AudioFlinger.cpp revision b071e9bc248865ef87a339044c0c5cbabfac175c
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include <math.h> 23#include <signal.h> 24#include <sys/time.h> 25#include <sys/resource.h> 26 27#include <binder/IPCThreadState.h> 28#include <binder/IServiceManager.h> 29#include <utils/Log.h> 30#include <binder/Parcel.h> 31#include <binder/IPCThreadState.h> 32#include <utils/String16.h> 33#include <utils/threads.h> 34#include <utils/Atomic.h> 35 36#include <cutils/bitops.h> 37#include <cutils/properties.h> 38#include <cutils/compiler.h> 39 40#include <media/IMediaPlayerService.h> 41#include <media/IMediaDeathNotifier.h> 42 43#include <private/media/AudioTrackShared.h> 44#include <private/media/AudioEffectShared.h> 45 46#include <system/audio.h> 47#include <hardware/audio.h> 48 49#include "AudioMixer.h" 50#include "AudioFlinger.h" 51#include "ServiceUtilities.h" 52 53#include <media/EffectsFactoryApi.h> 54#include <audio_effects/effect_visualizer.h> 55#include <audio_effects/effect_ns.h> 56#include <audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <cpustats/ThreadCpuUsage.h> 61#include <powermanager/PowerManager.h> 62// #define DEBUG_CPU_USAGE 10 // log statistics every n wall clock seconds 63 64#include <common_time/cc_helper.h> 65#include <common_time/local_clock.h> 66 67// ---------------------------------------------------------------------------- 68 69 70namespace android { 71 72static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 73static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 74 75static const float MAX_GAIN = 4096.0f; 76static const uint32_t MAX_GAIN_INT = 0x1000; 77 78// retry counts for buffer fill timeout 79// 50 * ~20msecs = 1 second 80static const int8_t kMaxTrackRetries = 50; 81static const int8_t kMaxTrackStartupRetries = 50; 82// allow less retry attempts on direct output thread. 83// direct outputs can be a scarce resource in audio hardware and should 84// be released as quickly as possible. 85static const int8_t kMaxTrackRetriesDirect = 2; 86 87static const int kDumpLockRetries = 50; 88static const int kDumpLockSleepUs = 20000; 89 90// don't warn about blocked writes or record buffer overflows more often than this 91static const nsecs_t kWarningThrottleNs = seconds(5); 92 93// RecordThread loop sleep time upon application overrun or audio HAL read error 94static const int kRecordThreadSleepUs = 5000; 95 96// maximum time to wait for setParameters to complete 97static const nsecs_t kSetParametersTimeoutNs = seconds(2); 98 99// minimum sleep time for the mixer thread loop when tracks are active but in underrun 100static const uint32_t kMinThreadSleepTimeUs = 5000; 101// maximum divider applied to the active sleep time in the mixer thread loop 102static const uint32_t kMaxThreadSleepTimeShift = 2; 103 104nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 105 106// ---------------------------------------------------------------------------- 107 108// To collect the amplifier usage 109static void addBatteryData(uint32_t params) { 110 sp<IMediaPlayerService> service = IMediaDeathNotifier::getMediaPlayerService(); 111 if (service == NULL) { 112 // it already logged 113 return; 114 } 115 116 service->addBatteryData(params); 117} 118 119static int load_audio_interface(const char *if_name, const hw_module_t **mod, 120 audio_hw_device_t **dev) 121{ 122 int rc; 123 124 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, mod); 125 if (rc) 126 goto out; 127 128 rc = audio_hw_device_open(*mod, dev); 129 ALOGE_IF(rc, "couldn't open audio hw device in %s.