AudioFlinger.cpp revision d4513b09123deebf8023b73a82d2d46d35806cea
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_IDLE; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_IDLE; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid count\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 assert(NULL != mPrimaryHardwareDev); 727 assert(NULL != mPrimaryHardwareDev->get_master_volume); 728 729 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 730 mHardwareStatus = AUDIO_HW_IDLE; 731 return ret_val; 732 } 733 734 return mMasterVolume; 735} 736 737status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 738 audio_io_handle_t output) 739{ 740 // check calling permissions 741 if (!settingsAllowed()) { 742 return PERMISSION_DENIED; 743 } 744 745 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 746 ALOGE("setStreamVolume() invalid stream %d", stream); 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 PlaybackThread *thread = NULL; 752 if (output) { 753 thread = checkPlaybackThread_l(output); 754 if (thread == NULL) { 755 return BAD_VALUE; 756 } 757 } 758 759 mStreamTypes[stream].volume = value; 760 761 if (thread == NULL) { 762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 763 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 764 } 765 } else { 766 thread->setStreamVolume(stream, value); 767 } 768 769 return NO_ERROR; 770} 771 772status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 773{ 774 // check calling permissions 775 if (!settingsAllowed()) { 776 return PERMISSION_DENIED; 777 } 778 779 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 780 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 781 ALOGE("setStreamMute() invalid stream %d", stream); 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 mStreamTypes[stream].mute = muted; 787 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 789 790 return NO_ERROR; 791} 792 793float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 794{ 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 return 0.0f; 797 } 798 799 AutoMutex lock(mLock); 800 float volume; 801 if (output) { 802 PlaybackThread *thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return 0.0f; 805 } 806 volume = thread->streamVolume(stream); 807 } else { 808 volume = streamVolume_l(stream); 809 } 810 811 return volume; 812} 813 814bool AudioFlinger::streamMute(audio_stream_type_t stream) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return true; 818 } 819 820 AutoMutex lock(mLock); 821 return streamMute_l(stream); 822} 823 824status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 825{ 826 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 827 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 828 // check calling permissions 829 if (!settingsAllowed()) { 830 return PERMISSION_DENIED; 831 } 832 833 // ioHandle == 0 means the parameters are global to the audio hardware interface 834 if (ioHandle == 0) { 835 status_t final_result = NO_ERROR; 836 { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs[i]; 841 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 842 final_result = result ?: final_result; 843 } 844 mHardwareStatus = AUDIO_HW_IDLE; 845 } 846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 847 AudioParameter param = AudioParameter(keyValuePairs); 848 String8 value; 849 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 850 Mutex::Autolock _l(mLock); 851 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 852 if (mBtNrecIsOff != btNrecIsOff) { 853 for (size_t i = 0; i < mRecordThreads.size(); i++) { 854 sp<RecordThread> thread = mRecordThreads.valueAt(i); 855 RecordThread::RecordTrack *track = thread->track(); 856 if (track != NULL) { 857 audio_devices_t device = (audio_devices_t)( 858 thread->device() & AUDIO_DEVICE_IN_ALL); 859 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 860 thread->setEffectSuspended(FX_IID_AEC, 861 suspend, 862 track->sessionId()); 863 thread->setEffectSuspended(FX_IID_NS, 864 suspend, 865 track->sessionId()); 866 } 867 } 868 mBtNrecIsOff = btNrecIsOff; 869 } 870 } 871 return final_result; 872 } 873 874 // hold a strong ref on thread in case closeOutput() or closeInput() is called 875 // and the thread is exited once the lock is released 876 sp<ThreadBase> thread; 877 { 878 Mutex::Autolock _l(mLock); 879 thread = checkPlaybackThread_l(ioHandle); 880 if (thread == NULL) { 881 thread = checkRecordThread_l(ioHandle); 882 } else if (thread == primaryPlaybackThread_l()) { 883 // indicate output device change to all input threads for pre processing 884 AudioParameter param = AudioParameter(keyValuePairs); 885 int value; 886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 887 for (size_t i = 0; i < mRecordThreads.size(); i++) { 888 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 889 } 890 } 891 } 892 } 893 if (thread != 0) { 894 return thread->setParameters(keyValuePairs); 895 } 896 return BAD_VALUE; 897} 898 899String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 900{ 901// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 902// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 903 904 if (ioHandle == 0) { 905 String8 out_s8; 906 907 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 908 char *s; 909 { 910 AutoMutex lock(mHardwareLock); 911 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 912 audio_hw_device_t *dev = mAudioHwDevs[i]; 913 s = dev->get_parameters(dev, keys.string()); 914 mHardwareStatus = AUDIO_HW_IDLE; 915 } 916 out_s8 += String8(s ? s : ""); 917 free(s); 918 } 919 return out_s8; 920 } 921 922 Mutex::Autolock _l(mLock); 923 924 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 925 if (playbackThread != NULL) { 926 return playbackThread->getParameters(keys); 927 } 928 RecordThread *recordThread = checkRecordThread_l(ioHandle); 929 if (recordThread != NULL) { 930 return recordThread->getParameters(keys); 931 } 932 return String8(""); 933} 934 935size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 936{ 937 status_t ret = initCheck(); 938 if (ret != NO_ERROR) { 939 return 0; 940 } 941 942 AutoMutex lock(mHardwareLock); 943 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 944 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 945 mHardwareStatus = AUDIO_HW_IDLE; 946 return size; 947} 948 949unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 950{ 951 if (ioHandle == 0) { 952 return 0; 953 } 954 955 Mutex::Autolock _l(mLock); 956 957 RecordThread *recordThread = checkRecordThread_l(ioHandle); 958 if (recordThread != NULL) { 959 return recordThread->getInputFramesLost(); 960 } 961 return 0; 962} 963 964status_t AudioFlinger::setVoiceVolume(float value) 965{ 966 status_t ret = initCheck(); 967 if (ret != NO_ERROR) { 968 return ret; 969 } 970 971 // check calling permissions 972 if (!settingsAllowed()) { 973 return PERMISSION_DENIED; 974 } 975 976 AutoMutex lock(mHardwareLock); 977 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 978 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 979 mHardwareStatus = AUDIO_HW_IDLE; 980 981 return ret; 982} 983 984status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 985 audio_io_handle_t output) const 986{ 987 status_t status; 988 989 Mutex::Autolock _l(mLock); 990 991 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 992 if (playbackThread != NULL) { 993 return playbackThread->getRenderPosition(halFrames, dspFrames); 994 } 995 996 return BAD_VALUE; 997} 998 999void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1000{ 1001 1002 Mutex::Autolock _l(mLock); 1003 1004 pid_t pid = IPCThreadState::self()->getCallingPid(); 1005 if (mNotificationClients.indexOfKey(pid) < 0) { 1006 sp<NotificationClient> notificationClient = new NotificationClient(this, 1007 client, 1008 pid); 1009 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1010 1011 mNotificationClients.add(pid, notificationClient); 1012 1013 sp<IBinder> binder = client->asBinder(); 1014 binder->linkToDeath(notificationClient); 1015 1016 // the config change is always sent from playback or record threads to avoid deadlock 1017 // with AudioSystem::gLock 1018 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1019 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1020 } 1021 1022 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1023 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1024 } 1025 } 1026} 1027 1028void AudioFlinger::removeNotificationClient(pid_t pid) 1029{ 1030 Mutex::Autolock _l(mLock); 1031 1032 mNotificationClients.removeItem(pid); 1033 1034 ALOGV("%d died, releasing its sessions", pid); 1035 size_t num = mAudioSessionRefs.size(); 1036 bool removed = false; 1037 for (size_t i = 0; i< num; ) { 1038 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1039 ALOGV(" pid %d @ %d", ref->mPid, i); 1040 if (ref->mPid == pid) { 1041 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1042 mAudioSessionRefs.removeAt(i); 1043 delete ref; 1044 removed = true; 1045 num--; 1046 } else { 1047 i++; 1048 } 1049 } 1050 if (removed) { 1051 purgeStaleEffects_l(); 1052 } 1053} 1054 1055// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1056void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, void *param2) 1057{ 1058 size_t size = mNotificationClients.size(); 1059 for (size_t i = 0; i < size; i++) { 1060 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1061 param2); 1062 } 1063} 1064 1065// removeClient_l() must be called with AudioFlinger::mLock held 1066void AudioFlinger::removeClient_l(pid_t pid) 1067{ 1068 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1069 mClients.removeItem(pid); 1070} 1071 1072 1073// ---------------------------------------------------------------------------- 1074 1075AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1076 uint32_t device, type_t type) 1077 : Thread(false), 1078 mType(type), 1079 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1080 // mChannelMask 1081 mChannelCount(0), 1082 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1083 mParamStatus(NO_ERROR), 1084 mStandby(false), mId(id), 1085 mDevice(device), 1086 mDeathRecipient(new PMDeathRecipient(this)) 1087{ 1088} 1089 1090AudioFlinger::ThreadBase::~ThreadBase() 1091{ 1092 mParamCond.broadcast(); 1093 // do not lock the mutex in destructor 1094 releaseWakeLock_l(); 1095 if (mPowerManager != 0) { 1096 sp<IBinder> binder = mPowerManager->asBinder(); 1097 binder->unlinkToDeath(mDeathRecipient); 1098 } 1099} 1100 1101void AudioFlinger::ThreadBase::exit() 1102{ 1103 ALOGV("ThreadBase::exit"); 1104 { 1105 // This lock prevents the following race in thread (uniprocessor for illustration): 1106 // if (!exitPending()) { 1107 // // context switch from here to exit() 1108 // // exit() calls requestExit(), what exitPending() observes 1109 // // exit() calls signal(), which is dropped since no waiters 1110 // // context switch back from exit() to here 1111 // mWaitWorkCV.wait(...); 1112 // // now thread is hung 1113 // } 1114 AutoMutex lock(mLock); 1115 requestExit(); 1116 mWaitWorkCV.signal(); 1117 } 1118 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1119 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1120 requestExitAndWait(); 1121} 1122 1123status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1124{ 1125 status_t status; 1126 1127 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1128 Mutex::Autolock _l(mLock); 1129 1130 mNewParameters.add(keyValuePairs); 1131 mWaitWorkCV.signal(); 1132 // wait condition with timeout in case the thread loop has exited 1133 // before the request could be processed 1134 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1135 status = mParamStatus; 1136 mWaitWorkCV.signal(); 1137 } else { 1138 status = TIMED_OUT; 1139 } 1140 return status; 1141} 1142 1143void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1144{ 1145 Mutex::Autolock _l(mLock); 1146 sendConfigEvent_l(event, param); 1147} 1148 1149// sendConfigEvent_l() must be called with ThreadBase::mLock held 1150void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1151{ 1152 ConfigEvent configEvent; 1153 configEvent.mEvent = event; 1154 configEvent.mParam = param; 1155 mConfigEvents.add(configEvent); 1156 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1157 mWaitWorkCV.signal(); 1158} 1159 1160void AudioFlinger::ThreadBase::processConfigEvents() 1161{ 1162 mLock.lock(); 1163 while(!mConfigEvents.isEmpty()) { 1164 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1165 ConfigEvent configEvent = mConfigEvents[0]; 1166 mConfigEvents.removeAt(0); 1167 // release mLock before locking AudioFlinger mLock: lock order is always 1168 // AudioFlinger then ThreadBase to avoid cross deadlock 1169 mLock.unlock(); 1170 mAudioFlinger->mLock.lock(); 1171 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1172 mAudioFlinger->mLock.unlock(); 1173 mLock.lock(); 1174 } 1175 mLock.unlock(); 1176} 1177 1178status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1179{ 1180 const size_t SIZE = 256; 1181 char buffer[SIZE]; 1182 String8 result; 1183 1184 bool locked = tryLock(mLock); 1185 if (!locked) { 1186 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1187 write(fd, buffer, strlen(buffer)); 1188 } 1189 1190 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1191 result.append(buffer); 1192 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1193 result.append(buffer); 1194 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1197 result.append(buffer); 1198 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1199 result.append(buffer); 1200 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1201 result.append(buffer); 1202 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1203 result.append(buffer); 1204 1205 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1206 result.append(buffer); 1207 result.append(" Index Command"); 1208 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1209 snprintf(buffer, SIZE, "\n %02d ", i); 1210 result.append(buffer); 1211 result.append(mNewParameters[i]); 1212 } 1213 1214 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, " Index event param\n"); 1217 result.append(buffer); 1218 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1219 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1220 result.append(buffer); 1221 } 1222 result.append("\n"); 1223 1224 write(fd, result.string(), result.size()); 1225 1226 if (locked) { 1227 mLock.unlock(); 1228 } 1229 return NO_ERROR; 1230} 1231 1232status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1233{ 1234 const size_t SIZE = 256; 1235 char buffer[SIZE]; 1236 String8 result; 1237 1238 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1239 write(fd, buffer, strlen(buffer)); 1240 1241 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1242 sp<EffectChain> chain = mEffectChains[i]; 1243 if (chain != 0) { 1244 chain->dump(fd, args); 1245 } 1246 } 1247 return NO_ERROR; 1248} 1249 1250void AudioFlinger::ThreadBase::acquireWakeLock() 1251{ 1252 Mutex::Autolock _l(mLock); 1253 acquireWakeLock_l(); 1254} 1255 1256void AudioFlinger::ThreadBase::acquireWakeLock_l() 1257{ 1258 if (mPowerManager == 0) { 1259 // use checkService() to avoid blocking if power service is not up yet 1260 sp<IBinder> binder = 1261 defaultServiceManager()->checkService(String16("power")); 1262 if (binder == 0) { 1263 ALOGW("Thread %s cannot connect to the power manager service", mName); 1264 } else { 1265 mPowerManager = interface_cast<IPowerManager>(binder); 1266 binder->linkToDeath(mDeathRecipient); 1267 } 1268 } 1269 if (mPowerManager != 0) { 1270 sp<IBinder> binder = new BBinder(); 1271 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1272 binder, 1273 String16(mName)); 1274 if (status == NO_ERROR) { 1275 mWakeLockToken = binder; 1276 } 1277 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1278 } 1279} 1280 1281void AudioFlinger::ThreadBase::releaseWakeLock() 1282{ 1283 Mutex::Autolock _l(mLock); 1284 releaseWakeLock_l(); 1285} 1286 1287void AudioFlinger::ThreadBase::releaseWakeLock_l() 1288{ 1289 if (mWakeLockToken != 0) { 1290 ALOGV("releaseWakeLock_l() %s", mName); 1291 if (mPowerManager != 0) { 1292 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1293 } 1294 mWakeLockToken.clear(); 1295 } 1296} 1297 1298void AudioFlinger::ThreadBase::clearPowerManager() 1299{ 1300 Mutex::Autolock _l(mLock); 1301 releaseWakeLock_l(); 1302 mPowerManager.clear(); 1303} 1304 1305void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1306{ 1307 sp<ThreadBase> thread = mThread.promote(); 1308 if (thread != 0) { 1309 thread->clearPowerManager(); 1310 } 1311 ALOGW("power manager service died !!!"); 1312} 1313 1314void AudioFlinger::ThreadBase::setEffectSuspended( 1315 const effect_uuid_t *type, bool suspend, int sessionId) 1316{ 1317 Mutex::Autolock _l(mLock); 1318 setEffectSuspended_l(type, suspend, sessionId); 1319} 1320 1321void AudioFlinger::ThreadBase::setEffectSuspended_l( 1322 const effect_uuid_t *type, bool suspend, int sessionId) 1323{ 1324 sp<EffectChain> chain = getEffectChain_l(sessionId); 1325 if (chain != 0) { 1326 if (type != NULL) { 1327 chain->setEffectSuspended_l(type, suspend); 1328 } else { 1329 chain->setEffectSuspendedAll_l(suspend); 1330 } 1331 } 1332 1333 updateSuspendedSessions_l(type, suspend, sessionId); 1334} 1335 1336void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1337{ 1338 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1339 if (index < 0) { 1340 return; 1341 } 1342 1343 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1344 mSuspendedSessions.editValueAt(index); 1345 1346 for (size_t i = 0; i < sessionEffects.size(); i++) { 1347 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1348 for (int j = 0; j < desc->mRefCount; j++) { 1349 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1350 chain->setEffectSuspendedAll_l(true); 1351 } else { 1352 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1353 desc->mType.timeLow); 1354 chain->setEffectSuspended_l(&desc->mType, true); 1355 } 1356 } 1357 } 1358} 1359 1360void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1361 bool suspend, 1362 int sessionId) 1363{ 1364 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1365 1366 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1367 1368 if (suspend) { 1369 if (index >= 0) { 1370 sessionEffects = mSuspendedSessions.editValueAt(index); 1371 } else { 1372 mSuspendedSessions.add(sessionId, sessionEffects); 1373 } 1374 } else { 1375 if (index < 0) { 1376 return; 1377 } 1378 sessionEffects = mSuspendedSessions.editValueAt(index); 1379 } 1380 1381 1382 int key = EffectChain::kKeyForSuspendAll; 1383 if (type != NULL) { 1384 key = type->timeLow; 1385 } 1386 index = sessionEffects.indexOfKey(key); 1387 1388 sp <SuspendedSessionDesc> desc; 1389 if (suspend) { 1390 if (index >= 0) { 1391 desc = sessionEffects.valueAt(index); 1392 } else { 1393 desc = new SuspendedSessionDesc(); 1394 if (type != NULL) { 1395 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1396 } 1397 sessionEffects.add(key, desc); 1398 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1399 } 1400 desc->mRefCount++; 1401 } else { 1402 if (index < 0) { 1403 return; 1404 } 1405 desc = sessionEffects.valueAt(index); 1406 if (--desc->mRefCount == 0) { 1407 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1408 sessionEffects.removeItemsAt(index); 1409 if (sessionEffects.isEmpty()) { 1410 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1411 sessionId); 1412 mSuspendedSessions.removeItem(sessionId); 1413 } 1414 } 1415 } 1416 if (!sessionEffects.isEmpty()) { 1417 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1418 } 1419} 1420 1421void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1422 bool enabled, 1423 int sessionId) 1424{ 1425 Mutex::Autolock _l(mLock); 1426 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1427} 1428 1429void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1430 bool enabled, 1431 int sessionId) 1432{ 1433 if (mType != RECORD) { 1434 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1435 // another session. This gives the priority to well behaved effect control panels 1436 // and applications not using global effects. 