.inputCfg.channels, 7210 mConfig.inputCfg.format); 7211 result.append(buffer); 7212 7213 result.append("\t\t- Output configuration:\n"); 7214 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7215 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7216 (uint32_t)mConfig.outputCfg.buffer.raw, 7217 mConfig.outputCfg.buffer.frameCount, 7218 mConfig.outputCfg.samplingRate, 7219 mConfig.outputCfg.channels, 7220 mConfig.outputCfg.format); 7221 result.append(buffer); 7222 7223 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7224 result.append(buffer); 7225 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7226 for (size_t i = 0; i < mHandles.size(); ++i) { 7227 sp<EffectHandle> handle = mHandles[i].promote(); 7228 if (handle != 0) { 7229 handle->dump(buffer, SIZE); 7230 result.append(buffer); 7231 } 7232 } 7233 7234 result.append("\n"); 7235 7236 write(fd, result.string(), result.length()); 7237 7238 if (locked) { 7239 mLock.unlock(); 7240 } 7241 7242 return NO_ERROR; 7243} 7244 7245// ---------------------------------------------------------------------------- 7246// EffectHandle implementation 7247// ---------------------------------------------------------------------------- 7248 7249#undef LOG_TAG 7250#define LOG_TAG "AudioFlinger::EffectHandle" 7251 7252AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7253 const sp<AudioFlinger::Client>& client, 7254 const sp<IEffectClient>& effectClient, 7255 int32_t priority) 7256 : BnEffect(), 7257 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7258 mPriority(priority), mHasControl(false), mEnabled(false) 7259{ 7260 ALOGV("constructor %p", this); 7261 7262 if (client == 0) { 7263 return; 7264 } 7265 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7266 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7267 if (mCblkMemory != 0) { 7268 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7269 7270 if (mCblk != NULL) { 7271 new(mCblk) effect_param_cblk_t(); 7272 mBuffer = (uint8_t *)mCblk + bufOffset; 7273 } 7274 } else { 7275 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7276 return; 7277 } 7278} 7279 7280AudioFlinger::EffectHandle::~EffectHandle() 7281{ 7282 ALOGV("Destructor %p", this); 7283 disconnect(false); 7284 ALOGV("Destructor DONE %p", this); 7285} 7286 7287status_t AudioFlinger::EffectHandle::enable() 7288{ 7289 ALOGV("enable %p", this); 7290 if (!mHasControl) return INVALID_OPERATION; 7291 if (mEffect == 0) return DEAD_OBJECT; 7292 7293 if (mEnabled) { 7294 return NO_ERROR; 7295 } 7296 7297 mEnabled = true; 7298 7299 sp<ThreadBase> thread = mEffect->thread().promote(); 7300 if (thread != 0) { 7301 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7302 } 7303 7304 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7305 if (mEffect->suspended()) { 7306 return NO_ERROR; 7307 } 7308 7309 status_t status = mEffect->setEnabled(true); 7310 if (status != NO_ERROR) { 7311 if (thread != 0) { 7312 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7313 } 7314 mEnabled = false; 7315 } 7316 return status; 7317} 7318 7319status_t AudioFlinger::EffectHandle::disable() 7320{ 7321 ALOGV("disable %p", this); 7322 if (!mHasControl) return INVALID_OPERATION; 7323 if (mEffect == 0) return DEAD_OBJECT; 7324 7325 if (!mEnabled) { 7326 return NO_ERROR; 7327 } 7328 mEnabled = false; 7329 7330 if (mEffect->suspended()) { 7331 return NO_ERROR; 7332 } 7333 7334 status_t status = mEffect->setEnabled(false); 7335 7336 sp<ThreadBase> thread = mEffect->thread().promote(); 7337 if (thread != 0) { 7338 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7339 } 7340 7341 return status; 7342} 7343 7344void AudioFlinger::EffectHandle::disconnect() 7345{ 7346 disconnect(true); 7347} 7348 7349void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7350{ 7351 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7352 if (mEffect == 0) { 7353 return; 7354 } 7355 mEffect->disconnect(this, unpinIfLast); 7356 7357 if (mHasControl && mEnabled) { 7358 sp<ThreadBase> thread = mEffect->thread().promote(); 7359 if (thread != 0) { 7360 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7361 } 7362 } 7363 7364 // release sp on module => module destructor can be called now 7365 mEffect.clear(); 7366 if (mClient != 0) { 7367 if (mCblk != NULL) { 7368 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7369 