%s (%s)", 130 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 131 if (rc) 132 goto out; 133 134 return 0; 135 136out: 137 *mod = NULL; 138 *dev = NULL; 139 return rc; 140} 141 142static const char * const audio_interfaces[] = { 143 "primary", 144 "a2dp", 145 "usb", 146}; 147#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 148 149// ---------------------------------------------------------------------------- 150 151AudioFlinger::AudioFlinger() 152 : BnAudioFlinger(), 153 mPrimaryHardwareDev(NULL), 154 mHardwareStatus(AUDIO_HW_IDLE), // see also onFirstRef() 155 mMasterVolume(1.0f), 156 mMasterVolumeSupportLvl(MVS_NONE), 157 mMasterMute(false), 158 mNextUniqueId(1), 159 mMode(AUDIO_MODE_INVALID), 160 mBtNrecIsOff(false) 161{ 162} 163 164void AudioFlinger::onFirstRef() 165{ 166 int rc = 0; 167 168 Mutex::Autolock _l(mLock); 169 170 /* TODO: move all this work into an Init() function */ 171 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 172 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 173 uint32_t int_val; 174 if (1 == sscanf(val_str, "%u", &int_val)) { 175 mStandbyTimeInNsecs = milliseconds(int_val); 176 ALOGI("Using %u mSec as standby time.", int_val); 177 } else { 178 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 179 ALOGI("Using default %u mSec as standby time.", 180 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 181 } 182 } 183 184 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 185 const hw_module_t *mod; 186 audio_hw_device_t *dev; 187 188 rc = load_audio_interface(audio_interfaces[i], &mod, &dev); 189 if (rc) 190 continue; 191 192 ALOGI("Loaded %s audio interface from %s (%s)", audio_interfaces[i], 193 mod->name, mod->id); 194 mAudioHwDevs.push(dev); 195 196 if (mPrimaryHardwareDev == NULL) { 197 mPrimaryHardwareDev = dev; 198 ALOGI("Using '%s' (%s.%s) as the primary audio interface", 199 mod->name, mod->id, audio_interfaces[i]); 200 } 201 } 202 203 if (mPrimaryHardwareDev == NULL) { 204 ALOGE("Primary audio interface not found"); 205 // proceed, all later accesses to mPrimaryHardwareDev verify it's safe with initCheck() 206 } 207 208 // Currently (mPrimaryHardwareDev == NULL) == (mAudioHwDevs.size() == 0), but the way the 209 // primary HW dev is selected can change so these conditions might not always be equivalent. 210 // When that happens, re-visit all the code that assumes this. 211 212 AutoMutex lock(mHardwareLock); 213 214 // Determine the level of master volume support the primary audio HAL has, 215 // and set the initial master volume at the same time. 216 float initialVolume = 1.0; 217 mMasterVolumeSupportLvl = MVS_NONE; 218 if (0 == mPrimaryHardwareDev->init_check(mPrimaryHardwareDev)) { 219 audio_hw_device_t *dev = mPrimaryHardwareDev; 220 221 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 222 if ((NULL != dev->get_master_volume) && 223 (NO_ERROR == dev->get_master_volume(dev, &initialVolume))) { 224 mMasterVolumeSupportLvl = MVS_FULL; 225 } else { 226 mMasterVolumeSupportLvl = MVS_SETONLY; 227 initialVolume = 1.0; 228 } 229 230 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 231 if ((NULL == dev->set_master_volume) || 232 (NO_ERROR != dev->set_master_volume(dev, initialVolume))) { 233 mMasterVolumeSupportLvl = MVS_NONE; 234 } 235 mHardwareStatus = AUDIO_HW_IDLE; 236 } 237 238 // Set the mode for each audio HAL, and try to set the initial volume (if 239 // supported) for all of the non-primary audio HALs. 240 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 241 audio_hw_device_t *dev = mAudioHwDevs[i]; 242 243 mHardwareStatus = AUDIO_HW_INIT; 244 rc = dev->init_check(dev); 245 mHardwareStatus = AUDIO_HW_IDLE; 246 if (rc == 0) { 247 mMode = AUDIO_MODE_NORMAL; // assigned multiple times with same value 248 mHardwareStatus = AUDIO_HW_SET_MODE; 249 dev->set_mode(dev, mMode); 250 251 if ((dev != mPrimaryHardwareDev) && 252 (NULL != dev->set_master_volume)) { 253 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 254 dev->set_master_volume(dev, initialVolume); 255 } 256 257 mHardwareStatus = AUDIO_HW_IDLE; 258 } 259 } 260 261 mMasterVolumeSW = (MVS_NONE == mMasterVolumeSupportLvl) 262 ? initialVolume 263 : 1.0; 264 mMasterVolume = initialVolume; 265 mHardwareStatus = AUDIO_HW_IDLE; 266} 267 268AudioFlinger::~AudioFlinger() 269{ 270 271 while (!mRecordThreads.isEmpty()) { 272 // closeInput() will remove first entry from mRecordThreads 273 closeInput(mRecordThreads.keyAt(0)); 