1437 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1438 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1439 } 1440 } 1441 1442 sp<EffectChain> chain = getEffectChain_l(sessionId); 1443 if (chain != 0) { 1444 chain->checkSuspendOnEffectEnabled(effect, enabled); 1445 } 1446} 1447 1448// ---------------------------------------------------------------------------- 1449 1450AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1451 AudioStreamOut* output, 1452 audio_io_handle_t id, 1453 uint32_t device, 1454 type_t type) 1455 : ThreadBase(audioFlinger, id, device, type), 1456 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1457 // Assumes constructor is called by AudioFlinger with it's mLock held, 1458 // but it would be safer to explicitly pass initial masterMute as parameter 1459 mMasterMute(audioFlinger->masterMute_l()), 1460 // mStreamTypes[] initialized in constructor body 1461 mOutput(output), 1462 // Assumes constructor is called by AudioFlinger with it's mLock held, 1463 // but it would be safer to explicitly pass initial masterVolume as parameter 1464 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1465 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1466{ 1467 snprintf(mName, kNameLength, "AudioOut_%X", id); 1468 1469 readOutputParameters(); 1470 1471 // mStreamTypes[AUDIO_STREAM_CNT] is initialized by stream_type_t default constructor 1472 // There is no AUDIO_STREAM_MIN, and ++ operator does not compile 1473 for (audio_stream_type_t stream = (audio_stream_type_t) 0; stream < AUDIO_STREAM_CNT; 1474 stream = (audio_stream_type_t) (stream + 1)) { 1475 mStreamTypes[stream].volume = mAudioFlinger->streamVolume_l(stream); 1476 mStreamTypes[stream].mute = mAudioFlinger->streamMute_l(stream); 1477 // initialized by stream_type_t default constructor 1478 // mStreamTypes[stream].valid = true; 1479 } 1480 // mStreamTypes[AUDIO_STREAM_CNT] exists but isn't explicitly initialized here, 1481 // because mAudioFlinger doesn't have one to copy from 1482} 1483 1484AudioFlinger::PlaybackThread::~PlaybackThread() 1485{ 1486 delete [] mMixBuffer; 1487} 1488 1489status_t AudioFlinger::PlaybackThread::dump(int fd, const Vector<String16>& args) 1490{ 1491 dumpInternals(fd, args); 1492 dumpTracks(fd, args); 1493 dumpEffectChains(fd, args); 1494 return NO_ERROR; 1495} 1496 1497status_t AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args) 1498{ 1499 const size_t SIZE = 256; 1500 char buffer[SIZE]; 1501 String8 result; 1502 1503 snprintf(buffer, SIZE, "Output thread %p tracks\n", this); 1504 result.append(buffer); 1505 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1506 for (size_t i = 0; i < mTracks.size(); ++i) { 1507 sp<Track> track = mTracks[i]; 1508 if (track != 0) { 1509 track->dump(buffer, SIZE); 1510 result.append(buffer); 1511 } 1512 } 1513 1514 snprintf(buffer, SIZE, "Output thread %p active tracks\n", this); 1515 result.append(buffer); 1516 result.append(" Name Clien Typ Fmt Chn mask Session Buf S M F SRate LeftV RighV Serv User Main buf Aux Buf\n"); 1517 for (size_t i = 0; i < mActiveTracks.size(); ++i) { 1518 sp<Track> track = mActiveTracks[i].promote(); 1519 if (track != 0) { 1520 track->dump(buffer, SIZE); 1521 result.append(buffer); 1522 } 1523 } 1524 write(fd, result.string(), result.size()); 1525 return NO_ERROR; 1526} 1527 1528status_t AudioFlinger::PlaybackThread::dumpInternals(int fd, const Vector<String16>& args) 1529{ 1530 const size_t SIZE = 256; 1531 char buffer[SIZE]; 1532 String8 result; 1533 1534 snprintf(buffer, SIZE, "\nOutput thread %p internals\n", this); 1535 result.append(buffer); 1536 snprintf(buffer, SIZE, "last write occurred (msecs): %llu\n", ns2ms(systemTime() - mLastWriteTime)); 1537 result.append(buffer); 1538 snprintf(buffer, SIZE, "total writes: %d\n", mNumWrites); 1539 result.append(buffer); 1540 snprintf(buffer, SIZE, "delayed writes: %d\n", mNumDelayedWrites); 1541 result.append(buffer); 1542 snprintf(buffer, SIZE, "blocked in write: %d\n", mInWrite); 1543 result.append(buffer); 1544 snprintf(buffer, SIZE, "suspend count: %d\n", mSuspended); 1545 result.append(buffer); 1546 snprintf(buffer, SIZE, "mix buffer : %p\n", mMixBuffer); 1547 result.append(buffer); 1548 write(fd, result.string(), result.size()); 1549 1550 dumpBase(fd, args); 1551 1552 return NO_ERROR; 1553} 1554 1555// Thread virtuals 1556status_t AudioFlinger::PlaybackThread::readyToRun() 1557{ 1558 status_t status = initCheck(); 1559 if (status == NO_ERROR) { 1560 ALOGI("AudioFlinger's thread %p ready to run", this); 1561 } else { 1562 ALOGE("No working audio driver found."); 1563 } 1564 return status; 1565} 1566 1567void AudioFlinger::PlaybackThread::onFirstRef() 1568{ 1569 run(mName, ANDROID_PRIORITY_URGENT_AUDIO); 1570} 1571 1572// PlaybackThread::createTrack_l() must be called with AudioFlinger::mLock held 1573sp<AudioFlinger::PlaybackThread::Track> AudioFlinger::PlaybackThread::createTrack_l( 1574 const sp<AudioFlinger::Client>& client, 1575 audio_stream_type_t streamType, 1576 uint32_t sampleRate, 1577 audio_format_t format, 1578 uint32_t channelMask, 1579 int frameCount, 1580 const sp<IMemory>& sharedBuffer, 1581 int sessionId, 1582 bool isTimed, 1583 status_t *status) 1584{ 1585 sp<Track> track; 1586 status_t lStatus; 1587 1588 if (mType == DIRECT) { 1589 if ((format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { 1590 if (sampleRate != mSampleRate || format != mFormat || channelMask != mChannelMask) { 1591 ALOGE("createTrack_l() Bad parameter: sampleRate %d format %d, channelMask 0x%08x \"" 1592 "for output %p with format %d", 1593 sampleRate, format, channelMask, mOutput, mFormat); 1594 lStatus = BAD_VALUE; 1595 goto Exit; 1596 } 1597 } 1598 } else { 1599 // Resampler implementation limits input sampling rate to 2 x output sampling rate. 1600 if (sampleRate > mSampleRate*2) { 1601 ALOGE("Sample rate out of range: %d mSampleRate %d", sampleRate, mSampleRate); 1602 lStatus = BAD_VALUE; 1603 goto Exit; 1604 } 1605 } 1606 1607 lStatus = initCheck(); 1608 if (lStatus != NO_ERROR) { 1609 ALOGE("Audio driver not initialized."); 1610 goto Exit; 1611 } 1612 1613 { // scope for mLock 1614 Mutex::Autolock _l(mLock); 1615 1616 // all tracks in same audio session must share the same routing strategy otherwise 1617 // conflicts will happen when tracks are moved from one output to another by audio policy 1618 // manager 1619 uint32_t strategy = AudioSystem::getStrategyForStream(streamType); 1620 for (size_t i = 0; i < mTracks.size(); ++i) { 1621 sp<Track> t = mTracks[i]; 1622 if (t != 0) { 1623 uint32_t actual = AudioSystem::getStrategyForStream(t->streamType()); 1624 if (sessionId == t->sessionId() && strategy != actual) { 1625 ALOGE("createTrack_l() mismatched strategy; expected %u but found %u", 1626 strategy, actual); 1627 lStatus = BAD_VALUE; 1628 goto Exit; 1629 } 1630 } 1631 } 1632 1633 if (!isTimed) { 1634 track = new Track(this, client, streamType, sampleRate, format, 1635 channelMask, frameCount, sharedBuffer, sessionId); 1636 } else { 1637 track = TimedTrack::create(this, client, streamType, sampleRate, format, 1638 channelMask, frameCount, sharedBuffer, sessionId); 1639 } 1640 if (track == NULL || track->getCblk() == NULL || track->name() < 0) { 1641 lStatus = NO_MEMORY; 1642 goto Exit; 1643 } 1644 mTracks.add(track); 1645 1646 sp<EffectChain> chain = getEffectChain_l(sessionId); 1647 if (chain != 0) { 1648 ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer()); 1649 track->setMainBuffer(chain->inBuffer()); 1650 chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType())); 1651 chain->incTrackCnt(); 1652 } 1653 1654 // invalidate track immediately if the stream type was moved to another thread since 1655 // createTrack() was called by the client process. 1656 if (!mStreamTypes[streamType].valid) { 1657 ALOGW("createTrack_l() on thread %p: invalidating track on stream %d", 1658 this, streamType); 1659 android_atomic_or(CBLK_INVALID_ON, &track->mCblk->flags); 1660 } 1661 } 1662 lStatus = NO_ERROR; 1663 1664Exit: 1665 if(status) { 1666 *status = lStatus; 1667 } 1668 return track; 1669} 1670 1671uint32_t AudioFlinger::PlaybackThread::latency() const 1672{ 1673 Mutex::Autolock _l(mLock); 1674 if (initCheck() == NO_ERROR) { 1675 return mOutput->stream->get_latency(mOutput->stream); 1676 } else { 1677 return 0; 1678 } 1679} 1680 1681void AudioFlinger::PlaybackThread::setMasterVolume(float value) 1682{ 1683 Mutex::Autolock _l(mLock); 1684 mMasterVolume = value; 1685} 1686 1687void AudioFlinger::PlaybackThread::setMasterMute(bool muted) 1688{ 1689 Mutex::Autolock _l(mLock); 1690 setMasterMute_l(muted); 1691} 1692 1693void AudioFlinger::PlaybackThread::setStreamVolume(audio_stream_type_t stream, float value) 1694{ 1695 Mutex::Autolock _l(mLock); 1696 mStreamTypes[stream].volume = value; 1697} 1698 1699void AudioFlinger::PlaybackThread::setStreamMute(audio_stream_type_t stream, bool muted) 1700{ 1701 Mutex::Autolock _l(mLock); 1702 mStreamTypes[stream].mute = muted; 1703} 1704 1705float AudioFlinger::PlaybackThread::streamVolume(audio_stream_type_t stream) const 1706{ 1707 Mutex::Autolock _l(mLock); 1708 return mStreamTypes[stream].volume; 1709} 1710 1711// addTrack_l() must be called with ThreadBase::mLock held 1712status_t AudioFlinger::PlaybackThread::addTrack_l(const sp<Track>& track) 1713{ 1714 status_t status = ALREADY_EXISTS; 1715 1716 // set retry count for buffer fill 1717 track->mRetryCount = kMaxTrackStartupRetries; 1718 if (mActiveTracks.indexOf(track) < 0) { 1719 // the track is newly added, make sure it fills up all its 1720 // buffers before playing. This is to ensure the client will 1721 // effectively get the latency it requested. 1722 track->mFillingUpStatus = Track::FS_FILLING; 1723 track->mResetDone = false; 1724 mActiveTracks.add(track); 1725 if (track->mainBuffer() != mMixBuffer) { 1726 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1727 if (chain != 0) { 1728 ALOGV("addTrack_l() starting track on chain %p for session %d", chain.get(), track->sessionId()); 1729 chain->incActiveTrackCnt(); 1730 } 1731 } 1732 1733 status = NO_ERROR; 1734 } 1735 1736 ALOGV("mWaitWorkCV.broadcast"); 1737 mWaitWorkCV.broadcast(); 1738 1739 return status; 1740} 1741 1742// destroyTrack_l() must be called with ThreadBase::mLock held 1743void AudioFlinger::PlaybackThread::destroyTrack_l(const sp<Track>& track) 1744{ 1745 track->mState = TrackBase::TERMINATED; 1746 if (mActiveTracks.indexOf(track) < 0) { 1747 removeTrack_l(track); 1748 } 1749} 1750 1751void AudioFlinger::PlaybackThread::removeTrack_l(const sp<Track>& track) 1752{ 1753 mTracks.remove(track); 1754 deleteTrackName_l(track->name()); 1755 sp<EffectChain> chain = getEffectChain_l(track->sessionId()); 1756 if (chain != 0) { 1757 chain->decTrackCnt(); 1758 } 1759} 1760 1761String8 AudioFlinger::PlaybackThread::getParameters(const String8& keys) 1762{ 1763 String8 out_s8 = String8(""); 1764 char *s; 1765 1766 Mutex::Autolock _l(mLock); 1767 if (initCheck() != NO_ERROR) { 1768 return out_s8; 1769 } 1770 1771 s = mOutput->stream->common.get_parameters(&mOutput->stream->common, keys.string()); 1772 out_s8 = String8(s); 1773 free(s); 1774 return out_s8; 1775} 1776 1777// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1778void AudioFlinger::PlaybackThread::audioConfigChanged_l(int event, int param) { 1779 AudioSystem::OutputDescriptor desc; 1780 void *param2 = NULL; 1781 1782 ALOGV("PlaybackThread::audioConfigChanged_l, thread %p, event %d, param %d", this, event, param); 1783 1784 switch (event) { 1785 case AudioSystem::OUTPUT_OPENED: 1786 case AudioSystem::OUTPUT_CONFIG_CHANGED: 1787 desc.channels = mChannelMask; 1788 desc.samplingRate = mSampleRate; 1789 desc.format = mFormat; 1790 desc.frameCount = mFrameCount; 1791 desc.latency = latency(); 1792 param2 = &desc; 1793 break; 1794 1795 case AudioSystem::STREAM_CONFIG_CHANGED: 1796 param2 = ¶m; 1797 case AudioSystem::OUTPUT_CLOSED: 1798 default: 1799 break; 1800 } 1801 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 1802} 1803 1804void AudioFlinger::PlaybackThread::readOutputParameters() 1805{ 1806 mSampleRate = mOutput->stream->common.get_sample_rate(&mOutput->stream->common); 1807 mChannelMask = mOutput->stream->common.get_channels(&mOutput->stream->common); 1808 mChannelCount = (uint16_t)popcount(mChannelMask); 1809 mFormat = mOutput->stream->common.get_format(&mOutput->stream->common); 1810 mFrameSize = audio_stream_frame_size(&mOutput->stream->common); 1811 mFrameCount = mOutput->stream->common.get_buffer_size(&mOutput->stream->common) / mFrameSize; 1812 1813 // FIXME - Current mixer implementation only supports stereo output: Always 1814 // Allocate a stereo buffer even if HW output is mono. 1815 delete[] mMixBuffer; 1816 mMixBuffer = new int16_t[mFrameCount * 2]; 1817 memset(mMixBuffer, 0, mFrameCount * 2 * sizeof(int16_t)); 1818 1819 // force reconfiguration of effect chains and engines to take new buffer size and audio 1820 // parameters into account 1821 // Note that mLock is not held when readOutputParameters() is called from the constructor 1822 // but in this case nothing is done below as no audio sessions have effect yet so it doesn't 1823 // matter. 1824 // create a copy of mEffectChains as calling moveEffectChain_l() can reorder some effect chains 1825 Vector< sp<EffectChain> > effectChains = mEffectChains; 1826 for (size_t i = 0; i < effectChains.size(); i ++) { 1827 mAudioFlinger->moveEffectChain_l(effectChains[i]->sessionId(), this, this, false); 1828 } 1829} 1830 1831status_t AudioFlinger::PlaybackThread::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames) 1832{ 1833 if (halFrames == NULL || dspFrames == NULL) { 1834 return BAD_VALUE; 1835 } 1836 Mutex::Autolock _l(mLock); 1837 if (initCheck() != NO_ERROR) { 1838 return INVALID_OPERATION; 1839 } 1840 *halFrames = mBytesWritten / audio_stream_frame_size(&mOutput->stream->common); 1841 1842 return mOutput->stream->get_render_position(mOutput->stream, dspFrames); 1843} 1844 1845uint32_t AudioFlinger::PlaybackThread::hasAudioSession(int sessionId) 1846{ 1847 Mutex::Autolock _l(mLock); 1848 uint32_t result = 0; 1849 if (getEffectChain_l(sessionId) != 0) { 1850 result = EFFECT_SESSION; 1851 } 1852 1853 for (size_t i = 0; i < mTracks.size(); ++i) { 1854 sp<Track> track = mTracks[i]; 1855 if (sessionId == track->sessionId() && 1856 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1857 result |= TRACK_SESSION; 1858 break; 1859 } 1860 } 1861 1862 return result; 1863} 1864 1865uint32_t AudioFlinger::PlaybackThread::getStrategyForSession_l(int sessionId) 1866{ 1867 // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that 1868 // it is moved to correct output by audio policy manager when A2DP is connected or disconnected 1869 if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 1870 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1871 } 1872 for (size_t i = 0; i < mTracks.size(); i++) { 1873 sp<Track> track = mTracks[i]; 1874 if (sessionId == track->sessionId() && 1875 !(track->mCblk->flags & CBLK_INVALID_MSK)) { 1876 return AudioSystem::getStrategyForStream(track->streamType()); 1877 } 1878 } 1879 return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 1880} 1881 1882 1883AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::getOutput() const 1884{ 1885 Mutex::Autolock _l(mLock); 1886 return mOutput; 1887} 1888 1889AudioFlinger::AudioStreamOut* AudioFlinger::PlaybackThread::clearOutput() 1890{ 1891 Mutex::Autolock _l(mLock); 1892 AudioStreamOut *output = mOutput; 1893 mOutput = NULL; 1894 return output; 1895} 1896 1897// this method must always be called either with ThreadBase mLock held or inside the thread loop 1898audio_stream_t* AudioFlinger::PlaybackThread::stream() 1899{ 1900 if (mOutput == NULL) { 1901 return NULL; 1902 } 1903 return &mOutput->stream->common; 1904} 1905 1906uint32_t AudioFlinger::PlaybackThread::activeSleepTimeUs() 1907{ 1908 // A2DP output latency is not due only to buffering capacity. It also reflects encoding, 1909 // decoding and transfer time. So sleeping for half of the latency would likely cause 1910 // underruns 1911 if (audio_is_a2dp_device((audio_devices_t)mDevice)) { 1912 return (uint32_t)((uint32_t)((mFrameCount * 1000) / mSampleRate) * 1000); 1913 } else { 1914 return (uint32_t)(mOutput->stream->get_latency(mOutput->stream) * 1000) / 2; 1915 } 1916} 1917 1918// ---------------------------------------------------------------------------- 1919 1920AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output, 1921 audio_io_handle_t id, uint32_t device, type_t type) 1922 : PlaybackThread(audioFlinger, output, id, device, type) 1923{ 1924 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 1925 mPrevMixerStatus = MIXER_IDLE; 1926 // FIXME - Current mixer implementation only supports stereo output 1927 if (mChannelCount == 1) { 1928 ALOGE("Invalid audio hardware channel count"); 1929 } 1930} 1931 1932AudioFlinger::MixerThread::~MixerThread() 1933{ 1934 delete mAudioMixer; 1935} 1936 1937class CpuStats { 1938public: 1939 void sample(); 1940#ifdef DEBUG_CPU_USAGE 1941private: 1942 ThreadCpuUsage mCpu; 1943#endif 1944}; 1945 1946void CpuStats::sample() { 1947#ifdef DEBUG_CPU_USAGE 1948 const CentralTendencyStatistics& stats = mCpu.statistics(); 1949 mCpu.sampleAndEnable(); 1950 unsigned n = stats.n(); 1951 // mCpu.elapsed() is expensive, so don't call it every loop 1952 if ((n & 127) == 1) { 1953 long long elapsed = mCpu.elapsed(); 1954 if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) { 1955 double perLoop = elapsed / (double) n; 1956 double perLoop100 = perLoop * 0.01; 1957 double mean = stats.mean(); 1958 double stddev = stats.stddev(); 1959 double minimum = stats.minimum(); 1960 double maximum = stats.maximum(); 1961 mCpu.resetStatistics(); 1962 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", 1963 elapsed * .000000001, n, perLoop * .000001, 1964 mean * .001, 1965 stddev * .001, 1966 minimum * .001, 1967 maximum * .001, 1968 mean / perLoop100, 1969 stddev / perLoop100, 1970 minimum / perLoop100, 1971 maximum / perLoop100); 1972 } 1973 } 1974#endif 1975}; 1976 1977void AudioFlinger::PlaybackThread::checkSilentMode_l() 1978{ 1979 if (!mMasterMute) { 1980 char value[PROPERTY_VALUE_MAX]; 1981 if (property_get("ro.audio.silent", value, "0") > 0) { 1982 char *endptr; 1983 unsigned long ul = strtoul(value, &endptr, 0); 1984 if (*endptr == '\0' && ul != 0) { 1985 ALOGD("Silence is golden"); 1986 // The setprop command will not allow a property to be changed after 1987 // the first time it is set, so we don't have to worry about un-muting. 1988 setMasterMute_l(true); 1989 } 1990 } 1991 } 1992} 1993 1994bool AudioFlinger::PlaybackThread::threadLoop() 1995{ 1996 Vector< sp<Track> > tracksToRemove; 1997 1998 standbyTime = systemTime(); 1999 mixBufferSize = mFrameCount * mFrameSize; 2000 2001 // MIXER 2002 // FIXME: Relaxed timing because of a certain device that can't meet latency 2003 // Should be reduced to 2x after the vendor fixes the driver issue 2004 // increase threshold again due to low power audio mode. The way this warning threshold is 2005 // calculated and its usefulness should be reconsidered anyway. 2006 nsecs_t maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2007 nsecs_t lastWarning = 0; 2008if (mType == MIXER) { 2009 longStandbyExit = false; 2010} 2011 2012 // DUPLICATING 2013 // FIXME could this be made local to while loop? 2014 writeFrames = 0; 2015 2016 activeSleepTime = activeSleepTimeUs(); 2017 idleSleepTime = idleSleepTimeUs(); 2018 sleepTime = idleSleepTime; 2019 2020if (mType == MIXER) { 2021 sleepTimeShift = 0; 2022} 2023 2024 // MIXER 2025 CpuStats cpuStats; 2026 2027 // DIRECT 2028if (mType == DIRECT) { 2029 // use shorter standby delay as on normal output to release 2030 // hardware resources as soon as possible 2031 standbyDelay = microseconds(activeSleepTime*2); 2032} 2033 2034 acquireWakeLock(); 2035 2036 while (!exitPending()) 2037 { 2038if (mType == MIXER) { 2039 cpuStats.sample(); 2040} 2041 2042 Vector< sp<EffectChain> > effectChains; 2043 2044 processConfigEvents(); 2045 2046if (mType == DIRECT) { 2047 activeTrack.clear(); 2048} 2049 2050 mixerStatus = MIXER_IDLE; 2051 { // scope for mLock 2052 2053 Mutex::Autolock _l(mLock); 2054 2055 if (checkForNewParameters_l()) { 2056 mixBufferSize = mFrameCount * mFrameSize; 2057 2058if (mType == MIXER) { 2059 // FIXME: Relaxed timing because of a certain device that can't meet latency 2060 // Should be reduced to 2x after the vendor fixes the driver issue 2061 // increase threshold again due to low power audio mode. The way this warning 2062 // threshold is calculated and its usefulness should be reconsidered anyway. 