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7370 } 7371 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7372 // Client destructor must run with AudioFlinger mutex locked 7373 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7374 mClient.clear(); 7375 } 7376} 7377 7378status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7379 uint32_t cmdSize, 7380 void *pCmdData, 7381 uint32_t *replySize, 7382 void *pReplyData) 7383{ 7384// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7385// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7386 7387 // only get parameter command is permitted for applications not controlling the effect 7388 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7389 return INVALID_OPERATION; 7390 } 7391 if (mEffect == 0) return DEAD_OBJECT; 7392 if (mClient == 0) return INVALID_OPERATION; 7393 7394 // handle commands that are not forwarded transparently to effect engine 7395 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7396 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7397 // no risk to block the whole media server process or mixer threads is we are stuck here 7398 Mutex::Autolock _l(mCblk->lock); 7399 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7400 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7401 mCblk->serverIndex = 0; 7402 mCblk->clientIndex = 0; 7403 return BAD_VALUE; 7404 } 7405 status_t status = NO_ERROR; 7406 while (mCblk->serverIndex < mCblk->clientIndex) { 7407 int reply; 7408 uint32_t rsize = sizeof(int); 7409 int *p = (int *)(mBuffer + mCblk->serverIndex); 7410 int size = *p++; 7411 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7412 ALOGW("command(): invalid parameter block size"); 7413 break; 7414 } 7415 effect_param_t *param = (effect_param_t *)p; 7416 if (param->psize == 0 || param->vsize == 0) { 7417 ALOGW("command(): null parameter or value size"); 7418 mCblk->serverIndex += size; 7419 continue; 7420 } 7421 uint32_t psize = sizeof(effect_param_t) + 7422 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7423 param->vsize; 7424 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7425 psize, 7426 p, 7427 &rsize, 7428 &reply); 7429 // stop at first error encountered 7430 if (ret != NO_ERROR) { 7431 status = ret; 7432 *(int *)pReplyData = reply; 7433 break; 7434 } else if (reply != NO_ERROR) { 7435 *(int *)pReplyData = reply; 7436 break; 7437 } 7438 mCblk->serverIndex += size; 7439 } 7440 mCblk->serverIndex = 0; 7441 mCblk->clientIndex = 0; 7442 return status; 7443 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7444 *(int *)pReplyData = NO_ERROR; 7445 return enable(); 7446 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7447 *(int *)pReplyData = NO_ERROR; 7448 return disable(); 7449 } 7450 7451 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7452} 7453 7454void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7455{ 7456 ALOGV("setControl %p control %d", this, hasControl); 7457 7458 mHasControl = hasControl; 7459 mEnabled = enabled; 7460 7461 if (signal && mEffectClient != 0) { 7462 mEffectClient->controlStatusChanged(hasControl); 7463 } 7464} 7465 7466void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7467 uint32_t cmdSize, 7468 void *pCmdData, 7469 uint32_t replySize, 7470 void *pReplyData) 7471{ 7472 if (mEffectClient != 0) { 7473 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7474 } 7475} 7476 7477 7478 7479void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7480{ 7481 if (mEffectClient != 0) { 7482 mEffectClient->enableStatusChanged(enabled); 7483 } 7484} 7485 7486status_t AudioFlinger::EffectHandle::onTransact( 7487 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7488{ 7489 return BnEffect::onTransact(code, data, reply, flags); 7490} 7491 7492 7493void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7494{ 7495 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7496 7497 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7498 (mClient == 0) ? getpid_cached : mClient->pid(), 7499 mPriority, 7500 mHasControl, 7501 !locked, 7502 mCblk ? mCblk->clientIndex : 0, 7503 mCblk ? mCblk->serverIndex : 0 7504 ); 7505 7506 if (locked) { 7507 mCblk->lock.unlock(); 7508 } 7509} 7510 7511#undef LOG_TAG 7512#define LOG_TAG "AudioFlinger::EffectChain" 7513 7514AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7515 int sessionId) 