274 } 275 while (!mPlaybackThreads.isEmpty()) { 276 // closeOutput() will remove first entry from mPlaybackThreads 277 closeOutput(mPlaybackThreads.keyAt(0)); 278 } 279 280 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 281 // no mHardwareLock needed, as there are no other references to this 282 audio_hw_device_close(mAudioHwDevs[i]); 283 } 284} 285 286audio_hw_device_t* AudioFlinger::findSuitableHwDev_l(uint32_t devices) 287{ 288 /* first matching HW device is returned */ 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 audio_hw_device_t *dev = mAudioHwDevs[i]; 291 if ((dev->get_supported_devices(dev) & devices) == devices) 292 return dev; 293 } 294 return NULL; 295} 296 297status_t AudioFlinger::dumpClients(int fd, const Vector<String16>& args) 298{ 299 const size_t SIZE = 256; 300 char buffer[SIZE]; 301 String8 result; 302 303 result.append("Clients:\n"); 304 for (size_t i = 0; i < mClients.size(); ++i) { 305 sp<Client> client = mClients.valueAt(i).promote(); 306 if (client != 0) { 307 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 308 result.append(buffer); 309 } 310 } 311 312 result.append("Global session refs:\n"); 313 result.append(" session pid count\n"); 314 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 315 AudioSessionRef *r = mAudioSessionRefs[i]; 316 snprintf(buffer, SIZE, " %7d %3d %3d\n", r->mSessionid, r->mPid, r->mCnt); 317 result.append(buffer); 318 } 319 write(fd, result.string(), result.size()); 320 return NO_ERROR; 321} 322 323 324status_t AudioFlinger::dumpInternals(int fd, const Vector<String16>& args) 325{ 326 const size_t SIZE = 256; 327 char buffer[SIZE]; 328 String8 result; 329 hardware_call_state hardwareStatus = mHardwareStatus; 330 331 snprintf(buffer, SIZE, "Hardware status: %d\n" 332 "Standby Time mSec: %u\n", 333 hardwareStatus, 334 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 335 result.append(buffer); 336 write(fd, result.string(), result.size()); 337 return NO_ERROR; 338} 339 340status_t AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args) 341{ 342 const size_t SIZE = 256; 343 char buffer[SIZE]; 344 String8 result; 345 snprintf(buffer, SIZE, "Permission Denial: " 346 "can't dump AudioFlinger from pid=%d, uid=%d\n", 347 IPCThreadState::self()->getCallingPid(), 348 IPCThreadState::self()->getCallingUid()); 349 result.append(buffer); 350 write(fd, result.string(), result.size()); 351 return NO_ERROR; 352} 353 354static bool tryLock(Mutex& mutex) 355{ 356 bool locked = false; 357 for (int i = 0; i < kDumpLockRetries; ++i) { 358 if (mutex.tryLock() == NO_ERROR) { 359 locked = true; 360 break; 361 } 362 usleep(kDumpLockSleepUs); 363 } 364 return locked; 365} 366 367status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 368{ 369 if (!dumpAllowed()) { 370 dumpPermissionDenial(fd, args); 371 } else { 372 // get state of hardware lock 373 bool hardwareLocked = tryLock(mHardwareLock); 374 if (!hardwareLocked) { 375 String8 result(kHardwareLockedString); 376 write(fd, result.string(), result.size()); 377 } else { 378 mHardwareLock.unlock(); 379 } 380 381 bool locked = tryLock(mLock); 382 383 // failed to lock - AudioFlinger is probably deadlocked 384 if (!locked) { 385 String8 result(kDeadlockedString); 386 write(fd, result.string(), result.size()); 387 } 388 389 dumpClients(fd, args); 390 dumpInternals(fd, args); 391 392 // dump playback threads 393 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 394 mPlaybackThreads.valueAt(i)->dump(fd, args); 395 } 396 397 // dump record threads 398 for (size_t i = 0; i < mRecordThreads.size(); i++) { 399 mRecordThreads.valueAt(i)->dump(fd, args); 400 } 401 402 // dump all hardware devs 403 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 404 audio_hw_device_t *dev = mAudioHwDevs[i]; 405 dev->dump(dev, fd); 406 } 407 if (locked) mLock.unlock(); 408 } 409 return NO_ERROR; 410} 411 412sp<AudioFlinger::Client> AudioFlinger::registerPid_l(pid_t pid) 413{ 414 // If pid is already in the mClients wp<> map, then use that entry 415 // (for which promote() is always != 0), otherwise create a new entry and Client. 