2063 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2064} 2065 2066 updateWaitTime_l(); 2067 2068 activeSleepTime = activeSleepTimeUs(); 2069 idleSleepTime = idleSleepTimeUs(); 2070 2071if (mType == DIRECT) { 2072 standbyDelay = microseconds(activeSleepTime*2); 2073} 2074 2075 } 2076 2077if (mType == DUPLICATING) { 2078#if 0 // see earlier FIXME 2079 // Now that this is a field instead of local variable, 2080 // clear it so it is empty the first time through the loop, 2081 // and later an assignment could combine the clear with the loop below 2082 outputTracks.clear(); 2083#endif 2084 for (size_t i = 0; i < mOutputTracks.size(); i++) { 2085 outputTracks.add(mOutputTracks[i]); 2086 } 2087} 2088 2089 // put audio hardware into standby after short delay 2090 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2091 mSuspended > 0)) { 2092 if (!mStandby) { 2093 2094 threadLoop_standby(); 2095 2096 mStandby = true; 2097 mBytesWritten = 0; 2098 } 2099 2100 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2101 // we're about to wait, flush the binder command buffer 2102 IPCThreadState::self()->flushCommands(); 2103 2104if (mType == DUPLICATING) { 2105 outputTracks.clear(); 2106} 2107 2108 if (exitPending()) break; 2109 2110 releaseWakeLock_l(); 2111 // wait until we have something to do... 2112 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2113 mWaitWorkCV.wait(mLock); 2114 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2115 acquireWakeLock_l(); 2116 2117if (mType == MIXER || mType == DUPLICATING) { 2118 mPrevMixerStatus = MIXER_IDLE; 2119} 2120 2121 checkSilentMode_l(); 2122 2123if (mType == MIXER || mType == DUPLICATING) { 2124 standbyTime = systemTime() + mStandbyTimeInNsecs; 2125} 2126 2127if (mType == DIRECT) { 2128 standbyTime = systemTime() + standbyDelay; 2129} 2130 2131 sleepTime = idleSleepTime; 2132 2133if (mType == MIXER) { 2134 sleepTimeShift = 0; 2135} 2136 2137 continue; 2138 } 2139 } 2140 2141 mixerStatus = prepareTracks_l(&tracksToRemove); 2142 // see FIXME in AudioFlinger.h 2143 if (mixerStatus == MIXER_CONTINUE) { 2144 continue; 2145 } 2146 2147 // prevent any changes in effect chain list and in each effect chain 2148 // during mixing and effect process as the audio buffers could be deleted 2149 // or modified if an effect is created or deleted 2150 lockEffectChains_l(effectChains); 2151 } 2152 2153if (mType == DIRECT) { 2154 // For DirectOutputThread, this test is equivalent to "activeTrack != 0" 2155} 2156 2157 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2158 threadLoop_mix(); 2159 } else { 2160 threadLoop_sleepTime(); 2161 } 2162 2163 if (mSuspended > 0) { 2164 sleepTime = suspendSleepTimeUs(); 2165 } 2166 2167 // only process effects if we're going to write 2168 if (sleepTime == 0) { 2169 for (size_t i = 0; i < effectChains.size(); i ++) { 2170 effectChains[i]->process_l(); 2171 } 2172 } 2173 2174 // enable changes in effect chain 2175 unlockEffectChains(effectChains); 2176 2177 // sleepTime == 0 means we must write to audio hardware 2178 if (sleepTime == 0) { 2179 2180 threadLoop_write(); 2181 2182if (mType == MIXER) { 2183 // write blocked detection 2184 nsecs_t now = systemTime(); 2185 nsecs_t delta = now - mLastWriteTime; 2186 if (!mStandby && delta > maxPeriod) { 2187 mNumDelayedWrites++; 2188 if ((now - lastWarning) > kWarningThrottleNs) { 2189 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2190 ns2ms(delta), mNumDelayedWrites, this); 2191 lastWarning = now; 2192 } 2193 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2194 // a different threshold. Or completely removed for what it is worth anyway... 2195 if (mStandby) { 2196 longStandbyExit = true; 2197 } 2198 } 2199} 2200 2201 mStandby = false; 2202 } else { 2203 usleep(sleepTime); 2204 } 2205 2206 // finally let go of removed track(s), without the lock held 2207 // since we can't guarantee the destructors won't acquire that 2208 // same lock. 2209 tracksToRemove.clear(); 2210 2211// FIXME merge these 2212if (mType == DIRECT) { 2213 activeTrack.clear(); 2214} 2215if (mType == DUPLICATING) { 2216 outputTracks.clear(); 2217} 2218 2219 // Effect chains will be actually deleted here if they were removed from 2220 // mEffectChains list during mixing or effects processing 2221 effectChains.clear(); 2222 2223 // FIXME Note that the above .clear() is no longer necessary since effectChains 2224 // is now local to this block, but will keep it for now (at least until merge done). 2225 } 2226 2227if (mType == MIXER || mType == DIRECT) { 2228 // put output stream into standby mode 2229 if (!mStandby) { 2230 mOutput->stream->common.standby(&mOutput->stream->common); 2231 } 2232} 2233if (mType == DUPLICATING) { 2234 // for DuplicatingThread, standby mode is handled by the outputTracks 2235} 2236 2237 releaseWakeLock(); 2238 2239 ALOGV("Thread %p type %d exiting", this, mType); 2240 return false; 2241} 2242 2243// shared by MIXER and DIRECT, overridden by DUPLICATING 2244void AudioFlinger::PlaybackThread::threadLoop_write() 2245{ 2246 // FIXME rewrite to reduce number of system calls 2247 mLastWriteTime = systemTime(); 2248 mInWrite = true; 2249 mBytesWritten += mixBufferSize; 2250 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2251 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2252 mNumWrites++; 2253 mInWrite = false; 2254} 2255 2256// shared by MIXER and DIRECT, overridden by DUPLICATING 2257void AudioFlinger::PlaybackThread::threadLoop_standby() 2258{ 2259 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2260 mOutput->stream->common.standby(&mOutput->stream->common); 2261} 2262 2263void AudioFlinger::MixerThread::threadLoop_mix() 2264{ 2265 // obtain the presentation timestamp of the next output buffer 2266 int64_t pts; 2267 status_t status = INVALID_OPERATION; 2268 2269 if (NULL != mOutput->stream->get_next_write_timestamp) { 2270 status = mOutput->stream->get_next_write_timestamp( 2271 mOutput->stream, &pts); 2272 } 2273 2274 if (status != NO_ERROR) { 2275 pts = AudioBufferProvider::kInvalidPTS; 2276 } 2277 2278 // mix buffers... 2279 mAudioMixer->process(pts); 2280 // increase sleep time progressively when application underrun condition clears. 2281 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2282 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2283 // such that we would underrun the audio HAL. 2284 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2285 sleepTimeShift--; 2286 } 2287 sleepTime = 0; 2288 standbyTime = systemTime() + mStandbyTimeInNsecs; 2289 //TODO: delay standby when effects have a tail 2290} 2291 2292void AudioFlinger::MixerThread::threadLoop_sleepTime() 2293{ 2294 // If no tracks are ready, sleep once for the duration of an output 2295 // buffer size, then write 0s to the output 2296 if (sleepTime == 0) { 2297 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2298 sleepTime = activeSleepTime >> sleepTimeShift; 2299 if (sleepTime < kMinThreadSleepTimeUs) { 2300 sleepTime = kMinThreadSleepTimeUs; 2301 } 2302 // reduce sleep time in case of consecutive application underruns to avoid 2303 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2304 // duration we would end up writing less data than needed by the audio HAL if 2305 // the condition persists. 2306 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2307 sleepTimeShift++; 2308 } 2309 } else { 2310 sleepTime = idleSleepTime; 2311 } 2312 } else if (mBytesWritten != 0 || 2313 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2314 memset (mMixBuffer, 0, mixBufferSize); 2315 sleepTime = 0; 2316 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2317 } 2318 // TODO add standby time extension fct of effect tail 2319} 2320 2321// prepareTracks_l() must be called with ThreadBase::mLock held 2322AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2323 Vector< sp<Track> > *tracksToRemove) 2324{ 2325 2326 mixer_state mixerStatus = MIXER_IDLE; 2327 // find out which tracks need to be processed 2328 size_t count = mActiveTracks.size(); 2329 size_t mixedTracks = 0; 2330 size_t tracksWithEffect = 0; 2331 2332 float masterVolume = mMasterVolume; 2333 bool masterMute = mMasterMute; 2334 2335 if (masterMute) { 2336 masterVolume = 0; 2337 } 2338 // Delegate master volume control to effect in output mix effect chain if needed 2339 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2340 if (chain != 0) { 2341 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2342 chain->setVolume_l(&v, &v); 2343 masterVolume = (float)((v + (1 << 23)) >> 24); 2344 chain.clear(); 2345 } 2346 2347 for (size_t i=0 ; i<count ; i++) { 2348 sp<Track> t = mActiveTracks[i].promote(); 2349 if (t == 0) continue; 2350 2351 // this const just means the local variable doesn't change 2352 Track* const track = t.get(); 2353 audio_track_cblk_t* cblk = track->cblk(); 2354 2355 // The first time a track is added we wait 2356 // for all its buffers to be filled before processing it 2357 int name = track->name(); 2358 // make sure that we have enough frames to mix one full buffer. 2359 // enforce this condition only once to enable draining the buffer in case the client 2360 // app does not call stop() and relies on underrun to stop: 2361 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2362 // during last round 2363 uint32_t minFrames = 1; 2364 if (!track->isStopped() && !track->isPausing() && 2365 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2366 if (t->sampleRate() == (int)mSampleRate) { 2367 minFrames = mFrameCount; 2368 } else { 2369 // +1 for rounding and +1 for additional sample needed for interpolation 2370 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2371 // add frames already consumed but not yet released by the resampler 2372 // because cblk->framesReady() will include these frames 2373 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2374 // the minimum track buffer size is normally twice the number of frames necessary 2375 // to fill one buffer and the resampler should not leave more than one buffer worth 2376 // of unreleased frames after each pass, but just in case... 2377 ALOG_ASSERT(minFrames <= cblk->frameCount); 2378 } 2379 } 2380 if ((track->framesReady() >= minFrames) && track->isReady() && 2381 !track->isPaused() && !track->isTerminated()) 2382 { 2383 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2384 2385 mixedTracks++; 2386 2387 // track->mainBuffer() != mMixBuffer means there is an effect chain 2388 // connected to the track 2389 chain.clear(); 2390 if (track->mainBuffer() != mMixBuffer) { 2391 chain = getEffectChain_l(track->sessionId()); 2392 // Delegate volume control to effect in track effect chain if needed 2393 if (chain != 0) { 2394 tracksWithEffect++; 2395 } else { 2396 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2397 name, track->sessionId()); 2398 } 2399 } 2400 2401 2402 int param = AudioMixer::VOLUME; 2403 if (track->mFillingUpStatus == Track::FS_FILLED) { 2404 // no ramp for the first volume setting 2405 track->mFillingUpStatus = Track::FS_ACTIVE; 2406 if (track->mState == TrackBase::RESUMING) { 2407 track->mState = TrackBase::ACTIVE; 2408 param = AudioMixer::RAMP_VOLUME; 2409 } 2410 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2411 } else if (cblk->server != 0) { 2412 // If the track is stopped before the first frame was mixed, 2413 // do not apply ramp 2414 param = AudioMixer::RAMP_VOLUME; 2415 } 2416 2417 // compute volume for this track 2418 uint32_t vl, vr, va; 2419 if (track->isMuted() || track->isPausing() || 2420 mStreamTypes[track->streamType()].mute) { 2421 vl = vr = va = 0; 2422 if (track->isPausing()) { 2423 track->setPaused(); 2424 } 2425 } else { 2426 2427 // read original volumes with volume control 2428 float typeVolume = mStreamTypes[track->streamType()].volume; 2429 float v = masterVolume * typeVolume; 2430 uint32_t vlr = cblk->getVolumeLR(); 2431 vl = vlr & 0xFFFF; 2432 vr = vlr >> 16; 2433 // track volumes come from shared memory, so can't be trusted and must be clamped 2434 if (vl > MAX_GAIN_INT) { 2435 ALOGV("Track left volume out of range: %04X", vl); 2436 vl = MAX_GAIN_INT; 2437 } 2438 if (vr > MAX_GAIN_INT) { 2439 ALOGV("Track right volume out of range: %04X", vr); 2440 vr = MAX_GAIN_INT; 2441 } 2442 // now apply the master volume and stream type volume 2443 vl = (uint32_t)(v * vl) << 12; 2444 vr = (uint32_t)(v * vr) << 12; 2445 // assuming master volume and stream type volume each go up to 1.0, 2446 // vl and vr are now in 8.24 format 2447 2448 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2449 // send level comes from shared memory and so may be corrupt 2450 if (sendLevel > MAX_GAIN_INT) { 2451 ALOGV("Track send level out of range: %04X", sendLevel); 2452 sendLevel = MAX_GAIN_INT; 2453 } 2454 va = (uint32_t)(v * sendLevel); 2455 } 2456 // Delegate volume control to effect in track effect chain if needed 2457 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2458 // Do not ramp volume if volume is controlled by effect 2459 param = AudioMixer::VOLUME; 2460 track->mHasVolumeController = true; 2461 } else { 2462 // force no volume ramp when volume controller was just disabled or removed 2463 // from effect chain to avoid volume spike 2464 if (track->mHasVolumeController) { 2465 param = AudioMixer::VOLUME; 2466 } 2467 track->mHasVolumeController = false; 2468 } 2469 2470 // Convert volumes from 8.24 to 4.12 format 2471 // This additional clamping is needed in case chain->setVolume_l() overshot 2472 vl = (vl + (1 << 11)) >> 12; 2473 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2474 vr = (vr + (1 << 11)) >> 12; 2475 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2476 2477 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2478 2479 // XXX: these things DON'T need to be done each time 2480 mAudioMixer->setBufferProvider(name, track); 2481 mAudioMixer->enable(name); 2482 2483 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2484 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2485 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2486 mAudioMixer->setParameter( 2487 name, 2488 AudioMixer::TRACK, 2489 AudioMixer::FORMAT, (void *)track->format()); 2490 mAudioMixer->setParameter( 2491 name, 2492 AudioMixer::TRACK, 2493 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2494 mAudioMixer->setParameter( 2495 name, 2496 AudioMixer::RESAMPLE, 2497 AudioMixer::SAMPLE_RATE, 2498 (void *)(cblk->sampleRate)); 2499 mAudioMixer->setParameter( 2500 name, 2501 AudioMixer::TRACK, 2502 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2503 mAudioMixer->setParameter( 2504 name, 2505 AudioMixer::TRACK, 2506 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2507 2508 // reset retry count 2509 track->mRetryCount = kMaxTrackRetries; 2510 // If one track is ready, set the mixer ready if: 2511 // - the mixer was not ready during previous round OR 2512 // - no other track is not ready 2513 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2514 mixerStatus != MIXER_TRACKS_ENABLED) { 2515 mixerStatus = MIXER_TRACKS_READY; 2516 } 2517 } else { 2518 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2519 if (track->isStopped()) { 2520 track->reset(); 2521 } 2522 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2523 // We have consumed all the buffers of this track. 2524 // Remove it from the list of active tracks. 2525 tracksToRemove->add(track); 2526 } else { 2527 // No buffers for this track. Give it a few chances to 2528 // fill a buffer, then remove it from active list. 2529 if (--(track->mRetryCount) <= 0) { 2530 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2531 tracksToRemove->add(track); 2532 // indicate to client process that the track was disabled because of underrun 2533 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2534 // If one track is not ready, mark the mixer also not ready if: 2535 // - the mixer was ready during previous round OR 2536 // - no other track is ready 2537 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2538 mixerStatus != MIXER_TRACKS_READY) { 2539 mixerStatus = MIXER_TRACKS_ENABLED; 2540 } 2541 } 2542 mAudioMixer->disable(name); 2543 } 2544 } 2545 2546 // remove all the tracks that need to be... 2547 count = tracksToRemove->size(); 2548 if (CC_UNLIKELY(count)) { 2549 for (size_t i=0 ; i<count ; i++) { 2550 const sp<Track>& track = tracksToRemove->itemAt(i); 2551 mActiveTracks.remove(track); 2552 if (track->mainBuffer() != mMixBuffer) { 2553 chain = getEffectChain_l(track->sessionId()); 2554 if (chain != 0) { 2555 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2556 chain->decActiveTrackCnt(); 2557 } 2558 } 2559 if (track->isTerminated()) { 2560 removeTrack_l(track); 2561 } 2562 } 2563 } 2564 2565 // mix buffer must be cleared if all tracks are connected to an 2566 // effect chain as in this case the mixer will not write to 2567 // mix buffer and track effects will accumulate into it 2568 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2569 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2570 } 2571 2572 mPrevMixerStatus = mixerStatus; 2573 return mixerStatus; 2574} 2575 2576void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2577{ 2578 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2579 this, streamType, mTracks.size()); 2580 Mutex::Autolock _l(mLock); 2581 2582 size_t size = mTracks.size(); 2583 for (size_t i = 0; i < size; i++) { 2584 sp<Track> t = mTracks[i]; 2585 if (t->streamType() == streamType) { 2586 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2587 t->mCblk->cv.signal(); 2588 } 2589 } 2590} 2591 2592void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2593{ 2594 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2595 this, streamType, valid); 2596 Mutex::Autolock _l(mLock); 2597 2598 mStreamTypes[streamType].valid = valid; 2599} 2600 2601// getTrackName_l() must be called with ThreadBase::mLock held 2602int AudioFlinger::MixerThread::getTrackName_l() 2603{ 2604 return mAudioMixer->getTrackName(); 2605} 2606 2607// deleteTrackName_l() must be called with ThreadBase::mLock held 2608void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2609{ 2610 ALOGV("remove track (%d) and delete from mixer", name); 2611 mAudioMixer->deleteTrackName(name); 2612} 2613 2614// checkForNewParameters_l() must be called with ThreadBase::mLock held 2615bool AudioFlinger::MixerThread::checkForNewParameters_l() 2616{ 2617 bool reconfig = false; 2618 2619 while (!mNewParameters.isEmpty()) { 2620 status_t status = NO_ERROR; 2621 String8 keyValuePair = mNewParameters[0]; 2622 AudioParameter param = AudioParameter(keyValuePair); 2623 int value; 2624 2625 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2626 reconfig = true; 2627 } 2628 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2629 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2630 status = BAD_VALUE; 2631 } else { 2632 reconfig = true; 2633 } 2634 } 2635 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2636 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2637 status = BAD_VALUE; 2638 } else { 2639 reconfig = true; 2640 } 2641 } 2642 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2643 // do not accept frame count changes if tracks are open as the track buffer 2644 // size depends on frame count and correct behavior would not be guaranteed 2645 // if frame count is changed after track creation 2646 if (!mTracks.isEmpty()) { 2647 status = INVALID_OPERATION; 2648 } else { 2649 reconfig = true; 2650 } 2651 } 2652 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2653 // when changing the audio output device, call addBatteryData to notify 2654 // the change 2655 if ((int)mDevice != value) { 2656 uint32_t params = 0; 2657 // check whether speaker is on 2658 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2659 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2660 } 2661 2662 int deviceWithoutSpeaker 2663 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2664 // check if any other device (except speaker) is on 2665 if (value & deviceWithoutSpeaker ) { 2666 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2667 } 2668 2669 if (params != 0) { 2670 addBatteryData(params); 2671 } 2672 } 2673 2674 // forward device change to effects that have requested to be 2675 // aware of attached audio device. 