7516 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7517 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7518 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7519{ 7520 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7521 if (thread == NULL) { 7522 return; 7523 } 7524 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7525 thread->frameCount(); 7526} 7527 7528AudioFlinger::EffectChain::~EffectChain() 7529{ 7530 if (mOwnInBuffer) { 7531 delete mInBuffer; 7532 } 7533 7534} 7535 7536// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7537sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7538{ 7539 size_t size = mEffects.size(); 7540 7541 for (size_t i = 0; i < size; i++) { 7542 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7543 return mEffects[i]; 7544 } 7545 } 7546 return 0; 7547} 7548 7549// getEffectFromId_l() must be called with ThreadBase::mLock held 7550sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7551{ 7552 size_t size = mEffects.size(); 7553 7554 for (size_t i = 0; i < size; i++) { 7555 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7556 if (id == 0 || mEffects[i]->id() == id) { 7557 return mEffects[i]; 7558 } 7559 } 7560 return 0; 7561} 7562 7563// getEffectFromType_l() must be called with ThreadBase::mLock held 7564sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7565 const effect_uuid_t *type) 7566{ 7567 size_t size = mEffects.size(); 7568 7569 for (size_t i = 0; i < size; i++) { 7570 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7571 return mEffects[i]; 7572 } 7573 } 7574 return 0; 7575} 7576 7577// Must be called with EffectChain::mLock locked 7578void AudioFlinger::EffectChain::process_l() 7579{ 7580 sp<ThreadBase> thread = mThread.promote(); 7581 if (thread == 0) { 7582 ALOGW("process_l(): cannot promote mixer thread"); 7583 return; 7584 } 7585 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7586 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7587 // always process effects unless no more tracks are on the session and the effect tail 7588 // has been rendered 7589 bool doProcess = true; 7590 if (!isGlobalSession) { 7591 bool tracksOnSession = (trackCnt() != 0); 7592 7593 if (!tracksOnSession && mTailBufferCount == 0) { 7594 doProcess = false; 7595 } 7596 7597 if (activeTrackCnt() == 0) { 7598 // if no track is active and the effect tail has not been rendered, 7599 // the input buffer must be cleared here as the mixer process will not do it 7600 if (tracksOnSession || mTailBufferCount > 0) { 7601 size_t numSamples = thread->frameCount() * thread->channelCount(); 7602 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7603 if (mTailBufferCount > 0) { 7604 mTailBufferCount--; 7605 } 7606 } 7607 } 7608 } 7609 7610 size_t size = mEffects.size(); 7611 if (doProcess) { 7612 for (size_t i = 0; i < size; i++) { 7613 mEffects[i]->process(); 7614 } 7615 } 7616 for (size_t i = 0; i < size; i++) { 7617 mEffects[i]->updateState(); 7618 } 7619} 7620 7621// addEffect_l() must be called with PlaybackThread::mLock held 7622status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7623{ 7624 effect_descriptor_t desc = effect->desc(); 7625 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7626 7627 Mutex::Autolock _l(mLock); 7628 effect->setChain(this); 7629 sp<ThreadBase> thread = mThread.promote(); 7630 if (thread == 0) { 7631 return NO_INIT; 7632 } 7633 effect->setThread(thread); 7634 7635 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7636 // Auxiliary effects are inserted at the beginning of mEffects vector as 7637 // they are processed first and accumulated in chain input buffer 7638 mEffects.insertAt(effect, 0); 7639 7640 // the input buffer for auxiliary effect contains mono samples in 7641 // 32 bit format. This is to avoid saturation in AudoMixer 7642 // accumulation stage. Saturation is done in EffectModule::process() before 7643 // calling the process in effect engine 7644 size_t numSamples = thread->frameCount(); 7645 int32_t *buffer = new int32_t[numSamples]; 7646 memset(buffer, 0, numSamples * sizeof(int32_t)); 7647 effect->setInBuffer((int16_t *)buffer); 7648 // auxiliary effects output samples to chain input buffer for further processing 7649 // by insert effects 7650 effect->setOutBuffer(mInBuffer); 7651 } else { 7652 // Insert effects are inserted at the end of mEffects vector as they are processed 7653 // after track and auxiliary effects. 