416 sp<Client> client = mClients.valueFor(pid).promote(); 417 if (client == 0) { 418 client = new Client(this, pid); 419 mClients.add(pid, client); 420 } 421 422 return client; 423} 424 425// IAudioFlinger interface 426 427 428sp<IAudioTrack> AudioFlinger::createTrack( 429 pid_t pid, 430 audio_stream_type_t streamType, 431 uint32_t sampleRate, 432 audio_format_t format, 433 uint32_t channelMask, 434 int frameCount, 435 // FIXME dead, remove from IAudioFlinger 436 uint32_t flags, 437 const sp<IMemory>& sharedBuffer, 438 audio_io_handle_t output, 439 bool isTimed, 440 int *sessionId, 441 status_t *status) 442{ 443 sp<PlaybackThread::Track> track; 444 sp<TrackHandle> trackHandle; 445 sp<Client> client; 446 status_t lStatus; 447 int lSessionId; 448 449 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 450 // but if someone uses binder directly they could bypass that and cause us to crash 451 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 452 ALOGE("createTrack() invalid stream type %d", streamType); 453 lStatus = BAD_VALUE; 454 goto Exit; 455 } 456 457 { 458 Mutex::Autolock _l(mLock); 459 PlaybackThread *thread = checkPlaybackThread_l(output); 460 PlaybackThread *effectThread = NULL; 461 if (thread == NULL) { 462 ALOGE("unknown output thread"); 463 lStatus = BAD_VALUE; 464 goto Exit; 465 } 466 467 client = registerPid_l(pid); 468 469 ALOGV("createTrack() sessionId: %d", (sessionId == NULL) ? -2 : *sessionId); 470 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 471 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 472 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 473 if (mPlaybackThreads.keyAt(i) != output) { 474 // prevent same audio session on different output threads 475 uint32_t sessions = t->hasAudioSession(*sessionId); 476 if (sessions & PlaybackThread::TRACK_SESSION) { 477 ALOGE("createTrack() session ID %d already in use", *sessionId); 478 lStatus = BAD_VALUE; 479 goto Exit; 480 } 481 // check if an effect with same session ID is waiting for a track to be created 482 if (sessions & PlaybackThread::EFFECT_SESSION) { 483 effectThread = t.get(); 484 } 485 } 486 } 487 lSessionId = *sessionId; 488 } else { 489 // if no audio session id is provided, create one here 490 lSessionId = nextUniqueId(); 491 if (sessionId != NULL) { 492 *sessionId = lSessionId; 493 } 494 } 495 ALOGV("createTrack() lSessionId: %d", lSessionId); 496 497 track = thread->createTrack_l(client, streamType, sampleRate, format, 498 channelMask, frameCount, sharedBuffer, lSessionId, isTimed, &lStatus); 499 500 // move effect chain to this output thread if an effect on same session was waiting 501 // for a track to be created 502 if (lStatus == NO_ERROR && effectThread != NULL) { 503 Mutex::Autolock _dl(thread->mLock); 504 Mutex::Autolock _sl(effectThread->mLock); 505 moveEffectChain_l(lSessionId, effectThread, thread, true); 506 } 507 } 508 if (lStatus == NO_ERROR) { 509 trackHandle = new TrackHandle(track); 510 } else { 511 // remove local strong reference to Client before deleting the Track so that the Client 512 // destructor is called by the TrackBase destructor with mLock held 513 client.clear(); 514 track.clear(); 515 } 516 517Exit: 518 if(status) { 519 *status = lStatus; 520 } 521 return trackHandle; 522} 523 524uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 525{ 526 Mutex::Autolock _l(mLock); 527 PlaybackThread *thread = checkPlaybackThread_l(output); 528 if (thread == NULL) { 529 ALOGW("sampleRate() unknown thread %d", output); 530 return 0; 531 } 532 return thread->sampleRate(); 533} 534 535int AudioFlinger::channelCount(audio_io_handle_t output) const 536{ 537 Mutex::Autolock _l(mLock); 538 PlaybackThread *thread = checkPlaybackThread_l(output); 539 if (thread == NULL) { 540 ALOGW("channelCount() unknown thread %d", output); 541 return 0; 542 } 543 return thread->channelCount(); 544} 545 546audio_format_t