2676 mDevice = (uint32_t)value; 2677 for (size_t i = 0; i < mEffectChains.size(); i++) { 2678 mEffectChains[i]->setDevice_l(mDevice); 2679 } 2680 } 2681 2682 if (status == NO_ERROR) { 2683 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2684 keyValuePair.string()); 2685 if (!mStandby && status == INVALID_OPERATION) { 2686 mOutput->stream->common.standby(&mOutput->stream->common); 2687 mStandby = true; 2688 mBytesWritten = 0; 2689 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2690 keyValuePair.string()); 2691 } 2692 if (status == NO_ERROR && reconfig) { 2693 delete mAudioMixer; 2694 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2695 mAudioMixer = NULL; 2696 readOutputParameters(); 2697 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2698 for (size_t i = 0; i < mTracks.size() ; i++) { 2699 int name = getTrackName_l(); 2700 if (name < 0) break; 2701 mTracks[i]->mName = name; 2702 // limit track sample rate to 2 x new output sample rate 2703 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2704 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2705 } 2706 } 2707 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2708 } 2709 } 2710 2711 mNewParameters.removeAt(0); 2712 2713 mParamStatus = status; 2714 mParamCond.signal(); 2715 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2716 // already timed out waiting for the status and will never signal the condition. 2717 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2718 } 2719 return reconfig; 2720} 2721 2722status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2723{ 2724 const size_t SIZE = 256; 2725 char buffer[SIZE]; 2726 String8 result; 2727 2728 PlaybackThread::dumpInternals(fd, args); 2729 2730 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2731 result.append(buffer); 2732 write(fd, result.string(), result.size()); 2733 return NO_ERROR; 2734} 2735 2736uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2737{ 2738 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2739} 2740 2741uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2742{ 2743 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2744} 2745 2746// ---------------------------------------------------------------------------- 2747AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2748 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2749 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2750 // mLeftVolFloat, mRightVolFloat 2751 // mLeftVolShort, mRightVolShort 2752{ 2753} 2754 2755AudioFlinger::DirectOutputThread::~DirectOutputThread() 2756{ 2757} 2758 2759void AudioFlinger::DirectOutputThread::applyVolume() 2760{ 2761 // Do not apply volume on compressed audio 2762 if (!audio_is_linear_pcm(mFormat)) { 2763 return; 2764 } 2765 2766 // convert to signed 16 bit before volume calculation 2767 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2768 size_t count = mFrameCount * mChannelCount; 2769 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2770 int16_t *dst = mMixBuffer + count-1; 2771 while(count--) { 2772 *dst-- = (int16_t)(*src--^0x80) << 8; 2773 } 2774 } 2775 2776 size_t frameCount = mFrameCount; 2777 int16_t *out = mMixBuffer; 2778 if (rampVolume) { 2779 if (mChannelCount == 1) { 2780 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2781 int32_t vlInc = d / (int32_t)frameCount; 2782 int32_t vl = ((int32_t)mLeftVolShort << 16); 2783 do { 2784 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2785 out++; 2786 vl += vlInc; 2787 } while (--frameCount); 2788 2789 } else { 2790 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2791 int32_t vlInc = d / (int32_t)frameCount; 2792 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2793 int32_t vrInc = d / (int32_t)frameCount; 2794 int32_t vl = ((int32_t)mLeftVolShort << 16); 2795 int32_t vr = ((int32_t)mRightVolShort << 16); 2796 do { 2797 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2798 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2799 out += 2; 2800 vl += vlInc; 2801 vr += vrInc; 2802 } while (--frameCount); 2803 } 2804 } else { 2805 if (mChannelCount == 1) { 2806 do { 2807 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2808 out++; 2809 } while (--frameCount); 2810 } else { 2811 do { 2812 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2813 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2814 out += 2; 2815 } while (--frameCount); 2816 } 2817 } 2818 2819 // convert back to unsigned 8 bit after volume calculation 2820 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2821 size_t count = mFrameCount * mChannelCount; 2822 int16_t *src = mMixBuffer; 2823 uint8_t *dst = (uint8_t *)mMixBuffer; 2824 while(count--) { 2825 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2826 } 2827 } 2828 2829 mLeftVolShort = leftVol; 2830 mRightVolShort = rightVol; 2831} 2832 2833AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2834 Vector< sp<Track> > *tracksToRemove 2835) 2836{ 2837 sp<Track> trackToRemove; 2838 2839 // FIXME Temporarily renamed to avoid confusion with the member "mixerStatus" 2840 mixer_state mixerStatus_ = MIXER_IDLE; 2841 2842 // find out which tracks need to be processed 2843 if (mActiveTracks.size() != 0) { 2844 sp<Track> t = mActiveTracks[0].promote(); 2845 // see FIXME in AudioFlinger.h, return MIXER_IDLE might also work 2846 if (t == 0) return MIXER_CONTINUE; 2847 //if (t == 0) continue; 2848 2849 Track* const track = t.get(); 2850 audio_track_cblk_t* cblk = track->cblk(); 2851 2852 // The first time a track is added we wait 2853 // for all its buffers to be filled before processing it 2854 if (cblk->framesReady() && track->isReady() && 2855 !track->isPaused() && !track->isTerminated()) 2856 { 2857 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2858 2859 if (track->mFillingUpStatus == Track::FS_FILLED) { 2860 track->mFillingUpStatus = Track::FS_ACTIVE; 2861 mLeftVolFloat = mRightVolFloat = 0; 2862 mLeftVolShort = mRightVolShort = 0; 2863 if (track->mState == TrackBase::RESUMING) { 2864 track->mState = TrackBase::ACTIVE; 2865 rampVolume = true; 2866 } 2867 } else if (cblk->server != 0) { 2868 // If the track is stopped before the first frame was mixed, 2869 // do not apply ramp 2870 rampVolume = true; 2871 } 2872 // compute volume for this track 2873 float left, right; 2874 if (track->isMuted() || mMasterMute || track->isPausing() || 2875 mStreamTypes[track->streamType()].mute) { 2876 left = right = 0; 2877 if (track->isPausing()) { 2878 track->setPaused(); 2879 } 2880 } else { 2881 float typeVolume = mStreamTypes[track->streamType()].volume; 2882 float v = mMasterVolume * typeVolume; 2883 uint32_t vlr = cblk->getVolumeLR(); 2884 float v_clamped = v * (vlr & 0xFFFF); 2885 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2886 left = v_clamped/MAX_GAIN; 2887 v_clamped = v * (vlr >> 16); 2888 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2889 right = v_clamped/MAX_GAIN; 2890 } 2891 2892 if (left != mLeftVolFloat || right != mRightVolFloat) { 2893 mLeftVolFloat = left; 2894 mRightVolFloat = right; 2895 2896 // If audio HAL implements volume control, 2897 // force software volume to nominal value 2898 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2899 left = 1.0f; 2900 right = 1.0f; 2901 } 2902 2903 // Convert volumes from float to 8.24 2904 uint32_t vl = (uint32_t)(left * (1 << 24)); 2905 uint32_t vr = (uint32_t)(right * (1 << 24)); 2906 2907 // Delegate volume control to effect in track effect chain if needed 2908 // only one effect chain can be present on DirectOutputThread, so if 2909 // there is one, the track is connected to it 2910 if (!mEffectChains.isEmpty()) { 2911 // Do not ramp volume if volume is controlled by effect 2912 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2913 rampVolume = false; 2914 } 2915 } 2916 2917 // Convert volumes from 8.24 to 4.12 format 2918 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2919 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2920 leftVol = (uint16_t)v_clamped; 2921 v_clamped = (vr + (1 << 11)) >> 12; 2922 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2923 rightVol = (uint16_t)v_clamped; 2924 } else { 2925 leftVol = mLeftVolShort; 2926 rightVol = mRightVolShort; 2927 rampVolume = false; 2928 } 2929 2930 // reset retry count 2931 track->mRetryCount = kMaxTrackRetriesDirect; 2932 activeTrack = t; 2933 mixerStatus_ = MIXER_TRACKS_READY; 2934 } else { 2935 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2936 if (track->isStopped()) { 2937 track->reset(); 2938 } 2939 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2940 // We have consumed all the buffers of this track. 2941 // Remove it from the list of active tracks. 2942 trackToRemove = track; 2943 } else { 2944 // No buffers for this track. Give it a few chances to 2945 // fill a buffer, then remove it from active list. 2946 if (--(track->mRetryCount) <= 0) { 2947 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2948 trackToRemove = track; 2949 } else { 2950 mixerStatus_ = MIXER_TRACKS_ENABLED; 2951 } 2952 } 2953 } 2954 } 2955 2956 // FIXME merge this with similar code for removing multiple tracks 2957 // remove all the tracks that need to be... 2958 if (CC_UNLIKELY(trackToRemove != 0)) { 2959 tracksToRemove->add(trackToRemove); 2960 mActiveTracks.remove(trackToRemove); 2961 if (!mEffectChains.isEmpty()) { 2962 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2963 trackToRemove->sessionId()); 2964 mEffectChains[0]->decActiveTrackCnt(); 2965 } 2966 if (trackToRemove->isTerminated()) { 2967 removeTrack_l(trackToRemove); 2968 } 2969 } 2970 2971 return mixerStatus_; 2972} 2973 2974void AudioFlinger::DirectOutputThread::threadLoop_mix() 2975{ 2976 AudioBufferProvider::Buffer buffer; 2977 size_t frameCount = mFrameCount; 2978 int8_t *curBuf = (int8_t *)mMixBuffer; 2979 // output audio to hardware 2980 while (frameCount) { 2981 buffer.frameCount = frameCount; 2982 activeTrack->getNextBuffer(&buffer); 2983 if (CC_UNLIKELY(buffer.raw == NULL)) { 2984 memset(curBuf, 0, frameCount * mFrameSize); 2985 break; 2986 } 2987 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2988 frameCount -= buffer.frameCount; 2989 curBuf += buffer.frameCount * mFrameSize; 2990 activeTrack->releaseBuffer(&buffer); 2991 } 2992 sleepTime = 0; 2993 standbyTime = systemTime() + standbyDelay; 2994 applyVolume(); 2995} 2996 2997void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 2998{ 2999 if (sleepTime == 0) { 3000 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3001 sleepTime = activeSleepTime; 3002 } else { 3003 sleepTime = idleSleepTime; 3004 } 3005 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 3006 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 3007 sleepTime = 0; 3008 } 3009} 3010 3011// getTrackName_l() must be called with ThreadBase::mLock held 3012int AudioFlinger::DirectOutputThread::getTrackName_l() 3013{ 3014 return 0; 3015} 3016 3017// deleteTrackName_l() must be called with ThreadBase::mLock held 3018void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 3019{ 3020} 3021 3022// checkForNewParameters_l() must be called with ThreadBase::mLock held 3023bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3024{ 3025 bool reconfig = false; 3026 3027 while (!mNewParameters.isEmpty()) { 3028 status_t status = NO_ERROR; 3029 String8 keyValuePair = mNewParameters[0]; 3030 AudioParameter param = AudioParameter(keyValuePair); 3031 int value; 3032 3033 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3034 // do not accept frame count changes if tracks are open as the track buffer 3035 // size depends on frame count and correct behavior would not be garantied 3036 // if frame count is changed after track creation 3037 if (!mTracks.isEmpty()) { 3038 status = INVALID_OPERATION; 3039 } else { 3040 reconfig = true; 3041 } 3042 } 3043 if (status == NO_ERROR) { 3044 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3045 keyValuePair.string()); 3046 if (!mStandby && status == INVALID_OPERATION) { 3047 mOutput->stream->common.standby(&mOutput->stream->common); 3048 mStandby = true; 3049 mBytesWritten = 0; 3050 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3051 keyValuePair.string()); 3052 } 3053 if (status == NO_ERROR && reconfig) { 3054 readOutputParameters(); 3055 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3056 } 3057 } 3058 3059 mNewParameters.removeAt(0); 3060 3061 mParamStatus = status; 3062 mParamCond.signal(); 3063 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3064 // already timed out waiting for the status and will never signal the condition. 3065 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3066 } 3067 return reconfig; 3068} 3069 3070uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3071{ 3072 uint32_t time; 3073 if (audio_is_linear_pcm(mFormat)) { 3074 time = PlaybackThread::activeSleepTimeUs(); 3075 } else { 3076 time = 10000; 3077 } 3078 return time; 3079} 3080 3081uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3082{ 3083 uint32_t time; 3084 if (audio_is_linear_pcm(mFormat)) { 3085 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3086 } else { 3087 time = 10000; 3088 } 3089 return time; 3090} 3091 3092uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3093{ 3094 uint32_t time; 3095 if (audio_is_linear_pcm(mFormat)) { 3096 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3097 } else { 3098 time = 10000; 3099 } 3100 return time; 3101} 3102 3103 3104// ---------------------------------------------------------------------------- 3105 3106AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3107 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3108 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3109 mWaitTimeMs(UINT_MAX) 3110{ 3111 addOutputTrack(mainThread); 3112} 3113 3114AudioFlinger::DuplicatingThread::~DuplicatingThread() 3115{ 3116 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3117 mOutputTracks[i]->destroy(); 3118 } 3119} 3120 3121void AudioFlinger::DuplicatingThread::threadLoop_mix() 3122{ 3123 // mix buffers... 3124 if (outputsReady(outputTracks)) { 3125 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3126 } else { 3127 memset(mMixBuffer, 0, mixBufferSize); 3128 } 3129 sleepTime = 0; 3130 writeFrames = mFrameCount; 3131} 3132 3133void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3134{ 3135 if (sleepTime == 0) { 3136 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3137 sleepTime = activeSleepTime; 3138 } else { 3139 sleepTime = idleSleepTime; 3140 } 3141 } else if (mBytesWritten != 0) { 3142 // flush remaining overflow buffers in output tracks 3143 for (size_t i = 0; i < outputTracks.size(); i++) { 3144 if (outputTracks[i]->isActive()) { 3145 sleepTime = 0; 3146 writeFrames = 0; 3147 memset(mMixBuffer, 0, mixBufferSize); 3148 break; 3149 } 3150 } 3151 } 3152} 3153 3154void AudioFlinger::DuplicatingThread::threadLoop_write() 3155{ 3156 standbyTime = systemTime() + mStandbyTimeInNsecs; 3157 for (size_t i = 0; i < outputTracks.size(); i++) { 3158 outputTracks[i]->write(mMixBuffer, writeFrames); 3159 } 3160 mBytesWritten += mixBufferSize; 3161} 3162 3163void AudioFlinger::DuplicatingThread::threadLoop_standby() 3164{ 3165 // DuplicatingThread implements standby by stopping all tracks 3166 for (size_t i = 0; i < outputTracks.size(); i++) { 3167 outputTracks[i]->stop(); 3168 } 3169} 3170 3171void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3172{ 3173 Mutex::Autolock _l(mLock); 3174 // FIXME explain this formula 3175 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3176 OutputTrack *outputTrack = new OutputTrack(thread, 3177 this, 3178 mSampleRate, 3179 mFormat, 3180 mChannelMask, 3181 frameCount); 3182 if (outputTrack->cblk() != NULL) { 3183 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3184 mOutputTracks.add(outputTrack); 3185 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3186 updateWaitTime_l(); 3187 } 3188} 3189 3190void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3191{ 3192 Mutex::Autolock _l(mLock); 3193 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3194 if (mOutputTracks[i]->thread() == thread) { 3195 mOutputTracks[i]->destroy(); 3196 mOutputTracks.removeAt(i); 3197 updateWaitTime_l(); 3198 return; 3199 } 3200 } 3201 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3202} 3203 3204// caller must hold mLock 3205void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3206{ 3207 mWaitTimeMs = UINT_MAX; 3208 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3209 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3210 if (strong != 0) { 3211 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3212 if (waitTimeMs < mWaitTimeMs) { 3213 mWaitTimeMs = waitTimeMs; 3214 } 3215 } 3216 } 3217} 3218 3219 3220bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3221{ 3222 for (size_t i = 0; i < outputTracks.size(); i++) { 3223 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3224 if (thread == 0) { 3225 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3226 return false; 3227 } 3228 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3229 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3230 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3231 return false; 3232 } 3233 } 3234 return true; 3235} 3236 3237uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3238{ 3239 return (mWaitTimeMs * 1000) / 2; 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 assert(mMediaTimeTransformValid); 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, keyValuePair.string()); 5205 } 5206 if (reconfig) { 5207 if (status == BAD_VALUE && 5208 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5209 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5210 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5211 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5212 (reqChannelCount < 3)) { 5213 status = NO_ERROR; 5214 } 5215 if (status == NO_ERROR) { 5216 readInputParameters(); 5217 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5218 } 5219 } 5220 } 5221 5222 mNewParameters.removeAt(0); 5223 5224 mParamStatus = status; 5225 mParamCond.signal(); 5226 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5227 // already timed out waiting for the status and will never signal the condition. 5228 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5229 } 5230 return reconfig; 5231} 5232 5233String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5234{ 5235 char *s; 5236 String8 out_s8 = String8(); 5237 5238 Mutex::Autolock _l(mLock); 5239 if (initCheck() != NO_ERROR) { 5240 return out_s8; 5241 } 5242 5243 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5244 out_s8 = String8(s); 5245 free(s); 5246 return out_s8; 5247} 5248 5249void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5250 AudioSystem::OutputDescriptor desc; 5251 void *param2 = NULL; 5252 5253 switch (event) { 5254 case AudioSystem::INPUT_OPENED: 5255 case AudioSystem::INPUT_CONFIG_CHANGED: 5256 desc.channels = mChannelMask; 5257 desc.samplingRate = mSampleRate; 5258 desc.format = mFormat; 5259 desc.frameCount = mFrameCount; 5260 desc.latency = 0; 5261 param2 = &desc; 5262 break; 5263 5264 case AudioSystem::INPUT_CLOSED: 5265 default: 5266 break; 5267 } 5268 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5269} 5270 5271void AudioFlinger::RecordThread::readInputParameters() 5272{ 5273 delete mRsmpInBuffer; 5274 // mRsmpInBuffer is always assigned a new[] below 5275 delete mRsmpOutBuffer; 5276 mRsmpOutBuffer = NULL; 5277 delete mResampler; 5278 mResampler = NULL; 5279 5280 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5281 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5282 mChannelCount = (uint16_t)popcount(mChannelMask); 5283 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5284 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5285 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5286 mFrameCount = mInputBytes / mFrameSize; 5287 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5288 5289 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5290 { 5291 int channelCount; 5292 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5293 // stereo to mono post process as the resampler always outputs stereo. 