7654 // Insert effect order as a function of indicated preference: 7655 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7656 // another effect is present 7657 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7658 // last effect claiming first position 7659 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7660 // first effect claiming last position 7661 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7662 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7663 // already present 7664 7665 size_t size = mEffects.size(); 7666 size_t idx_insert = size; 7667 ssize_t idx_insert_first = -1; 7668 ssize_t idx_insert_last = -1; 7669 7670 for (size_t i = 0; i < size; i++) { 7671 effect_descriptor_t d = mEffects[i]->desc(); 7672 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7673 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7674 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7675 // check invalid effect chaining combinations 7676 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7677 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7678 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7679 return INVALID_OPERATION; 7680 } 7681 // remember position of first insert effect and by default 7682 // select this as insert position for new effect 7683 if (idx_insert == size) { 7684 idx_insert = i; 7685 } 7686 // remember position of last insert effect claiming 7687 // first position 7688 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7689 idx_insert_first = i; 7690 } 7691 // remember position of first insert effect claiming 7692 // last position 7693 if (iPref == EFFECT_FLAG_INSERT_LAST && 7694 idx_insert_last == -1) { 7695 idx_insert_last = i; 7696 } 7697 } 7698 } 7699 7700 // modify idx_insert from first position if needed 7701 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7702 if (idx_insert_last != -1) { 7703 idx_insert = idx_insert_last; 7704 } else { 7705 idx_insert = size; 7706 } 7707 } else { 7708 if (idx_insert_first != -1) { 7709 idx_insert = idx_insert_first + 1; 7710 } 7711 } 7712 7713 // always read samples from chain input buffer 7714 effect->setInBuffer(mInBuffer); 7715 7716 // if last effect in the chain, output samples to chain 7717 // output buffer, otherwise to chain input buffer 7718 if (idx_insert == size) { 7719 if (idx_insert != 0) { 7720 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7721 mEffects[idx_insert-1]->configure(); 7722 } 7723 effect->setOutBuffer(mOutBuffer); 7724 } else { 7725 effect->setOutBuffer(mInBuffer); 7726 } 7727 mEffects.insertAt(effect, idx_insert); 7728 7729 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7730 } 7731 effect->configure(); 7732 return NO_ERROR; 7733} 7734 7735// removeEffect_l() must be called with PlaybackThread::mLock held 7736size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7737{ 7738 Mutex::Autolock _l(mLock); 7739 size_t size = mEffects.size(); 7740 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7741 7742 for (size_t i = 0; i < size; i++) { 7743 if (effect == mEffects[i]) { 7744 // calling stop here will remove pre-processing effect from the audio HAL. 7745 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7746 // the middle of a read from audio HAL 7747 if (mEffects[i]->state() == EffectModule::ACTIVE || 7748 mEffects[i]->state() == EffectModule::STOPPING) { 7749 mEffects[i]->stop(); 7750 } 7751 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7752 delete[] effect->inBuffer(); 7753 } else { 7754 if (i == size - 1 && i != 0) { 7755 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7756 mEffects[i - 1]->configure(); 7757 } 7758 } 7759 mEffects.removeAt(i); 7760 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7761 break; 7762 } 7763 } 7764 7765 return mEffects.size(); 7766} 7767 7768// setDevice_l() must be called with PlaybackThread::mLock held 7769void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7770{ 7771 size_t size = mEffects.size(); 7772 for (size_t i = 0; i < size; i++) { 7773 mEffects[i]->setDevice(device); 7774 } 7775} 7776 7777// setMode_l() must be called with PlaybackThread::mLock held 7778void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7779{ 7780 size_t size = mEffects.size(); 7781 for (size_t i = 0; i < size; i++) { 7782 mEffects[i]->setMode(mode); 7783 } 7784} 7785 7786// setVolume_l() must be called with PlaybackThread::mLock held 7787bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7788{ 7789 uint32_t newLeft = *left; 7790 uint32_t newRight = *right; 7791 bool hasControl = false; 7792 int ctrlIdx = -1; 7793 size_t size = mEffects.size(); 7794 7795 // first update volume controller 7796 for (size_t i = size; i > 0; i--) { 7797 if (mEffects[i - 1]->isProcessEnabled() && 7798 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7799 ctrlIdx = i - 1; 7800 hasControl = true; 7801 break; 7802 } 7803 } 7804 7805 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7806 if (hasControl) { 7807 *left = mNewLeftVolume; 7808 *right = mNewRightVolume; 7809 } 7810 return hasControl; 7811 } 7812 7813 mVolumeCtrlIdx = ctrlIdx; 7814 mLeftVolume = newLeft; 7815 mRightVolume = newRight; 7816 7817 // second get volume update from volume controller 7818 if (ctrlIdx >= 0) { 7819 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7820 mNewLeftVolume = newLeft; 7821 mNewRightVolume = newRight; 7822 } 7823 // then indicate volume to all other effects in chain. 7824 // Pass altered volume to effects before volume controller 7825 // and requested volume to effects after controller 7826 uint32_t lVol = newLeft; 7827 uint32_t rVol = newRight; 7828 7829 for (size_t i = 0; i < size; i++) { 7830 if ((int)i == ctrlIdx) continue; 7831 // this also works for ctrlIdx == -1 when there is no volume controller 7832 if ((int)i > ctrlIdx) { 7833 lVol = *left; 7834 rVol = *right; 7835 } 7836 mEffects[i]->setVolume(&lVol, &rVol, false); 7837 } 7838 *left = newLeft; 7839 *right = newRight; 7840 7841 return hasControl; 7842} 7843 7844status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7845{ 7846 const size_t SIZE = 256; 7847 char buffer[SIZE]; 7848 String8 result; 7849 7850 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7851 result.append(buffer); 7852 7853 bool locked = tryLock(mLock); 7854 // failed to lock - AudioFlinger is probably deadlocked 7855 if (!locked) { 7856 result.append("\tCould not lock mutex:\n"); 7857 } 7858 7859 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7860 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7861 mEffects.size(), 7862 (uint32_t)mInBuffer, 7863 (uint32_t)mOutBuffer, 7864 mActiveTrackCnt); 7865 result.append(buffer); 7866 write(fd, result.string(), result.size()); 7867 7868 for (size_t i = 0; i < mEffects.size(); ++i) { 7869 sp<EffectModule> effect = mEffects[i]; 7870 if (effect != 0) { 7871 effect->dump(fd, args); 7872 } 7873 } 7874 7875 if (locked) { 7876 mLock.unlock(); 7877 } 7878 7879 return NO_ERROR; 7880} 7881 7882// must be called with ThreadBase::mLock held 7883void AudioFlinger::EffectChain::setEffectSuspended_l( 7884 const effect_uuid_t *type, bool suspend) 7885{ 7886 sp<SuspendedEffectDesc> desc; 7887 // use effect type UUID timelow as key as there is no real risk of identical 7888 // timeLow fields among effect type UUIDs. 7889 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7890 if (suspend) { 7891 if (index >= 0) { 7892 desc = mSuspendedEffects.valueAt(index); 7893 } else { 7894 desc = new SuspendedEffectDesc(); 7895 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7896 mSuspendedEffects.add(type->timeLow, desc); 7897 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7898 } 7899 if (desc->mRefCount++ == 0) { 7900 sp<EffectModule> effect = getEffectIfEnabled(type); 7901 if (effect != 0) { 7902 desc->mEffect = effect; 7903 effect->setSuspended(true); 7904 effect->setEnabled(false); 7905 } 7906 } 7907 } else { 7908 if (index < 0) { 7909 return; 7910 } 7911 desc = mSuspendedEffects.valueAt(index); 7912 if (desc->mRefCount <= 0) { 7913 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7914 desc->mRefCount = 1; 7915 } 7916 if (--desc->mRefCount == 0) { 7917 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7918 if (desc->mEffect != 0) { 7919 sp<EffectModule> effect = desc->mEffect.promote(); 7920 if (effect != 0) { 7921 effect->setSuspended(false); 7922 sp<EffectHandle> handle = effect->controlHandle(); 7923 if (handle != 0) { 7924 effect->setEnabled(handle->enabled()); 7925 } 7926 } 7927 desc->mEffect.clear(); 7928 } 7929 mSuspendedEffects.removeItemsAt(index); 7930 } 7931 } 7932} 7933 7934// must