AudioFlinger::format(audio_io_handle_t output) const 547{ 548 Mutex::Autolock _l(mLock); 549 PlaybackThread *thread = checkPlaybackThread_l(output); 550 if (thread == NULL) { 551 ALOGW("format() unknown thread %d", output); 552 return AUDIO_FORMAT_INVALID; 553 } 554 return thread->format(); 555} 556 557size_t AudioFlinger::frameCount(audio_io_handle_t output) const 558{ 559 Mutex::Autolock _l(mLock); 560 PlaybackThread *thread = checkPlaybackThread_l(output); 561 if (thread == NULL) { 562 ALOGW("frameCount() unknown thread %d", output); 563 return 0; 564 } 565 return thread->frameCount(); 566} 567 568uint32_t AudioFlinger::latency(audio_io_handle_t output) const 569{ 570 Mutex::Autolock _l(mLock); 571 PlaybackThread *thread = checkPlaybackThread_l(output); 572 if (thread == NULL) { 573 ALOGW("latency() unknown thread %d", output); 574 return 0; 575 } 576 return thread->latency(); 577} 578 579status_t AudioFlinger::setMasterVolume(float value) 580{ 581 status_t ret = initCheck(); 582 if (ret != NO_ERROR) { 583 return ret; 584 } 585 586 // check calling permissions 587 if (!settingsAllowed()) { 588 return PERMISSION_DENIED; 589 } 590 591 float swmv = value; 592 593 // when hw supports master volume, don't scale in sw mixer 594 if (MVS_NONE != mMasterVolumeSupportLvl) { 595 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 596 AutoMutex lock(mHardwareLock); 597 audio_hw_device_t *dev = mAudioHwDevs[i]; 598 599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 600 if (NULL != dev->set_master_volume) { 601 dev->set_master_volume(dev, value); 602 } 603 mHardwareStatus = AUDIO_HW_IDLE; 604 } 605 606 swmv = 1.0; 607 } 608 609 Mutex::Autolock _l(mLock); 610 mMasterVolume = value; 611 mMasterVolumeSW = swmv; 612 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 613 mPlaybackThreads.valueAt(i)->setMasterVolume(swmv); 614 615 return NO_ERROR; 616} 617 618status_t AudioFlinger::setMode(audio_mode_t mode) 619{ 620 status_t ret = initCheck(); 621 if (ret != NO_ERROR) { 622 return ret; 623 } 624 625 // check calling permissions 626 if (!settingsAllowed()) { 627 return PERMISSION_DENIED; 628 } 629 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 630 ALOGW("Illegal value: setMode(%d)", mode); 631 return BAD_VALUE; 632 } 633 634 { // scope for the lock 635 AutoMutex lock(mHardwareLock); 636 mHardwareStatus = AUDIO_HW_SET_MODE; 637 ret = mPrimaryHardwareDev->set_mode(mPrimaryHardwareDev, mode); 638 mHardwareStatus = AUDIO_HW_IDLE; 639 } 640 641 if (NO_ERROR == ret) { 642 Mutex::Autolock _l(mLock); 643 mMode = mode; 644 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 645 mPlaybackThreads.valueAt(i)->setMode(mode); 646 } 647 648 return ret; 649} 650 651status_t AudioFlinger::setMicMute(bool state) 652{ 653 status_t ret = initCheck(); 654 if (ret != NO_ERROR) { 655 return ret; 656 } 657 658 // check calling permissions 659 if (!settingsAllowed()) { 660 return PERMISSION_DENIED; 661 } 662 663 AutoMutex lock(mHardwareLock); 664 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 665 ret = mPrimaryHardwareDev->set_mic_mute(mPrimaryHardwareDev, state); 666 mHardwareStatus = AUDIO_HW_IDLE; 667 return ret; 668} 669 670bool AudioFlinger::getMicMute() const 671{ 672 status_t ret = initCheck(); 673 if (ret != NO_ERROR) { 674 return false; 675 } 676 677 bool state = AUDIO_MODE_INVALID; 678 AutoMutex lock(mHardwareLock); 679 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 680 mPrimaryHardwareDev->get_mic_mute(mPrimaryHardwareDev, &state); 681 mHardwareStatus = AUDIO_HW_IDLE; 682 return state; 683} 684 685status_t AudioFlinger::setMasterMute(bool muted) 686{ 687 // check calling permissions 688 if (!settingsAllowed()) { 689 return PERMISSION_DENIED; 690 } 691 692 Mutex::Autolock _l(mLock); 693 // This is an optimization, so PlaybackThread doesn't have to look at the one from AudioFlinger 694 mMasterMute = muted; 695 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 696 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 697 698 