5294 if (mChannelCount == 1 && mReqChannelCount == 2) { 5295 channelCount = 1; 5296 } else { 5297 channelCount = 2; 5298 } 5299 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5300 mResampler->setSampleRate(mSampleRate); 5301 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5302 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5303 5304 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5305 if (mChannelCount == 1 && mReqChannelCount == 1) { 5306 mFrameCount >>= 1; 5307 } 5308 5309 } 5310 mRsmpInIndex = mFrameCount; 5311} 5312 5313unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5314{ 5315 Mutex::Autolock _l(mLock); 5316 if (initCheck() != NO_ERROR) { 5317 return 0; 5318 } 5319 5320 return mInput->stream->get_input_frames_lost(mInput->stream); 5321} 5322 5323uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5324{ 5325 Mutex::Autolock _l(mLock); 5326 uint32_t result = 0; 5327 if (getEffectChain_l(sessionId) != 0) { 5328 result = EFFECT_SESSION; 5329 } 5330 5331 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5332 result |= TRACK_SESSION; 5333 } 5334 5335 return result; 5336} 5337 5338AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5339{ 5340 Mutex::Autolock _l(mLock); 5341 return mTrack; 5342} 5343 5344AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5345{ 5346 Mutex::Autolock _l(mLock); 5347 return mInput; 5348} 5349 5350AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5351{ 5352 Mutex::Autolock _l(mLock); 5353 AudioStreamIn *input = mInput; 5354 mInput = NULL; 5355 return input; 5356} 5357 5358// this method must always be called either with ThreadBase mLock held or inside the thread loop 5359audio_stream_t* AudioFlinger::RecordThread::stream() 5360{ 5361 if (mInput == NULL) { 5362 return NULL; 5363 } 5364 return &mInput->stream->common; 5365} 5366 5367 5368// ---------------------------------------------------------------------------- 5369 5370audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5371 uint32_t *pSamplingRate, 5372 audio_format_t *pFormat, 5373 uint32_t *pChannels, 5374 uint32_t *pLatencyMs, 5375 uint32_t flags) 5376{ 5377 status_t status; 5378 PlaybackThread *thread = NULL; 5379 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5380 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5381 uint32_t channels = pChannels ? *pChannels : 0; 5382 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5383 audio_stream_out_t *outStream; 5384 audio_hw_device_t *outHwDev; 5385 5386 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5387 pDevices ? *pDevices : 0, 5388 samplingRate, 5389 format, 5390 channels, 5391 flags); 5392 5393 if (pDevices == NULL || *pDevices == 0) { 5394 return 0; 5395 } 5396 5397 Mutex::Autolock _l(mLock); 5398 5399 outHwDev = findSuitableHwDev_l(*pDevices); 5400 if (outHwDev == NULL) 5401 return 0; 5402 5403 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5404 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5405 &channels, &samplingRate, &outStream); 5406 mHardwareStatus = AUDIO_HW_IDLE; 5407 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5408 outStream, 5409 samplingRate, 5410 format, 5411 channels, 5412 status); 5413 5414 if (outStream != NULL) { 5415 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5416 audio_io_handle_t id = nextUniqueId(); 5417 5418 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5419 (format != AUDIO_FORMAT_PCM_16_BIT) || 5420 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5421 thread = new DirectOutputThread(this, output, id, *pDevices); 5422 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5423 } else { 5424 thread = new MixerThread(this, output, id, *pDevices); 5425 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5426 } 5427 mPlaybackThreads.add(id, thread); 5428 5429 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5430 if (pFormat != NULL) *pFormat = format; 5431 if (pChannels != NULL) *pChannels = channels; 5432 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5433 5434 // notify client processes of the new output creation 5435 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5436 return id; 5437 } 5438 5439 return 0; 5440} 5441 5442audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5443 audio_io_handle_t output2) 5444{ 5445 Mutex::Autolock _l(mLock); 5446 MixerThread *thread1 = checkMixerThread_l(output1); 5447 MixerThread *thread2 = checkMixerThread_l(output2); 5448 5449 if (thread1 == NULL || thread2 == NULL) { 5450 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5451 return 0; 5452 } 5453 5454 audio_io_handle_t id = nextUniqueId(); 5455 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5456 thread->addOutputTrack(thread2); 5457 mPlaybackThreads.add(id, thread); 5458 // notify client processes of the new output creation 5459 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5460 return id; 5461} 5462 5463status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5464{ 5465 // keep strong reference on the playback thread so that 5466 // it is not destroyed while exit() is executed 5467 sp <PlaybackThread> thread; 5468 { 5469 Mutex::Autolock _l(mLock); 5470 thread = checkPlaybackThread_l(output); 5471 if (thread == NULL) { 5472 return BAD_VALUE; 5473 } 5474 5475 ALOGV("closeOutput() %d", output); 5476 5477 if (thread->type() == ThreadBase::MIXER) { 5478 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5479 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5480 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5481 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5482 } 5483 } 5484 } 5485 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5486 mPlaybackThreads.removeItem(output); 5487 } 5488 thread->exit(); 5489 // The thread entity (active unit of execution) is no longer running here, 5490 // but the ThreadBase container still exists. 5491 5492 if (thread->type() != ThreadBase::DUPLICATING) { 5493 AudioStreamOut *out = thread->clearOutput(); 5494 assert(out != NULL); 5495 // from now on thread->mOutput is NULL 5496 out->hwDev->close_output_stream(out->hwDev, out->stream); 5497 delete out; 5498 } 5499 return NO_ERROR; 5500} 5501 5502status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5503{ 5504 Mutex::Autolock _l(mLock); 5505 PlaybackThread *thread = checkPlaybackThread_l(output); 5506 5507 if (thread == NULL) { 5508 return BAD_VALUE; 5509 } 5510 5511 ALOGV("suspendOutput() %d", output); 5512 thread->suspend(); 5513 5514 return NO_ERROR; 5515} 5516 5517status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5518{ 5519 Mutex::Autolock _l(mLock); 5520 PlaybackThread *thread = checkPlaybackThread_l(output); 5521 5522 if (thread == NULL) { 5523 return BAD_VALUE; 5524 } 5525 5526 ALOGV("restoreOutput() %d", output); 5527 5528 thread->restore(); 5529 5530 return NO_ERROR; 5531} 5532 5533audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5534 uint32_t *pSamplingRate, 5535 audio_format_t *pFormat, 5536 uint32_t *pChannels, 5537 audio_in_acoustics_t acoustics) 5538{ 5539 status_t status; 5540 RecordThread *thread = NULL; 5541 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5542 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5543 uint32_t channels = pChannels ? *pChannels : 0; 5544 uint32_t reqSamplingRate = samplingRate; 5545 audio_format_t reqFormat = format; 5546 uint32_t reqChannels = channels; 5547 audio_stream_in_t *inStream; 5548 audio_hw_device_t *inHwDev; 5549 5550 if (pDevices == NULL || *pDevices == 0) { 5551 return 0; 5552 } 5553 5554 Mutex::Autolock _l(mLock); 5555 5556 inHwDev = findSuitableHwDev_l(*pDevices); 5557 if (inHwDev == NULL) 5558 return 0; 5559 5560 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5561 &channels, &samplingRate, 5562 acoustics, 5563 &inStream); 5564 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5565 inStream, 5566 samplingRate, 5567 format, 5568 channels, 5569 acoustics, 5570 status); 5571 5572 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5573 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5574 // or stereo to mono conversions on 16 bit PCM inputs. 5575 if (inStream == NULL && status == BAD_VALUE && 5576 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5577 (samplingRate <= 2 * reqSamplingRate) && 5578 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5579 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5580 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5581 &channels, &samplingRate, 5582 acoustics, 5583 &inStream); 5584 } 5585 5586 if (inStream != NULL) { 5587 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5588 5589 audio_io_handle_t id = nextUniqueId(); 5590 // Start record thread 5591 // RecorThread require both input and output device indication to forward to audio 5592 // pre processing modules 5593 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5594 thread = new RecordThread(this, 5595 input, 5596 reqSamplingRate, 5597 reqChannels, 5598 id, 5599 device); 5600 mRecordThreads.add(id, thread); 5601 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5602 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5603 if (pFormat != NULL) *pFormat = format; 5604 if (pChannels != NULL) *pChannels = reqChannels; 5605 5606 input->stream->common.standby(&input->stream->common); 5607 5608 // notify client processes of the new input creation 5609 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5610 return id; 5611 } 5612 5613 return 0; 5614} 5615 5616status_t AudioFlinger::closeInput(audio_io_handle_t input) 5617{ 5618 // keep strong reference on the record thread so that 5619 // it is not destroyed while exit() is executed 5620 sp <RecordThread> thread; 5621 { 5622 Mutex::Autolock _l(mLock); 5623 thread = checkRecordThread_l(input); 5624 if (thread == NULL) { 5625 return BAD_VALUE; 5626 } 5627 5628 ALOGV("closeInput() %d", input); 5629 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5630 mRecordThreads.removeItem(input); 5631 } 5632 thread->exit(); 5633 // The thread entity (active unit of execution) is no longer running here, 5634 // but the ThreadBase container still exists. 5635 5636 AudioStreamIn *in = thread->clearInput(); 5637 assert(in != NULL); 5638 // from now on thread->mInput is NULL 5639 in->hwDev->close_input_stream(in->hwDev, in->stream); 5640 delete in; 5641 5642 return NO_ERROR; 5643} 5644 5645status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5646{ 5647 Mutex::Autolock _l(mLock); 5648 MixerThread *dstThread = checkMixerThread_l(output); 5649 if (dstThread == NULL) { 5650 ALOGW("setStreamOutput() bad output id %d", output); 5651 return BAD_VALUE; 5652 } 5653 5654 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5655 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5656 5657 dstThread->setStreamValid(stream, true); 5658 5659 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5660 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5661 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5662 MixerThread *srcThread = (MixerThread *)thread; 5663 srcThread->setStreamValid(stream, false); 5664 srcThread->invalidateTracks(stream); 5665 } 5666 } 5667 5668 return NO_ERROR; 5669} 5670 5671 5672int AudioFlinger::newAudioSessionId() 5673{ 5674 return nextUniqueId(); 5675} 5676 5677void AudioFlinger::acquireAudioSessionId(int audioSession) 5678{ 5679 Mutex::Autolock _l(mLock); 5680 pid_t caller = IPCThreadState::self()->getCallingPid(); 5681 ALOGV("acquiring %d from %d", audioSession, caller); 5682 size_t num = mAudioSessionRefs.size(); 5683 for (size_t i = 0; i< num; i++) { 5684 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5685 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5686 ref->mCnt++; 5687 ALOGV(" incremented refcount to %d", ref->mCnt); 5688 return; 5689 } 5690 } 5691 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5692 ALOGV(" added new entry for %d", audioSession); 5693} 5694 5695void AudioFlinger::releaseAudioSessionId(int audioSession) 5696{ 5697 Mutex::Autolock _l(mLock); 5698 pid_t caller = IPCThreadState::self()->getCallingPid(); 5699 ALOGV("releasing %d from %d", audioSession, caller); 5700 size_t num = mAudioSessionRefs.size(); 5701 for (size_t i = 0; i< num; i++) { 5702 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5703 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5704 ref->mCnt--; 5705 ALOGV(" decremented refcount to %d", ref->mCnt); 5706 if (ref->mCnt == 0) { 5707 mAudioSessionRefs.removeAt(i); 5708 delete ref; 5709 purgeStaleEffects_l(); 5710 } 5711 return; 5712 } 5713 } 5714 ALOGW("session id %d not found for pid %d", audioSession, caller); 5715} 5716 5717void AudioFlinger::purgeStaleEffects_l() { 5718 5719 ALOGV("purging stale effects"); 5720 5721 Vector< sp<EffectChain> > chains; 5722 5723 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5724 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5725 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5726 sp<EffectChain> ec = t->mEffectChains[j]; 5727 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5728 chains.push(ec); 5729 } 5730 } 5731 } 5732 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5733 sp<RecordThread> t = mRecordThreads.valueAt(i); 5734 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5735 sp<EffectChain> ec = t->mEffectChains[j]; 5736 chains.push(ec); 5737 } 5738 } 5739 5740 for (size_t i = 0; i < chains.size(); i++) { 5741 sp<EffectChain> ec = chains[i]; 5742 int sessionid = ec->sessionId(); 5743 sp<ThreadBase> t = ec->mThread.promote(); 5744 if (t == 0) { 5745 continue; 5746 } 5747 size_t numsessionrefs = mAudioSessionRefs.size(); 5748 bool found = false; 5749 for (size_t k = 0; k < numsessionrefs; k++) { 5750 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5751 if (ref->mSessionid == sessionid) { 5752 ALOGV(" session %d still exists for %d with %d refs", 5753 sessionid, ref->mPid, ref->mCnt); 5754 found = true; 5755 break; 5756 } 5757 } 5758 if (!found) { 5759 // remove all effects from the chain 5760 while (ec->mEffects.size()) { 5761 sp<EffectModule> effect = ec->mEffects[0]; 5762 effect->unPin(); 5763 Mutex::Autolock _l (t->mLock); 5764 t->removeEffect_l(effect); 5765 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5766 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5767 if (handle != 0) { 5768 handle->mEffect.clear(); 5769 if (handle->mHasControl && handle->mEnabled) { 5770 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5771 } 5772 } 5773 } 5774 AudioSystem::unregisterEffect(effect->id()); 5775 } 5776 } 5777 } 5778 return; 5779} 5780 5781// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5782AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5783{ 5784 return mPlaybackThreads.valueFor(output).get(); 5785} 5786 5787// checkMixerThread_l() must be called with AudioFlinger::mLock held 5788AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5789{ 5790 PlaybackThread *thread = checkPlaybackThread_l(output); 5791 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5792} 5793 5794// checkRecordThread_l() must be called with AudioFlinger::mLock held 5795AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5796{ 5797 return mRecordThreads.valueFor(input).get(); 5798} 5799 5800uint32_t AudioFlinger::nextUniqueId() 5801{ 5802 return android_atomic_inc(&mNextUniqueId); 5803} 5804 5805AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5806{ 5807 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5808 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5809 AudioStreamOut *output = thread->getOutput(); 5810 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5811 return thread; 5812 } 5813 } 5814 return NULL; 5815} 5816 5817uint32_t AudioFlinger::primaryOutputDevice_l() const 5818{ 5819 PlaybackThread *thread = primaryPlaybackThread_l(); 5820 5821 if (thread == NULL) { 5822 return 0; 5823 } 5824 5825 return thread->device(); 5826} 5827 5828 5829// ---------------------------------------------------------------------------- 5830// Effect management 5831// ---------------------------------------------------------------------------- 5832 5833 5834status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5835{ 5836 Mutex::Autolock _l(mLock); 5837 return EffectQueryNumberEffects(numEffects); 5838} 5839 5840status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5841{ 5842 Mutex::Autolock _l(mLock); 5843 return EffectQueryEffect(index, descriptor); 5844} 5845 5846status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5847 effect_descriptor_t *descriptor) const 5848{ 5849 Mutex::Autolock _l(mLock); 5850 return EffectGetDescriptor(pUuid, descriptor); 5851} 5852 5853 5854sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5855 effect_descriptor_t *pDesc, 5856 const sp<IEffectClient>& effectClient, 5857 int32_t priority, 5858 audio_io_handle_t io, 5859 int sessionId, 5860 status_t *status, 5861 int *id, 5862 int *enabled) 5863{ 5864 status_t lStatus = NO_ERROR; 5865 sp<EffectHandle> handle; 5866 effect_descriptor_t desc; 5867 5868 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5869 pid, effectClient.get(), priority, sessionId, io); 5870 5871 if (pDesc == NULL) { 5872 lStatus = BAD_VALUE; 5873 goto Exit; 5874 } 5875 5876 // check audio settings permission for global effects 5877 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5878 lStatus = PERMISSION_DENIED; 5879 goto Exit; 5880 } 5881 5882 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5883 // that can only be created by audio policy manager (running in same process) 5884 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5885 lStatus = PERMISSION_DENIED; 5886 goto Exit; 5887 } 5888 5889 if (io == 0) { 5890 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5891 // output must be specified by AudioPolicyManager when using session 5892 // AUDIO_SESSION_OUTPUT_STAGE 5893 lStatus = BAD_VALUE; 5894 goto Exit; 5895 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5896 // if the output returned by getOutputForEffect() is removed before we lock the 5897 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5898 // and we will exit safely 5899 io = AudioSystem::getOutputForEffect(&desc); 5900 } 5901 } 5902 5903 { 5904 Mutex::Autolock _l(mLock); 5905 5906 5907 if (!EffectIsNullUuid(&pDesc->uuid)) { 5908 // if uuid is specified, request effect descriptor 5909 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5910 if (lStatus < 0) { 5911 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5912 goto Exit; 5913 } 5914 } else { 5915 // if uuid is not specified, look for an available implementation 5916 // of the required type in effect factory 5917 if (EffectIsNullUuid(&pDesc->type)) { 5918 ALOGW("createEffect() no effect type"); 5919 lStatus = BAD_VALUE; 5920 goto Exit; 5921 } 5922 uint32_t numEffects = 0; 5923 effect_descriptor_t d; 5924 d.flags = 0; // prevent compiler warning 5925 bool found = false; 5926 5927 lStatus = EffectQueryNumberEffects(&numEffects); 5928 if (lStatus < 0) { 5929 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5930 goto Exit; 5931 } 5932 for (uint32_t i = 0; i < numEffects; i++) { 5933 lStatus = EffectQueryEffect(i, &desc); 5934 if (lStatus < 0) { 5935 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5936 continue; 5937 } 5938 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5939 // If matching type found save effect descriptor. If the session is 5940 // 0 and the effect is not auxiliary, continue enumeration in case 5941 // an auxiliary version of this effect type is available 5942 found = true; 5943 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5944 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5945 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5946 break; 5947 } 5948 } 5949 } 5950 if (!found) { 5951 lStatus = BAD_VALUE; 5952 ALOGW("createEffect() effect not found"); 5953 goto Exit; 5954 } 5955 // For same effect type, chose auxiliary version over insert version if 5956 // connect to output mix (Compliance to OpenSL ES) 5957 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5958 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5959 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5960 } 5961 } 5962 5963 // Do not allow auxiliary effects on a session different from 0 (output mix) 5964 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5965 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5966 lStatus = INVALID_OPERATION; 5967 goto Exit; 5968 } 5969 5970 // check recording permission for visualizer 5971 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5972 !recordingAllowed()) { 5973 lStatus = PERMISSION_DENIED; 5974 goto Exit; 5975 } 5976 5977 // return effect descriptor 5978 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5979 5980 // If output is not specified try to find a matching audio session ID in one of the 5981 // output threads. 5982 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5983 // because of code checking output when entering the function. 