be called with ThreadBase::mLock held 7935void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7936{ 7937 sp<SuspendedEffectDesc> desc; 7938 7939 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7940 if (suspend) { 7941 if (index >= 0) { 7942 desc = mSuspendedEffects.valueAt(index); 7943 } else { 7944 desc = new SuspendedEffectDesc(); 7945 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7946 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7947 } 7948 if (desc->mRefCount++ == 0) { 7949 Vector< sp<EffectModule> > effects; 7950 getSuspendEligibleEffects(effects); 7951 for (size_t i = 0; i < effects.size(); i++) { 7952 setEffectSuspended_l(&effects[i]->desc().type, true); 7953 } 7954 } 7955 } else { 7956 if (index < 0) { 7957 return; 7958 } 7959 desc = mSuspendedEffects.valueAt(index); 7960 if (desc->mRefCount <= 0) { 7961 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7962 desc->mRefCount = 1; 7963 } 7964 if (--desc->mRefCount == 0) { 7965 Vector<const effect_uuid_t *> types; 7966 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7967 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7968 continue; 7969 } 7970 types.add(&mSuspendedEffects.valueAt(i)->mType); 7971 } 7972 for (size_t i = 0; i < types.size(); i++) { 7973 setEffectSuspended_l(types[i], false); 7974 } 7975 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7976 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7977 } 7978 } 7979} 7980 7981 7982// The volume effect is used for automated tests only 7983#ifndef OPENSL_ES_H_ 7984static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7985 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7986const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7987#endif //OPENSL_ES_H_ 7988 7989bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7990{ 7991 // auxiliary effects and visualizer are never suspended on output mix 7992 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7993 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7994 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7995 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7996 return false; 7997 } 7998 return true; 7999} 8000 8001void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8002{ 8003 effects.clear(); 8004 for (size_t i = 0; i < mEffects.size(); i++) { 8005 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8006 effects.add(mEffects[i]); 8007 } 8008 } 8009} 8010 8011sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8012 const effect_uuid_t *type) 8013{ 8014 sp<EffectModule> effect = getEffectFromType_l(type); 8015 return effect != 0 && effect->isEnabled() ? effect : 0; 8016} 8017 8018void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8019 bool enabled) 8020{ 8021 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8022 if (enabled) { 8023 if (index < 0) { 8024 // if the effect is not suspend check if all effects are suspended 8025 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8026 if (index < 0) { 8027 return; 8028 } 8029 if (!isEffectEligibleForSuspend(effect->desc())) { 8030 return; 8031 } 8032 setEffectSuspended_l(&effect->desc().type, enabled); 8033 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8034 if (index < 0) { 8035 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8036 return; 8037 } 8038 } 8039 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8040 effect->desc().type.timeLow); 8041 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8042 // if effect is requested to suspended but was not yet enabled, supend it now. 8043 if (desc->mEffect == 0) { 8044 desc->mEffect = effect; 8045 effect->setEnabled(false); 8046 effect->setSuspended(true); 8047 } 8048 } else { 8049 if (index < 0) { 8050 return; 8051 } 8052 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8053 effect->desc().type.timeLow); 8054 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8055 desc->mEffect.clear(); 8056 effect->setSuspended(false); 8057 } 8058} 8059 8060#undef LOG_TAG 8061#define LOG_TAG "AudioFlinger" 8062 8063// ---------------------------------------------------------------------------- 8064 8065status_t AudioFlinger::onTransact( 8066 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8067{ 8068 return BnAudioFlinger::onTransact(code, data, reply, flags); 8069} 8070 8071}; // namespace android 8072