return NO_ERROR; 699} 700 701float AudioFlinger::masterVolume() const 702{ 703 Mutex::Autolock _l(mLock); 704 return masterVolume_l(); 705} 706 707float AudioFlinger::masterVolumeSW() const 708{ 709 Mutex::Autolock _l(mLock); 710 return masterVolumeSW_l(); 711} 712 713bool AudioFlinger::masterMute() const 714{ 715 Mutex::Autolock _l(mLock); 716 return masterMute_l(); 717} 718 719float AudioFlinger::masterVolume_l() const 720{ 721 if (MVS_FULL == mMasterVolumeSupportLvl) { 722 float ret_val; 723 AutoMutex lock(mHardwareLock); 724 725 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 726 assert(NULL != mPrimaryHardwareDev); 727 assert(NULL != mPrimaryHardwareDev->get_master_volume); 728 729 mPrimaryHardwareDev->get_master_volume(mPrimaryHardwareDev, &ret_val); 730 mHardwareStatus = AUDIO_HW_IDLE; 731 return ret_val; 732 } 733 734 return mMasterVolume; 735} 736 737status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 738 audio_io_handle_t output) 739{ 740 // check calling permissions 741 if (!settingsAllowed()) { 742 return PERMISSION_DENIED; 743 } 744 745 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 746 ALOGE("setStreamVolume() invalid stream %d", stream); 747 return BAD_VALUE; 748 } 749 750 AutoMutex lock(mLock); 751 PlaybackThread *thread = NULL; 752 if (output) { 753 thread = checkPlaybackThread_l(output); 754 if (thread == NULL) { 755 return BAD_VALUE; 756 } 757 } 758 759 mStreamTypes[stream].volume = value; 760 761 if (thread == NULL) { 762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 763 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 764 } 765 } else { 766 thread->setStreamVolume(stream, value); 767 } 768 769 return NO_ERROR; 770} 771 772status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 773{ 774 // check calling permissions 775 if (!settingsAllowed()) { 776 return PERMISSION_DENIED; 777 } 778 779 if (uint32_t(stream) >= AUDIO_STREAM_CNT || 780 uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 781 ALOGE("setStreamMute() invalid stream %d", stream); 782 return BAD_VALUE; 783 } 784 785 AutoMutex lock(mLock); 786 mStreamTypes[stream].mute = muted; 787 for (uint32_t i = 0; i < mPlaybackThreads.size(); i++) 788 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 789 790 return NO_ERROR; 791} 792 793float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 794{ 795 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 796 return 0.0f; 797 } 798 799 AutoMutex lock(mLock); 800 float volume; 801 if (output) { 802 PlaybackThread *thread = checkPlaybackThread_l(output); 803 if (thread == NULL) { 804 return 0.0f; 805 } 806 volume = thread->streamVolume(stream); 807 } else { 808 volume = streamVolume_l(stream); 809 } 810 811 return volume; 812} 813 814bool AudioFlinger::streamMute(audio_stream_type_t stream) const 815{ 816 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 817 return true; 818 } 819 820 AutoMutex lock(mLock); 821 return streamMute_l(stream); 822} 823 824status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 825{ 826 ALOGV("setParameters(): io %d, keyvalue %s, tid %d, calling pid %d", 827 ioHandle, keyValuePairs.string(), gettid(), IPCThreadState::self()->getCallingPid()); 828 // check calling permissions 829 if (!settingsAllowed()) { 830 return PERMISSION_DENIED; 831 } 832 833 // ioHandle == 0 means the parameters are global to the audio hardware interface 834 if (ioHandle == 0) { 835 status_t final_result = NO_ERROR; 836 { 837 AutoMutex lock(mHardwareLock); 838 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 839 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 840 audio_hw_device_t *dev = mAudioHwDevs[i]; 841 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 842 final_result = result ?: final_result; 843 } 844 mHardwareStatus = AUDIO_HW_IDLE; 845 } 846 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 847 AudioParameter param = AudioParameter(keyValuePairs); 848 String8 value; 849 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 850 