5984 // Note: io is never 0 when creating an effect on an input 5985 if (io == 0) { 5986 // look for the thread where the specified audio session is present 5987 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5988 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5989 io = mPlaybackThreads.keyAt(i); 5990 break; 5991 } 5992 } 5993 if (io == 0) { 5994 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5995 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5996 io = mRecordThreads.keyAt(i); 5997 break; 5998 } 5999 } 6000 } 6001 // If no output thread contains the requested session ID, default to 6002 // first output. The effect chain will be moved to the correct output 6003 // thread when a track with the same session ID is created 6004 if (io == 0 && mPlaybackThreads.size()) { 6005 io = mPlaybackThreads.keyAt(0); 6006 } 6007 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 6008 } 6009 ThreadBase *thread = checkRecordThread_l(io); 6010 if (thread == NULL) { 6011 thread = checkPlaybackThread_l(io); 6012 if (thread == NULL) { 6013 ALOGE("createEffect() unknown output thread"); 6014 lStatus = BAD_VALUE; 6015 goto Exit; 6016 } 6017 } 6018 6019 sp<Client> client = registerPid_l(pid); 6020 6021 // create effect on selected output thread 6022 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6023 &desc, enabled, &lStatus); 6024 if (handle != 0 && id != NULL) { 6025 *id = handle->id(); 6026 } 6027 } 6028 6029Exit: 6030 if(status) { 6031 *status = lStatus; 6032 } 6033 return handle; 6034} 6035 6036status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6037 audio_io_handle_t dstOutput) 6038{ 6039 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6040 sessionId, srcOutput, dstOutput); 6041 Mutex::Autolock _l(mLock); 6042 if (srcOutput == dstOutput) { 6043 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6044 return NO_ERROR; 6045 } 6046 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6047 if (srcThread == NULL) { 6048 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6049 return BAD_VALUE; 6050 } 6051 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6052 if (dstThread == NULL) { 6053 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6054 return BAD_VALUE; 6055 } 6056 6057 Mutex::Autolock _dl(dstThread->mLock); 6058 Mutex::Autolock _sl(srcThread->mLock); 6059 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6060 6061 return NO_ERROR; 6062} 6063 6064// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6065status_t AudioFlinger::moveEffectChain_l(int sessionId, 6066 AudioFlinger::PlaybackThread *srcThread, 6067 AudioFlinger::PlaybackThread *dstThread, 6068 bool reRegister) 6069{ 6070 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6071 sessionId, srcThread, dstThread); 6072 6073 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6074 if (chain == 0) { 6075 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6076 sessionId, srcThread); 6077 return INVALID_OPERATION; 6078 } 6079 6080 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6081 // so that a new chain is created with correct parameters when first effect is added. This is 6082 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6083 // removed. 6084 srcThread->removeEffectChain_l(chain); 6085 6086 // transfer all effects one by one so that new effect chain is created on new thread with 6087 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6088 audio_io_handle_t dstOutput = dstThread->id(); 6089 sp<EffectChain> dstChain; 6090 uint32_t strategy = 0; // prevent compiler warning 6091 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6092 while (effect != 0) { 6093 srcThread->removeEffect_l(effect); 6094 dstThread->addEffect_l(effect); 6095 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6096 if (effect->state() == EffectModule::ACTIVE || 6097 effect->state() == EffectModule::STOPPING) { 6098 effect->start(); 6099 } 6100 // if the move request is not received from audio policy manager, the effect must be 6101 // re-registered with the new strategy and output 6102 if (dstChain == 0) { 6103 dstChain = effect->chain().promote(); 6104 if (dstChain == 0) { 6105 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6106 srcThread->addEffect_l(effect); 6107 return NO_INIT; 6108 } 6109 strategy = dstChain->strategy(); 6110 } 6111 if (reRegister) { 6112 AudioSystem::unregisterEffect(effect->id()); 6113 AudioSystem::registerEffect(&effect->desc(), 6114 dstOutput, 6115 strategy, 6116 sessionId, 6117 effect->id()); 6118 } 6119 effect = chain->getEffectFromId_l(0); 6120 } 6121 6122 return NO_ERROR; 6123} 6124 6125 6126// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6127sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6128 const sp<AudioFlinger::Client>& client, 6129 const sp<IEffectClient>& effectClient, 6130 int32_t priority, 6131 int sessionId, 6132 effect_descriptor_t *desc, 6133 int *enabled, 6134 status_t *status 6135 ) 6136{ 6137 sp<EffectModule> effect; 6138 sp<EffectHandle> handle; 6139 status_t lStatus; 6140 sp<EffectChain> chain; 6141 bool chainCreated = false; 6142 bool effectCreated = false; 6143 bool effectRegistered = false; 6144 6145 lStatus = initCheck(); 6146 if (lStatus != NO_ERROR) { 6147 ALOGW("createEffect_l() Audio driver not initialized."); 6148 goto Exit; 6149 } 6150 6151 // Do not allow effects with session ID 0 on direct output or duplicating threads 6152 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6153 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6154 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6155 desc->name, sessionId); 6156 lStatus = BAD_VALUE; 6157 goto Exit; 6158 } 6159 // Only Pre processor effects are allowed on input threads and only on input threads 6160 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6161 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6162 desc->name, desc->flags, mType); 6163 lStatus = BAD_VALUE; 6164 goto Exit; 6165 } 6166 6167 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6168 6169 { // scope for mLock 6170 Mutex::Autolock _l(mLock); 6171 6172 // check for existing effect chain with the requested audio session 6173 chain = getEffectChain_l(sessionId); 6174 if (chain == 0) { 6175 // create a new chain for this session 6176 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6177 chain = new EffectChain(this, sessionId); 6178 addEffectChain_l(chain); 6179 chain->setStrategy(getStrategyForSession_l(sessionId)); 6180 chainCreated = true; 6181 } else { 6182 effect = chain->getEffectFromDesc_l(desc); 6183 } 6184 6185 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6186 6187 if (effect == 0) { 6188 int id = mAudioFlinger->nextUniqueId(); 6189 // Check CPU and memory usage 6190 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6191 if (lStatus != NO_ERROR) { 6192 goto Exit; 6193 } 6194 effectRegistered = true; 6195 // create a new effect module if none present in the chain 6196 effect = new EffectModule(this, chain, desc, id, sessionId); 6197 lStatus = effect->status(); 6198 if (lStatus != NO_ERROR) { 6199 goto Exit; 6200 } 6201 lStatus = chain->addEffect_l(effect); 6202 if (lStatus != NO_ERROR) { 6203 goto Exit; 6204 } 6205 effectCreated = true; 6206 6207 effect->setDevice(mDevice); 6208 effect->setMode(mAudioFlinger->getMode()); 6209 } 6210 // create effect handle and connect it to effect module 6211 handle = new EffectHandle(effect, client, effectClient, priority); 6212 lStatus = effect->addHandle(handle); 6213 if (enabled != NULL) { 6214 *enabled = (int)effect->isEnabled(); 6215 } 6216 } 6217 6218Exit: 6219 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6220 Mutex::Autolock _l(mLock); 6221 if (effectCreated) { 6222 chain->removeEffect_l(effect); 6223 } 6224 if (effectRegistered) { 6225 AudioSystem::unregisterEffect(effect->id()); 6226 } 6227 if (chainCreated) { 6228 removeEffectChain_l(chain); 6229 } 6230 handle.clear(); 6231 } 6232 6233 if(status) { 6234 *status = lStatus; 6235 } 6236 return handle; 6237} 6238 6239sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6240{ 6241 sp<EffectChain> chain = getEffectChain_l(sessionId); 6242 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6243} 6244 6245// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6246// PlaybackThread::mLock held 6247status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6248{ 6249 // check for existing effect chain with the requested audio session 6250 int sessionId = effect->sessionId(); 6251 sp<EffectChain> chain = getEffectChain_l(sessionId); 6252 bool chainCreated = false; 6253 6254 if (chain == 0) { 6255 // create a new chain for this session 6256 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6257 chain = new EffectChain(this, sessionId); 6258 addEffectChain_l(chain); 6259 chain->setStrategy(getStrategyForSession_l(sessionId)); 6260 chainCreated = true; 6261 } 6262 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6263 6264 if (chain->getEffectFromId_l(effect->id()) != 0) { 6265 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6266 this, effect->desc().name, chain.get()); 6267 return BAD_VALUE; 6268 } 6269 6270 status_t status = chain->addEffect_l(effect); 6271 if (status != NO_ERROR) { 6272 if (chainCreated) { 6273 removeEffectChain_l(chain); 6274 } 6275 return status; 6276 } 6277 6278 effect->setDevice(mDevice); 6279 effect->setMode(mAudioFlinger->getMode()); 6280 return NO_ERROR; 6281} 6282 6283void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6284 6285 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6286 effect_descriptor_t desc = effect->desc(); 6287 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6288 detachAuxEffect_l(effect->id()); 6289 } 6290 6291 sp<EffectChain> chain = effect->chain().promote(); 6292 if (chain != 0) { 6293 // remove effect chain if removing last effect 6294 if (chain->removeEffect_l(effect) == 0) { 6295 removeEffectChain_l(chain); 6296 } 6297 } else { 6298 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6299 } 6300} 6301 6302void AudioFlinger::ThreadBase::lockEffectChains_l( 6303 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6304{ 6305 effectChains = mEffectChains; 6306 for (size_t i = 0; i < mEffectChains.size(); i++) { 6307 mEffectChains[i]->lock(); 6308 } 6309} 6310 6311void AudioFlinger::ThreadBase::unlockEffectChains( 6312 const Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6313{ 6314 for (size_t i = 0; i < effectChains.size(); i++) { 6315 effectChains[i]->unlock(); 6316 } 6317} 6318 6319sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6320{ 6321 Mutex::Autolock _l(mLock); 6322 return getEffectChain_l(sessionId); 6323} 6324 6325sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6326{ 6327 size_t size = mEffectChains.size(); 6328 for (size_t i = 0; i < size; i++) { 6329 if (mEffectChains[i]->sessionId() == sessionId) { 6330 return mEffectChains[i]; 6331 } 6332 } 6333 return 0; 6334} 6335 6336void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6337{ 6338 Mutex::Autolock _l(mLock); 6339 size_t size = mEffectChains.size(); 6340 for (size_t i = 0; i < size; i++) { 6341 mEffectChains[i]->setMode_l(mode); 6342 } 6343} 6344 6345void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6346 const wp<EffectHandle>& handle, 6347 bool unpinIfLast) { 6348 6349 Mutex::Autolock _l(mLock); 6350 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6351 // delete the effect module if removing last handle on it 6352 if (effect->removeHandle(handle) == 0) { 6353 if (!effect->isPinned() || unpinIfLast) { 6354 removeEffect_l(effect); 6355 AudioSystem::unregisterEffect(effect->id()); 6356 } 6357 } 6358} 6359 6360status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6361{ 6362 int session = chain->sessionId(); 6363 int16_t *buffer = mMixBuffer; 6364 bool ownsBuffer = false; 6365 6366 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6367 if (session > 0) { 6368 // Only one effect chain can be present in direct output thread and it uses 6369 // the mix buffer as input 6370 if (mType != DIRECT) { 6371 size_t numSamples = mFrameCount * mChannelCount; 6372 buffer = new int16_t[numSamples]; 6373 memset(buffer, 0, numSamples * sizeof(int16_t)); 6374 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6375 ownsBuffer = true; 6376 } 6377 6378 // Attach all tracks with same session ID to this chain. 6379 for (size_t i = 0; i < mTracks.size(); ++i) { 6380 sp<Track> track = mTracks[i]; 6381 if (session == track->sessionId()) { 6382 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6383 track->setMainBuffer(buffer); 6384 chain->incTrackCnt(); 6385 } 6386 } 6387 6388 // indicate all active tracks in the chain 6389 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6390 sp<Track> track = mActiveTracks[i].promote(); 6391 if (track == 0) continue; 6392 if (session == track->sessionId()) { 6393 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6394 chain->incActiveTrackCnt(); 6395 } 6396 } 6397 } 6398 6399 chain->setInBuffer(buffer, ownsBuffer); 6400 chain->setOutBuffer(mMixBuffer); 6401 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6402 // chains list in order to be processed last as it contains output stage effects 6403 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6404 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6405 // after track specific effects and before output stage 6406 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6407 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6408 // Effect chain for other sessions are inserted at beginning of effect 6409 // chains list to be processed before output mix effects. Relative order between other 6410 // sessions is not important 6411 size_t size = mEffectChains.size(); 6412 size_t i = 0; 6413 for (i = 0; i < size; i++) { 6414 if (mEffectChains[i]->sessionId() < session) break; 6415 } 6416 mEffectChains.insertAt(chain, i); 6417 checkSuspendOnAddEffectChain_l(chain); 6418 6419 return NO_ERROR; 6420} 6421 6422size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6423{ 6424 int session = chain->sessionId(); 6425 6426 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6427 6428 for (size_t i = 0; i < mEffectChains.size(); i++) { 6429 if (chain == mEffectChains[i]) { 6430 mEffectChains.removeAt(i); 6431 // detach all active tracks from the chain 6432 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6433 sp<Track> track = mActiveTracks[i].promote(); 6434 if (track == 0) continue; 6435 if (session == track->sessionId()) { 6436 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6437 chain.get(), session); 6438 chain->decActiveTrackCnt(); 6439 } 6440 } 6441 6442 // detach all tracks with same session ID from this chain 6443 for (size_t i = 0; i < mTracks.size(); ++i) { 6444 sp<Track> track = mTracks[i]; 6445 if (session == track->sessionId()) { 6446 track->setMainBuffer(mMixBuffer); 6447 chain->decTrackCnt(); 6448 } 6449 } 6450 break; 6451 } 6452 } 6453 return mEffectChains.size(); 6454} 6455 6456status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6457 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6458{ 6459 Mutex::Autolock _l(mLock); 6460 return attachAuxEffect_l(track, EffectId); 6461} 6462 6463status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6464 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6465{ 6466 status_t status = NO_ERROR; 6467 6468 if (EffectId == 0) { 6469 track->setAuxBuffer(0, NULL); 6470 } else { 6471 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6472 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6473 if (effect != 0) { 6474 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6475 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6476 } else { 6477 status = INVALID_OPERATION; 6478 } 6479 } else { 6480 status = BAD_VALUE; 6481 } 6482 } 6483 return status; 6484} 6485 6486void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6487{ 6488 for (size_t i = 0; i < mTracks.size(); ++i) { 6489 sp<Track> track = mTracks[i]; 6490 if (track->auxEffectId() == effectId) { 6491 attachAuxEffect_l(track, 0); 6492 } 6493 } 6494} 6495 6496status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6497{ 6498 // only one chain per input thread 6499 if (mEffectChains.size() != 0) { 6500 return INVALID_OPERATION; 6501 } 6502 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6503 6504 chain->setInBuffer(NULL); 6505 chain->setOutBuffer(NULL); 6506 6507 checkSuspendOnAddEffectChain_l(chain); 6508 6509 mEffectChains.add(chain); 6510 6511 return NO_ERROR; 6512} 6513 6514size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6515{ 6516 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6517 ALOGW_IF(mEffectChains.size() != 1, 6518 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6519 chain.get(), mEffectChains.size(), this); 6520 if (mEffectChains.size() == 1) { 6521 mEffectChains.removeAt(0); 6522 } 6523 return 0; 6524} 6525 6526// ---------------------------------------------------------------------------- 6527// EffectModule implementation 6528// ---------------------------------------------------------------------------- 6529 6530#undef LOG_TAG 6531#define LOG_TAG "AudioFlinger::EffectModule" 6532 6533AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6534 const wp<AudioFlinger::EffectChain>& chain, 6535 effect_descriptor_t *desc, 6536 int id, 6537 int sessionId) 6538 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6539 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6540{ 6541 ALOGV("Constructor %p", this); 6542 int lStatus; 6543 if (thread == NULL) { 6544 return; 6545 } 6546 6547 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6548 6549 // create effect engine from effect factory 6550 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6551 6552 if (mStatus != NO_ERROR) { 6553 return; 6554 } 6555 lStatus = init(); 6556 if (lStatus < 0) { 6557 mStatus = lStatus; 6558 goto Error; 6559 } 6560 6561 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6562 mPinned = true; 6563 } 6564 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6565 return; 6566Error: 6567 EffectRelease(mEffectInterface); 6568 mEffectInterface = NULL; 6569 ALOGV("Constructor Error %d", mStatus); 6570} 6571 6572AudioFlinger::EffectModule::~EffectModule() 6573{ 6574 ALOGV("Destructor %p", this); 6575 if (mEffectInterface != NULL) { 6576 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6577 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6578 sp<ThreadBase> thread = mThread.promote(); 6579 if (thread != 0) { 6580 audio_stream_t *stream = thread->stream(); 6581 if (stream != NULL) { 6582 stream->remove_audio_effect(stream, mEffectInterface); 6583 } 6584 } 6585 } 6586 // release effect engine 6587 EffectRelease(mEffectInterface); 6588 } 6589} 6590 6591status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6592{ 6593 status_t status; 6594 6595 Mutex::Autolock _l(mLock); 6596 int priority = handle->priority(); 6597 size_t size = mHandles.size(); 6598 sp<EffectHandle> h; 6599 size_t i; 6600 for (i = 0; i < size; i++) { 6601 h = mHandles[i].promote(); 6602 if (h == 0) continue; 6603 if (h->priority() <= priority) break; 6604 } 6605 // if inserted in first place, move effect control from previous owner to this handle 6606 if (i == 0) { 6607 bool enabled = false; 6608 if (h != 0) { 6609 enabled = h->enabled(); 6610 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6611 } 6612 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6613 status = NO_ERROR; 6614 } else { 6615 status = ALREADY_EXISTS; 6616 } 6617 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6618 mHandles.insertAt(handle, i); 6619 return status; 6620} 6621 6622size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6623{ 6624 Mutex::Autolock _l(mLock); 6625 size_t size = mHandles.size(); 6626 size_t i; 6627 for (i = 0; i < size; i++) { 6628 if (mHandles[i] == handle) break; 6629 } 6630 if (i == size) { 6631 return size; 6632 } 6633 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6634 6635 bool enabled = false; 6636 EffectHandle *hdl = handle.unsafe_get(); 6637 if (hdl != NULL) { 6638 ALOGV("removeHandle() unsafe_get OK"); 6639 enabled = hdl->enabled(); 6640 } 6641 mHandles.removeAt(i); 6642 size = mHandles.size(); 6643 // if removed from first place, move effect control from this handle to next in line 6644 if (i == 0 && size != 0) { 6645 sp<EffectHandle> h = mHandles[0].promote(); 6646 if (h != 0) { 6647 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6648 } 6649 } 6650 6651 // Prevent calls to process() and other functions on effect interface from now on. 6652 // The effect engine will be released by the destructor when the last strong reference on 6653 // this object is released which can happen after next process is called. 