Mutex::Autolock _l(mLock); 851 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 852 if (mBtNrecIsOff != btNrecIsOff) { 853 for (size_t i = 0; i < mRecordThreads.size(); i++) { 854 sp<RecordThread> thread = mRecordThreads.valueAt(i); 855 RecordThread::RecordTrack *track = thread->track(); 856 if (track != NULL) { 857 audio_devices_t device = (audio_devices_t)( 858 thread->device() & AUDIO_DEVICE_IN_ALL); 859 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 860 thread->setEffectSuspended(FX_IID_AEC, 861 suspend, 862 track->sessionId()); 863 thread->setEffectSuspended(FX_IID_NS, 864 suspend, 865 track->sessionId()); 866 } 867 } 868 mBtNrecIsOff = btNrecIsOff; 869 } 870 } 871 return final_result; 872 } 873 874 // hold a strong ref on thread in case closeOutput() or closeInput() is called 875 // and the thread is exited once the lock is released 876 sp<ThreadBase> thread; 877 { 878 Mutex::Autolock _l(mLock); 879 thread = checkPlaybackThread_l(ioHandle); 880 if (thread == NULL) { 881 thread = checkRecordThread_l(ioHandle); 882 } else if (thread == primaryPlaybackThread_l()) { 883 // indicate output device change to all input threads for pre processing 884 AudioParameter param = AudioParameter(keyValuePairs); 885 int value; 886 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 887 for (size_t i = 0; i < mRecordThreads.size(); i++) { 888 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 889 } 890 } 891 } 892 } 893 if (thread != 0) { 894 return thread->setParameters(keyValuePairs); 895 } 896 return BAD_VALUE; 897} 898 899String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 900{ 901// ALOGV("getParameters() io %d, keys %s, tid %d, calling pid %d", 902// ioHandle, keys.string(), gettid(), IPCThreadState::self()->getCallingPid()); 903 904 if (ioHandle == 0) { 905 String8 out_s8; 906 907 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 908 char *s; 909 { 910 AutoMutex lock(mHardwareLock); 911 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 912 audio_hw_device_t *dev = mAudioHwDevs[i]; 913 s = dev->get_parameters(dev, keys.string()); 914 mHardwareStatus = AUDIO_HW_IDLE; 915 } 916 out_s8 += String8(s ? s : ""); 917 free(s); 918 } 919 return out_s8; 920 } 921 922 Mutex::Autolock _l(mLock); 923 924 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 925 if (playbackThread != NULL) { 926 return playbackThread->getParameters(keys); 927 } 928 RecordThread *recordThread = checkRecordThread_l(ioHandle); 929 if (recordThread != NULL) { 930 return recordThread->getParameters(keys); 931 } 932 return String8(""); 933} 934 935size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, int channelCount) const 936{ 937 status_t ret = initCheck(); 938 if (ret != NO_ERROR) { 939 return 0; 940 } 941 942 AutoMutex lock(mHardwareLock); 943 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 944 size_t size = mPrimaryHardwareDev->get_input_buffer_size(mPrimaryHardwareDev, sampleRate, format, channelCount); 945 mHardwareStatus = AUDIO_HW_IDLE; 946 return size; 947} 948 949unsigned int AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 950{ 951 if (ioHandle == 0) { 952 return 0; 953 } 954 955 Mutex::Autolock _l(mLock); 956 957 RecordThread *recordThread = checkRecordThread_l(ioHandle); 958 if (recordThread != NULL) { 959 return recordThread->getInputFramesLost(); 960 } 961 return 0; 962} 963 964status_t AudioFlinger::setVoiceVolume(float value) 965{ 966 status_t ret = initCheck(); 967 if (ret != NO_ERROR) { 968 return ret; 969 } 970 971 // check calling permissions 972 if (!settingsAllowed()) { 973 return PERMISSION_DENIED; 974 } 975 976 AutoMutex lock(mHardwareLock); 977 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 978 ret = mPrimaryHardwareDev->set_voice_volume(mPrimaryHardwareDev, value); 979 mHardwareStatus = AUDIO_HW_IDLE; 980 981 return ret; 982} 983 984status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 985 audio_io_handle_t output) const 986{ 987 status_t status; 988 989 Mutex::Autolock _l(mLock); 