6654 if (size == 0 && !mPinned) { 6655 mState = DESTROYED; 6656 } 6657 6658 return size; 6659} 6660 6661sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6662{ 6663 Mutex::Autolock _l(mLock); 6664 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6665} 6666 6667void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6668{ 6669 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6670 // keep a strong reference on this EffectModule to avoid calling the 6671 // destructor before we exit 6672 sp<EffectModule> keep(this); 6673 { 6674 sp<ThreadBase> thread = mThread.promote(); 6675 if (thread != 0) { 6676 thread->disconnectEffect(keep, handle, unpinIfLast); 6677 } 6678 } 6679} 6680 6681void AudioFlinger::EffectModule::updateState() { 6682 Mutex::Autolock _l(mLock); 6683 6684 switch (mState) { 6685 case RESTART: 6686 reset_l(); 6687 // FALL THROUGH 6688 6689 case STARTING: 6690 // clear auxiliary effect input buffer for next accumulation 6691 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6692 memset(mConfig.inputCfg.buffer.raw, 6693 0, 6694 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6695 } 6696 start_l(); 6697 mState = ACTIVE; 6698 break; 6699 case STOPPING: 6700 stop_l(); 6701 mDisableWaitCnt = mMaxDisableWaitCnt; 6702 mState = STOPPED; 6703 break; 6704 case STOPPED: 6705 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6706 // turn off sequence. 6707 if (--mDisableWaitCnt == 0) { 6708 reset_l(); 6709 mState = IDLE; 6710 } 6711 break; 6712 default: //IDLE , ACTIVE, DESTROYED 6713 break; 6714 } 6715} 6716 6717void AudioFlinger::EffectModule::process() 6718{ 6719 Mutex::Autolock _l(mLock); 6720 6721 if (mState == DESTROYED || mEffectInterface == NULL || 6722 mConfig.inputCfg.buffer.raw == NULL || 6723 mConfig.outputCfg.buffer.raw == NULL) { 6724 return; 6725 } 6726 6727 if (isProcessEnabled()) { 6728 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6729 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6730 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6731 mConfig.inputCfg.buffer.s32, 6732 mConfig.inputCfg.buffer.frameCount/2); 6733 } 6734 6735 // do the actual processing in the effect engine 6736 int ret = (*mEffectInterface)->process(mEffectInterface, 6737 &mConfig.inputCfg.buffer, 6738 &mConfig.outputCfg.buffer); 6739 6740 // force transition to IDLE state when engine is ready 6741 if (mState == STOPPED && ret == -ENODATA) { 6742 mDisableWaitCnt = 1; 6743 } 6744 6745 // clear auxiliary effect input buffer for next accumulation 6746 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6747 memset(mConfig.inputCfg.buffer.raw, 0, 6748 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6749 } 6750 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6751 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6752 // If an insert effect is idle and input buffer is different from output buffer, 6753 // accumulate input onto output 6754 sp<EffectChain> chain = mChain.promote(); 6755 if (chain != 0 && chain->activeTrackCnt() != 0) { 6756 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6757 int16_t *in = mConfig.inputCfg.buffer.s16; 6758 int16_t *out = mConfig.outputCfg.buffer.s16; 6759 for (size_t i = 0; i < frameCnt; i++) { 6760 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6761 } 6762 } 6763 } 6764} 6765 6766void AudioFlinger::EffectModule::reset_l() 6767{ 6768 if (mEffectInterface == NULL) { 6769 return; 6770 } 6771 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6772} 6773 6774status_t AudioFlinger::EffectModule::configure() 6775{ 6776 uint32_t channels; 6777 if (mEffectInterface == NULL) { 6778 return NO_INIT; 6779 } 6780 6781 sp<ThreadBase> thread = mThread.promote(); 6782 if (thread == 0) { 6783 return DEAD_OBJECT; 6784 } 6785 6786 // TODO: handle configuration of effects replacing track process 6787 if (thread->channelCount() == 1) { 6788 channels = AUDIO_CHANNEL_OUT_MONO; 6789 } else { 6790 channels = AUDIO_CHANNEL_OUT_STEREO; 6791 } 6792 6793 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6794 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6795 } else { 6796 mConfig.inputCfg.channels = channels; 6797 } 6798 mConfig.outputCfg.channels = channels; 6799 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6800 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6801 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6802 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6803 mConfig.inputCfg.bufferProvider.cookie = NULL; 6804 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6805 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6806 mConfig.outputCfg.bufferProvider.cookie = NULL; 6807 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6808 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6809 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6810 // Insert effect: 6811 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6812 // always overwrites output buffer: input buffer == output buffer 6813 // - in other sessions: 6814 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6815 // other effect: overwrites output buffer: input buffer == output buffer 6816 // Auxiliary effect: 6817 // accumulates in output buffer: input buffer != output buffer 6818 // Therefore: accumulate <=> input buffer != output buffer 6819 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6820 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6821 } else { 6822 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6823 } 6824 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6825 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6826 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6827 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6828 6829 ALOGV("configure() %p thread %p buffer %p framecount %d", 6830 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6831 6832 status_t cmdStatus; 6833 uint32_t size = sizeof(int); 6834 status_t status = (*mEffectInterface)->command(mEffectInterface, 6835 EFFECT_CMD_SET_CONFIG, 6836 sizeof(effect_config_t), 6837 &mConfig, 6838 &size, 6839 &cmdStatus); 6840 if (status == 0) { 6841 status = cmdStatus; 6842 } 6843 6844 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6845 (1000 * mConfig.outputCfg.buffer.frameCount); 6846 6847 return status; 6848} 6849 6850status_t AudioFlinger::EffectModule::init() 6851{ 6852 Mutex::Autolock _l(mLock); 6853 if (mEffectInterface == NULL) { 6854 return NO_INIT; 6855 } 6856 status_t cmdStatus; 6857 uint32_t size = sizeof(status_t); 6858 status_t status = (*mEffectInterface)->command(mEffectInterface, 6859 EFFECT_CMD_INIT, 6860 0, 6861 NULL, 6862 &size, 6863 &cmdStatus); 6864 if (status == 0) { 6865 status = cmdStatus; 6866 } 6867 return status; 6868} 6869 6870status_t AudioFlinger::EffectModule::start() 6871{ 6872 Mutex::Autolock _l(mLock); 6873 return start_l(); 6874} 6875 6876status_t AudioFlinger::EffectModule::start_l() 6877{ 6878 if (mEffectInterface == NULL) { 6879 return NO_INIT; 6880 } 6881 status_t cmdStatus; 6882 uint32_t size = sizeof(status_t); 6883 status_t status = (*mEffectInterface)->command(mEffectInterface, 6884 EFFECT_CMD_ENABLE, 6885 0, 6886 NULL, 6887 &size, 6888 &cmdStatus); 6889 if (status == 0) { 6890 status = cmdStatus; 6891 } 6892 if (status == 0 && 6893 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6894 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6895 sp<ThreadBase> thread = mThread.promote(); 6896 if (thread != 0) { 6897 audio_stream_t *stream = thread->stream(); 6898 if (stream != NULL) { 6899 stream->add_audio_effect(stream, mEffectInterface); 6900 } 6901 } 6902 } 6903 return status; 6904} 6905 6906status_t AudioFlinger::EffectModule::stop() 6907{ 6908 Mutex::Autolock _l(mLock); 6909 return stop_l(); 6910} 6911 6912status_t AudioFlinger::EffectModule::stop_l() 6913{ 6914 if (mEffectInterface == NULL) { 6915 return NO_INIT; 6916 } 6917 status_t cmdStatus; 6918 uint32_t size = sizeof(status_t); 6919 status_t status = (*mEffectInterface)->command(mEffectInterface, 6920 EFFECT_CMD_DISABLE, 6921 0, 6922 NULL, 6923 &size, 6924 &cmdStatus); 6925 if (status == 0) { 6926 status = cmdStatus; 6927 } 6928 if (status == 0 && 6929 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6930 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6931 sp<ThreadBase> thread = mThread.promote(); 6932 if (thread != 0) { 6933 audio_stream_t *stream = thread->stream(); 6934 if (stream != NULL) { 6935 stream->remove_audio_effect(stream, mEffectInterface); 6936 } 6937 } 6938 } 6939 return status; 6940} 6941 6942status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6943 uint32_t cmdSize, 6944 void *pCmdData, 6945 uint32_t *replySize, 6946 void *pReplyData) 6947{ 6948 Mutex::Autolock _l(mLock); 6949// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6950 6951 if (mState == DESTROYED || mEffectInterface == NULL) { 6952 return NO_INIT; 6953 } 6954 status_t status = (*mEffectInterface)->command(mEffectInterface, 6955 cmdCode, 6956 cmdSize, 6957 pCmdData, 6958 replySize, 6959 pReplyData); 6960 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6961 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6962 for (size_t i = 1; i < mHandles.size(); i++) { 6963 sp<EffectHandle> h = mHandles[i].promote(); 6964 if (h != 0) { 6965 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6966 } 6967 } 6968 } 6969 return status; 6970} 6971 6972status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6973{ 6974 6975 Mutex::Autolock _l(mLock); 6976 ALOGV("setEnabled %p enabled %d", this, enabled); 6977 6978 if (enabled != isEnabled()) { 6979 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6980 if (enabled && status != NO_ERROR) { 6981 return status; 6982 } 6983 6984 switch (mState) { 6985 // going from disabled to enabled 6986 case IDLE: 6987 mState = STARTING; 6988 break; 6989 case STOPPED: 6990 mState = RESTART; 6991 break; 6992 case STOPPING: 6993 mState = ACTIVE; 6994 break; 6995 6996 // going from enabled to disabled 6997 case RESTART: 6998 mState = STOPPED; 6999 break; 7000 case STARTING: 7001 mState = IDLE; 7002 break; 7003 case ACTIVE: 7004 mState = STOPPING; 7005 break; 7006 case DESTROYED: 7007 return NO_ERROR; // simply ignore as we are being destroyed 7008 } 7009 for (size_t i = 1; i < mHandles.size(); i++) { 7010 sp<EffectHandle> h = mHandles[i].promote(); 7011 if (h != 0) { 7012 h->setEnabled(enabled); 7013 } 7014 } 7015 } 7016 return NO_ERROR; 7017} 7018 7019bool AudioFlinger::EffectModule::isEnabled() const 7020{ 7021 switch (mState) { 7022 case RESTART: 7023 case STARTING: 7024 case ACTIVE: 7025 return true; 7026 case IDLE: 7027 case STOPPING: 7028 case STOPPED: 7029 case DESTROYED: 7030 default: 7031 return false; 7032 } 7033} 7034 7035bool AudioFlinger::EffectModule::isProcessEnabled() const 7036{ 7037 switch (mState) { 7038 case RESTART: 7039 case ACTIVE: 7040 case STOPPING: 7041 case STOPPED: 7042 return true; 7043 case IDLE: 7044 case STARTING: 7045 case DESTROYED: 7046 default: 7047 return false; 7048 } 7049} 7050 7051status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7052{ 7053 Mutex::Autolock _l(mLock); 7054 status_t status = NO_ERROR; 7055 7056 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7057 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7058 if (isProcessEnabled() && 7059 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7060 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7061 status_t cmdStatus; 7062 uint32_t volume[2]; 7063 uint32_t *pVolume = NULL; 7064 uint32_t size = sizeof(volume); 7065 volume[0] = *left; 7066 volume[1] = *right; 7067 if (controller) { 7068 pVolume = volume; 7069 } 7070 status = (*mEffectInterface)->command(mEffectInterface, 7071 EFFECT_CMD_SET_VOLUME, 7072 size, 7073 volume, 7074 &size, 7075 pVolume); 7076 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7077 *left = volume[0]; 7078 *right = volume[1]; 7079 } 7080 } 7081 return status; 7082} 7083 7084status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7085{ 7086 Mutex::Autolock _l(mLock); 7087 status_t status = NO_ERROR; 7088 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7089 // audio pre processing modules on RecordThread can receive both output and 7090 // input device indication in the same call 7091 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7092 if (dev) { 7093 status_t cmdStatus; 7094 uint32_t size = sizeof(status_t); 7095 7096 status = (*mEffectInterface)->command(mEffectInterface, 7097 EFFECT_CMD_SET_DEVICE, 7098 sizeof(uint32_t), 7099 &dev, 7100 &size, 7101 &cmdStatus); 7102 if (status == NO_ERROR) { 7103 status = cmdStatus; 7104 } 7105 } 7106 dev = device & AUDIO_DEVICE_IN_ALL; 7107 if (dev) { 7108 status_t cmdStatus; 7109 uint32_t size = sizeof(status_t); 7110 7111 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7112 EFFECT_CMD_SET_INPUT_DEVICE, 7113 sizeof(uint32_t), 7114 &dev, 7115 &size, 7116 &cmdStatus); 7117 if (status2 == NO_ERROR) { 7118 status2 = cmdStatus; 7119 } 7120 if (status == NO_ERROR) { 7121 status = status2; 7122 } 7123 } 7124 } 7125 return status; 7126} 7127 7128status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7129{ 7130 Mutex::Autolock _l(mLock); 7131 status_t status = NO_ERROR; 7132 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7133 status_t cmdStatus; 7134 uint32_t size = sizeof(status_t); 7135 status = (*mEffectInterface)->command(mEffectInterface, 7136 EFFECT_CMD_SET_AUDIO_MODE, 7137 sizeof(audio_mode_t), 7138 &mode, 7139 &size, 7140 &cmdStatus); 7141 if (status == NO_ERROR) { 7142 status = cmdStatus; 7143 } 7144 } 7145 return status; 7146} 7147 7148void AudioFlinger::EffectModule::setSuspended(bool suspended) 7149{ 7150 Mutex::Autolock _l(mLock); 7151 mSuspended = suspended; 7152} 7153 7154bool AudioFlinger::EffectModule::suspended() const 7155{ 7156 Mutex::Autolock _l(mLock); 7157 return mSuspended; 7158} 7159 7160status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7161{ 7162 const size_t SIZE = 256; 7163 char buffer[SIZE]; 7164 String8 result; 7165 7166 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7167 result.append(buffer); 7168 7169 bool locked = tryLock(mLock); 7170 // failed to lock - AudioFlinger is probably deadlocked 7171 if (!locked) { 7172 result.append("\t\tCould not lock Fx mutex:\n"); 7173 } 7174 7175 result.append("\t\tSession Status State Engine:\n"); 7176 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7177 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7178 result.append(buffer); 7179 7180 result.append("\t\tDescriptor:\n"); 7181 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7182 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7183 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7184 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7185 result.append(buffer); 7186 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7187 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7188 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7189 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7190 result.append(buffer); 7191 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7192 mDescriptor.apiVersion, 7193 mDescriptor.flags); 7194 result.append(buffer); 7195 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7196 mDescriptor.name); 7197 result.append(buffer); 7198 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7199 mDescriptor.implementor); 7200 result.append(buffer); 7201 7202 result.append("\t\t- Input configuration:\n"); 7203 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7204 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7205 (uint32_t)mConfig.inputCfg.buffer.raw, 7206 mConfig.inputCfg.buffer.frameCount, 7207 mConfig.inputCfg.samplingRate, 7208 mConfig.inputCfg.channels, 7209 mConfig.inputCfg.format); 7210 result.append(buffer); 7211 7212 result.append("\t\t- Output configuration:\n"); 7213 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7214 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7215 (uint32_t)mConfig.outputCfg.buffer.raw, 7216 mConfig.outputCfg.buffer.frameCount, 7217 mConfig.outputCfg.samplingRate, 7218 mConfig.outputCfg.channels, 7219 mConfig.outputCfg.format); 7220 result.append(buffer); 7221 7222 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7223 result.append(buffer); 7224 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7225 for (size_t i = 0; i < mHandles.size(); ++i) { 7226 sp<EffectHandle> handle = mHandles[i].promote(); 7227 if (handle != 0) { 7228 handle->dump(buffer, SIZE); 7229 result.append(buffer); 7230 } 7231 } 7232 7233 result.append("\n"); 7234 7235 write(fd, result.string(), result.length()); 7236 7237 if (locked) { 7238 mLock.unlock(); 7239 } 7240 7241 return NO_ERROR; 7242} 7243 7244// ---------------------------------------------------------------------------- 7245// EffectHandle implementation 7246// ---------------------------------------------------------------------------- 7247 7248#undef LOG_TAG 7249#define LOG_TAG "AudioFlinger::EffectHandle" 7250 7251AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7252 const sp<AudioFlinger::Client>& client, 7253 const sp<IEffectClient>& effectClient, 7254 int32_t priority) 7255 : BnEffect(), 7256 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7257 mPriority(priority), mHasControl(false), mEnabled(false) 7258{ 7259 ALOGV("constructor %p", this); 7260 7261 if (client == 0) { 7262 return; 7263 } 7264 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7265 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7266 if (mCblkMemory != 0) { 7267 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7268 7269 if (mCblk != NULL) { 7270 new(mCblk) effect_param_cblk_t(); 7271 mBuffer = (uint8_t *)mCblk + bufOffset; 7272 } 7273 } else { 7274 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7275 return; 7276 } 7277} 7278 7279AudioFlinger::EffectHandle::~EffectHandle() 7280{ 7281 ALOGV("Destructor %p", this); 7282 disconnect(false); 7283 ALOGV("Destructor DONE %p", this); 7284} 7285 7286status_t AudioFlinger::EffectHandle::enable() 7287{ 7288 ALOGV("enable %p", this); 7289 if (!mHasControl) return INVALID_OPERATION; 7290 if (mEffect == 0) return DEAD_OBJECT; 7291 7292 if (mEnabled) { 7293 return NO_ERROR; 7294 } 7295 7296 mEnabled = true; 7297 7298 sp<ThreadBase> thread = mEffect->thread().promote(); 7299 if (thread != 0) { 7300 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7301 } 7302 7303 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7304 if (mEffect->suspended()) { 7305 return NO_ERROR; 7306 } 7307 7308 status_t status = mEffect->setEnabled(true); 7309 if (status != NO_ERROR) { 7310 if (thread != 0) { 7311 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7312 } 7313 mEnabled = false; 7314 } 7315 return status; 7316} 7317 7318status_t AudioFlinger::EffectHandle::disable() 7319{ 7320 ALOGV("disable %p", this); 7321 if (!mHasControl) return INVALID_OPERATION; 7322 if (mEffect == 0) return DEAD_OBJECT; 7323 7324 if (!mEnabled) { 7325 return NO_ERROR; 7326 } 7327 mEnabled = false; 7328 7329 if (mEffect->suspended()) { 7330 return NO_ERROR; 7331 } 7332 7333 status_t status = mEffect->setEnabled(false); 7334 7335 sp<ThreadBase> thread = mEffect->thread().promote(); 7336 if (thread != 0) { 7337 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7338 } 7339 7340 return status; 7341} 7342 7343void AudioFlinger::EffectHandle::disconnect() 7344{ 7345 disconnect(true); 7346} 7347 7348void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7349{ 7350 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7351 if (mEffect == 0) { 7352 return; 7353 } 7354 mEffect->disconnect(this, unpinIfLast); 7355 7356 if (mHasControl && mEnabled) { 7357 sp<ThreadBase> thread = mEffect->thread().promote(); 7358 if (thread != 0) { 7359 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7360 } 7361 } 7362 7363 // release sp on module => module destructor can be called now 7364 mEffect.clear(); 7365 if (mClient != 0) { 7366 if (mCblk != NULL) { 7367 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7368 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7369 } 7370 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7371 // Client destructor must run with AudioFlinger mutex locked 7372 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7373 mClient.clear(); 7374 } 7375} 7376 7377status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7378 uint32_t cmdSize, 7379 void *pCmdData, 7380 uint32_t *replySize, 7381 void *pReplyData) 7382{ 7383// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7384// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7385 7386 // only get parameter command is permitted for applications not controlling the effect 7387 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7388 return INVALID_OPERATION; 7389 } 7390 if (mEffect == 0) return DEAD_OBJECT; 7391 if (mClient == 0) return INVALID_OPERATION; 7392 7393 // handle commands that are not forwarded transparently to effect engine 7394 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7395 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7396 // no risk to block the whole media server process or mixer threads is we are stuck here 7397 Mutex::Autolock _l(mCblk->lock); 7398 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7399 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7400 mCblk->serverIndex = 0; 7401 mCblk->clientIndex = 0; 7402 return BAD_VALUE; 7403 } 7404 status_t status = NO_ERROR; 7405 while (mCblk->serverIndex < mCblk->clientIndex) { 7406 int reply; 7407 uint32_t rsize = sizeof(int); 7408 int *p = (int *)(mBuffer + mCblk->serverIndex); 7409 int size = *p++; 7410 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7411 ALOGW("command(): invalid parameter block size"); 7412 break; 7413 } 7414 effect_param_t *param = (effect_param_t *)p; 7415 if (param->psize == 0 || param->vsize == 0) { 7416 ALOGW("command(): null parameter or value size"); 7417 mCblk->serverIndex += size; 7418 continue; 7419 } 7420 uint32_t psize = sizeof(effect_param_t) + 7421 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7422 param->vsize; 7423 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7424 psize, 7425 p, 7426 &rsize, 7427 &reply); 7428 // stop at first error encountered 7429 if (ret != NO_ERROR) { 7430 status = ret; 7431 *(int *)pReplyData = reply; 7432 break; 7433 } else if (reply != NO_ERROR) { 7434 *(int *)pReplyData = reply; 7435 break; 7436 } 7437 mCblk->serverIndex += size; 7438 } 7439 mCblk->serverIndex = 0; 7440 mCblk->clientIndex = 0; 7441 return status; 7442 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7443 *(int *)pReplyData = NO_ERROR; 7444 return enable(); 7445 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7446 *(int *)pReplyData = NO_ERROR; 7447 return disable(); 7448 } 7449 7450 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7451} 7452 7453void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7454{ 7455 ALOGV("setControl %p control %d", this, hasControl); 7456 7457 mHasControl = hasControl; 7458 mEnabled = enabled; 7459 7460 if (signal && mEffectClient != 0) { 7461 mEffectClient->controlStatusChanged(hasControl); 7462 } 7463} 7464 7465void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7466 uint32_t cmdSize, 7467 void *pCmdData, 7468 uint32_t replySize, 7469 void *pReplyData) 7470{ 7471 if (mEffectClient != 0) { 7472 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7473 } 7474} 7475 7476 7477 7478void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7479{ 7480 if (mEffectClient != 0) { 7481 mEffectClient->enableStatusChanged(enabled); 7482 } 7483} 7484 7485status_t AudioFlinger::EffectHandle::onTransact( 7486 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7487{ 7488 return BnEffect::onTransact(code, data, reply, flags); 7489} 7490 7491 7492void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7493{ 7494 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7495 7496 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7497 (mClient == 0) ? getpid_cached : mClient->pid(), 7498 mPriority, 7499 mHasControl, 7500 !locked, 7501 mCblk ? mCblk->clientIndex : 0, 7502 mCblk ? mCblk->serverIndex : 0 7503 ); 7504 7505 if (locked) { 7506 mCblk->lock.unlock(); 7507 } 7508} 7509 7510#undef LOG_TAG 7511#define LOG_TAG "AudioFlinger::EffectChain" 7512 7513AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7514 int sessionId) 7515 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7516 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7517 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7518{ 7519 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7520 if (thread == NULL) { 7521 return; 7522 } 7523 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7524 thread->frameCount(); 7525} 7526 7527AudioFlinger::EffectChain::~EffectChain() 7528{ 7529 if (mOwnInBuffer) { 7530 delete mInBuffer; 7531 } 7532 7533} 7534 7535// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7536sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7537{ 7538 size_t size = mEffects.size(); 7539 7540 for (size_t i = 0; i < size; i++) { 7541 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7542 return mEffects[i]; 7543 } 7544 } 7545 return 0; 7546} 7547 7548// getEffectFromId_l() must be called with ThreadBase::mLock held 7549sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7550{ 7551 size_t size = mEffects.size(); 7552 7553 for (size_t i = 0; i < size; i++) { 7554 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7555 if (id == 0 || mEffects[i]->id() == id) { 7556 return mEffects[i]; 7557 } 7558 } 7559 return 0; 7560} 7561 7562// getEffectFromType_l() must be called with ThreadBase::mLock held 7563sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7564 const effect_uuid_t *type) 7565{ 7566 size_t size = mEffects.size(); 7567 7568 for (size_t i = 0; i < size; i++) { 7569 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7570 return mEffects[i]; 7571 } 7572 } 7573 return 0; 7574} 7575 7576// Must be called with EffectChain::mLock locked 7577void AudioFlinger::EffectChain::process_l() 7578{ 7579 sp<ThreadBase> thread = mThread.promote(); 7580 if (thread == 0) { 7581 ALOGW("process_l(): cannot promote mixer thread"); 7582 return; 7583 } 7584 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7585 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7586 // always process effects unless no more tracks are on the session and the effect tail 7587 // has been rendered 7588 bool doProcess = true; 7589 if (!isGlobalSession) { 7590 bool tracksOnSession = (trackCnt() != 0); 7591 7592 if (!tracksOnSession && mTailBufferCount == 0) { 7593 doProcess = false; 7594 } 7595 7596 if (activeTrackCnt() == 0) { 7597 // if no track is active and the effect tail has not been rendered, 7598 // the input buffer must be cleared here as the mixer process will not do it 7599 if (tracksOnSession || mTailBufferCount > 0) { 7600 size_t numSamples = thread->frameCount() * thread->channelCount(); 7601 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7602 if (mTailBufferCount > 0) { 7603 mTailBufferCount--; 7604 } 7605 } 7606 } 7607 } 7608 7609 size_t size = mEffects.size(); 7610 if (doProcess) { 7611 for (size_t i = 0; i < size; i++) { 7612 mEffects[i]->process(); 7613 } 7614 } 7615 for (size_t i = 0; i < size; i++) { 7616 mEffects[i]->updateState(); 7617 } 7618} 7619 7620// addEffect_l() must be called with PlaybackThread::mLock held 7621status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7622{ 7623 effect_descriptor_t desc = effect->desc(); 7624 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7625 7626 Mutex::Autolock _l(mLock); 7627 effect->setChain(this); 7628 sp<ThreadBase> thread = mThread.promote(); 7629 if (thread == 0) { 7630 return NO_INIT; 7631 } 7632 effect->setThread(thread); 7633 7634 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7635 // Auxiliary effects are inserted at the beginning of mEffects vector as 7636 // they are processed first and accumulated in chain input buffer 7637 mEffects.insertAt(effect, 0); 7638 7639 // the input buffer for auxiliary effect contains mono samples in 7640 // 32 bit format. This is to avoid saturation in AudoMixer 7641 // accumulation stage. Saturation is done in EffectModule::process() before 7642 // calling the process in effect engine 7643 size_t numSamples = thread->frameCount(); 7644 int32_t *buffer = new int32_t[numSamples]; 7645 memset(buffer, 0, numSamples * sizeof(int32_t)); 7646 effect->setInBuffer((int16_t *)buffer); 7647 // auxiliary effects output samples to chain input buffer for further processing 7648 // by insert effects 7649 effect->setOutBuffer(mInBuffer); 7650 } else { 7651 // Insert effects are inserted at the end of mEffects vector as they are processed 7652 // after track and auxiliary effects. 7653 // Insert effect order as a function of indicated preference: 7654 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7655 // another effect is present 7656 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7657 // last effect claiming first position 7658 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7659 // first effect claiming last position 7660 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7661 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7662 // already present 7663 7664 size_t size = mEffects.size(); 7665 size_t idx_insert = size; 7666 ssize_t idx_insert_first = -1; 7667 ssize_t idx_insert_last = -1; 7668 7669 for (size_t i = 0; i < size; i++) { 7670 effect_descriptor_t d = mEffects[i]->desc(); 7671 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7672 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7673 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7674 // check invalid effect chaining combinations 7675 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7676 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7677 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7678 return INVALID_OPERATION; 7679 } 7680 // remember position of first insert effect and by default 7681 // select this as insert position for new effect 7682 if (idx_insert == size) { 7683 idx_insert = i; 7684 } 7685 // remember position of last insert effect claiming 7686 // first position 7687 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7688 idx_insert_first = i; 7689 } 7690 // remember position of first insert effect claiming 7691 // last position 7692 if (iPref == EFFECT_FLAG_INSERT_LAST && 7693 idx_insert_last == -1) { 7694 idx_insert_last = i; 7695 } 7696 } 7697 } 7698 7699 // modify idx_insert from first position if needed 7700 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7701 if (idx_insert_last != -1) { 7702 idx_insert = idx_insert_last; 7703 } else { 7704 idx_insert = size; 7705 } 7706 } else { 7707 if (idx_insert_first != -1) { 7708 idx_insert = idx_insert_first + 1; 7709 } 7710 } 7711 7712 // always read samples from chain input buffer 7713 effect->setInBuffer(mInBuffer); 7714 7715 // if last effect in the chain, output samples to chain 7716 // output buffer, otherwise to chain input buffer 7717 if (idx_insert == size) { 7718 if (idx_insert != 0) { 7719 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7720 mEffects[idx_insert-1]->configure(); 7721 } 7722 effect->setOutBuffer(mOutBuffer); 7723 } else { 7724 effect->setOutBuffer(mInBuffer); 7725 } 7726 mEffects.insertAt(effect, idx_insert); 7727 7728 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7729 } 7730 effect->configure(); 7731 return NO_ERROR; 7732} 7733 7734// removeEffect_l() must be called with PlaybackThread::mLock held 7735size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7736{ 7737 Mutex::Autolock _l(mLock); 7738 size_t size = mEffects.size(); 7739 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7740 7741 for (size_t i = 0; i < size; i++) { 7742 if (effect == mEffects[i]) { 7743 // calling stop here will remove pre-processing effect from the audio HAL. 7744 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7745 // the middle of a read from audio HAL 7746 if (mEffects[i]->state() == EffectModule::ACTIVE || 7747 mEffects[i]->state() == EffectModule::STOPPING) { 7748 mEffects[i]->stop(); 7749 } 7750 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7751 delete[] effect->inBuffer(); 7752 } else { 7753 if (i == size - 1 && i != 0) { 7754 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7755 mEffects[i - 1]->configure(); 7756 } 7757 } 7758 mEffects.removeAt(i); 7759 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7760 break; 7761 } 7762 } 7763 7764 return mEffects.size(); 7765} 7766 7767// setDevice_l() must be called with PlaybackThread::mLock held 7768void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7769{ 7770 size_t size = mEffects.size(); 7771 for (size_t i = 0; i < size; i++) { 7772 mEffects[i]->setDevice(device); 7773 } 7774} 7775 7776// setMode_l() must be called with PlaybackThread::mLock held 7777void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7778{ 7779 size_t size = mEffects.size(); 7780 for (size_t i = 0; i < size; i++) { 7781 mEffects[i]->setMode(mode); 7782 } 7783} 7784 7785// setVolume_l() must be called with PlaybackThread::mLock held 7786bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7787{ 7788 uint32_t newLeft = *left; 7789 uint32_t newRight = *right; 7790 bool hasControl = false; 7791 int ctrlIdx = -1; 7792 size_t size = mEffects.size(); 7793 7794 // first update volume controller 7795 for (size_t i = size; i > 0; i--) { 7796 if (mEffects[i - 1]->isProcessEnabled() && 7797 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7798 ctrlIdx = i - 1; 7799 hasControl = true; 7800 break; 7801 } 7802 } 7803 7804 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7805 if (hasControl) { 7806 *left = mNewLeftVolume; 7807 *right = mNewRightVolume; 7808 } 7809 return hasControl; 7810 } 7811 7812 mVolumeCtrlIdx = ctrlIdx; 7813 mLeftVolume = newLeft; 7814 mRightVolume = newRight; 7815 7816 // second get volume update from volume controller 7817 if (ctrlIdx >= 0) { 7818 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7819 mNewLeftVolume = newLeft; 7820 mNewRightVolume = newRight; 7821 } 7822 // then indicate volume to all other effects in chain. 7823 // Pass altered volume to effects before volume controller 7824 // and requested volume to effects after controller 7825 uint32_t lVol = newLeft; 7826 uint32_t rVol = newRight; 7827 7828 for (size_t i = 0; i < size; i++) { 7829 if ((int)i == ctrlIdx) continue; 7830 // this also works for ctrlIdx == -1 when there is no volume controller 7831 if ((int)i > ctrlIdx) { 7832 lVol = *left; 7833 rVol = *right; 7834 } 7835 mEffects[i]->setVolume(&lVol, &rVol, false); 7836 } 7837 *left = newLeft; 7838 *right = newRight; 7839 7840 return hasControl; 7841} 7842 7843status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7844{ 7845 const size_t SIZE = 256; 7846 char buffer[SIZE]; 7847 String8 result; 7848 7849 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7850 result.append(buffer); 7851 7852 bool locked = tryLock(mLock); 7853 // failed to lock - AudioFlinger is probably deadlocked 7854 if (!locked) { 7855 result.append("\tCould not lock mutex:\n"); 7856 } 7857 7858 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7859 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7860 mEffects.size(), 7861 (uint32_t)mInBuffer, 7862 (uint32_t)mOutBuffer, 7863 mActiveTrackCnt); 7864 result.append(buffer); 7865 write(fd, result.string(), result.size()); 7866 7867 for (size_t i = 0; i < mEffects.size(); ++i) { 7868 sp<EffectModule> effect = mEffects[i]; 7869 if (effect != 0) { 7870 effect->dump(fd, args); 7871 } 7872 } 7873 7874 if (locked) { 7875 mLock.unlock(); 7876 } 7877 7878 return NO_ERROR; 7879} 7880 7881// must be called with ThreadBase::mLock held 7882void AudioFlinger::EffectChain::setEffectSuspended_l( 7883 const effect_uuid_t *type, bool suspend) 7884{ 7885 sp<SuspendedEffectDesc> desc; 7886 // use effect type UUID timelow as key as there is no real risk of identical 7887 // timeLow fields among effect type UUIDs. 7888 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7889 if (suspend) { 7890 if (index >= 0) { 7891 desc = mSuspendedEffects.valueAt(index); 7892 } else { 7893 desc = new SuspendedEffectDesc(); 7894 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7895 mSuspendedEffects.add(type->timeLow, desc); 7896 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7897 } 7898 if (desc->mRefCount++ == 0) { 7899 sp<EffectModule> effect = getEffectIfEnabled(type); 7900 if (effect != 0) { 7901 desc->mEffect = effect; 7902 effect->setSuspended(true); 7903 effect->setEnabled(false); 7904 } 7905 } 7906 } else { 7907 if (index < 0) { 7908 return; 7909 } 7910 desc = mSuspendedEffects.valueAt(index); 7911 if (desc->mRefCount <= 0) { 7912 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7913 desc->mRefCount = 1; 7914 } 7915 if (--desc->mRefCount == 0) { 7916 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7917 if (desc->mEffect != 0) { 7918 sp<EffectModule> effect = desc->mEffect.promote(); 7919 if (effect != 0) { 7920 effect->setSuspended(false); 7921 sp<EffectHandle> handle = effect->controlHandle(); 7922 if (handle != 0) { 7923 effect->setEnabled(handle->enabled()); 7924 } 7925 } 7926 desc->mEffect.clear(); 7927 } 7928 mSuspendedEffects.removeItemsAt(index); 7929 } 7930 } 7931} 7932 7933// must be called with ThreadBase::mLock held 7934void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7935{ 7936 sp<SuspendedEffectDesc> desc; 7937 7938 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7939 if (suspend) { 7940 if (index >= 0) { 7941 desc = mSuspendedEffects.valueAt(index); 7942 } else { 7943 desc = new SuspendedEffectDesc(); 7944 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7945 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7946 } 7947 if (desc->mRefCount++ == 0) { 7948 Vector< sp<EffectModule> > effects; 7949 getSuspendEligibleEffects(effects); 7950 for (size_t i = 0; i < effects.size(); i++) { 7951 setEffectSuspended_l(&effects[i]->desc().type, true); 7952 } 7953 } 7954 } else { 7955 if (index < 0) { 7956 return; 7957 } 7958 desc = mSuspendedEffects.valueAt(index); 7959 if (desc->mRefCount <= 0) { 7960 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7961 desc->mRefCount = 1; 7962 } 7963 if (--desc->mRefCount == 0) { 7964 Vector<const effect_uuid_t *> types; 7965 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7966 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7967 continue; 7968 } 7969 types.add(&mSuspendedEffects.valueAt(i)->mType); 7970 } 7971 for (size_t i = 0; i < types.size(); i++) { 7972 setEffectSuspended_l(types[i], false); 7973 } 7974 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7975 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7976 } 7977 } 7978} 7979 7980 7981// The volume effect is used for automated tests only 7982#ifndef OPENSL_ES_H_ 7983static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7984 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7985const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7986#endif //OPENSL_ES_H_ 7987 7988bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7989{ 7990 // auxiliary effects and visualizer are never suspended on output mix 7991 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7992 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7993 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7994 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7995 return false; 7996 } 7997 return true; 7998} 7999 8000void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 8001{ 8002 effects.clear(); 8003 for (size_t i = 0; i < mEffects.size(); i++) { 8004 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 8005 effects.add(mEffects[i]); 8006 } 8007 } 8008} 8009 8010sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 8011 const effect_uuid_t *type) 8012{ 8013 sp<EffectModule> effect = getEffectFromType_l(type); 8014 return effect != 0 && effect->isEnabled() ? effect : 0; 8015} 8016 8017void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8018 bool enabled) 8019{ 8020 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8021 if (enabled) { 8022 if (index < 0) { 8023 // if the effect is not suspend check if all effects are suspended 8024 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8025 if (index < 0) { 8026 return; 8027 } 8028 if (!isEffectEligibleForSuspend(effect->desc())) { 8029 return; 8030 } 8031 setEffectSuspended_l(&effect->desc().type, enabled); 8032 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8033 if (index < 0) { 8034 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8035 return; 8036 } 8037 } 8038 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8039 effect->desc().type.timeLow); 8040 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8041 // if effect is requested to suspended but was not yet enabled, supend it now. 8042 if (desc->mEffect == 0) { 8043 desc->mEffect = effect; 8044 effect->setEnabled(false); 8045 effect->setSuspended(true); 8046 } 8047 } else { 8048 if (index < 0) { 8049 return; 8050 } 8051 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8052 effect->desc().type.timeLow); 8053 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8054 desc->mEffect.clear(); 8055 effect->setSuspended(false); 8056 } 8057} 8058 8059#undef LOG_TAG 8060#define LOG_TAG "AudioFlinger" 8061 8062// ---------------------------------------------------------------------------- 8063 8064status_t AudioFlinger::onTransact( 8065 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8066{ 8067 return BnAudioFlinger::onTransact(code, data, reply, flags); 8068} 8069 8070}; // namespace android 8071