990 991 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 992 if (playbackThread != NULL) { 993 return playbackThread->getRenderPosition(halFrames, dspFrames); 994 } 995 996 return BAD_VALUE; 997} 998 999void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1000{ 1001 1002 Mutex::Autolock _l(mLock); 1003 1004 pid_t pid = IPCThreadState::self()->getCallingPid(); 1005 if (mNotificationClients.indexOfKey(pid) < 0) { 1006 sp<NotificationClient> notificationClient = new NotificationClient(this, 1007 client, 1008 pid); 1009 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1010 1011 mNotificationClients.add(pid, notificationClient); 1012 1013 sp<IBinder> binder = client->asBinder(); 1014 binder->linkToDeath(notificationClient); 1015 1016 // the config change is always sent from playback or record threads to avoid deadlock 1017 // with AudioSystem::gLock 1018 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1019 mPlaybackThreads.valueAt(i)->sendConfigEvent(AudioSystem::OUTPUT_OPENED); 1020 } 1021 1022 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1023 mRecordThreads.valueAt(i)->sendConfigEvent(AudioSystem::INPUT_OPENED); 1024 } 1025 } 1026} 1027 1028void AudioFlinger::removeNotificationClient(pid_t pid) 1029{ 1030 Mutex::Autolock _l(mLock); 1031 1032 mNotificationClients.removeItem(pid); 1033 1034 ALOGV("%d died, releasing its sessions", pid); 1035 size_t num = mAudioSessionRefs.size(); 1036 bool removed = false; 1037 for (size_t i = 0; i< num; ) { 1038 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1039 ALOGV(" pid %d @ %d", ref->mPid, i); 1040 if (ref->mPid == pid) { 1041 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1042 mAudioSessionRefs.removeAt(i); 1043 delete ref; 1044 removed = true; 1045 num--; 1046 } else { 1047 i++; 1048 } 1049 } 1050 if (removed) { 1051 purgeStaleEffects_l(); 1052 } 1053} 1054 1055// audioConfigChanged_l() must be called with AudioFlinger::mLock held 1056void AudioFlinger::audioConfigChanged_l(int event, audio_io_handle_t ioHandle, const void *param2) 1057{ 1058 size_t size = mNotificationClients.size(); 1059 for (size_t i = 0; i < size; i++) { 1060 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioHandle, 1061 param2); 1062 } 1063} 1064 1065// removeClient_l() must be called with AudioFlinger::mLock held 1066void AudioFlinger::removeClient_l(pid_t pid) 1067{ 1068 ALOGV("removeClient_l() pid %d, tid %d, calling tid %d", pid, gettid(), IPCThreadState::self()->getCallingPid()); 1069 mClients.removeItem(pid); 1070} 1071 1072 1073// ---------------------------------------------------------------------------- 1074 1075AudioFlinger::ThreadBase::ThreadBase(const sp<AudioFlinger>& audioFlinger, audio_io_handle_t id, 1076 uint32_t device, type_t type) 1077 : Thread(false), 1078 mType(type), 1079 mAudioFlinger(audioFlinger), mSampleRate(0), mFrameCount(0), 1080 // mChannelMask 1081 mChannelCount(0), 1082 mFrameSize(1), mFormat(AUDIO_FORMAT_INVALID), 1083 mParamStatus(NO_ERROR), 1084 mStandby(false), mId(id), 1085 mDevice(device), 1086 mDeathRecipient(new PMDeathRecipient(this)) 1087{ 1088} 1089 1090AudioFlinger::ThreadBase::~ThreadBase() 1091{ 1092 mParamCond.broadcast(); 1093 // do not lock the mutex in destructor 1094 releaseWakeLock_l(); 1095 if (mPowerManager != 0) { 1096 sp<IBinder> binder = mPowerManager->asBinder(); 1097 binder->unlinkToDeath(mDeathRecipient); 1098 } 1099} 1100 1101void AudioFlinger::ThreadBase::exit() 1102{ 1103 ALOGV("ThreadBase::exit"); 1104 { 1105 // This lock prevents the following race in thread (uniprocessor for illustration): 1106 // if (!exitPending()) { 1107 // // context switch from here to exit() 1108 // // exit() calls requestExit(), what exitPending() observes 1109 // // exit() calls signal(), which is dropped since no waiters 1110 // // context switch back from exit() to here 1111 // mWaitWorkCV.wait(...); 1112 // // now thread is hung 1113 // } 1114 AutoMutex lock(mLock); 1115 requestExit(); 1116 mWaitWorkCV.signal(); 1117 } 1118 // When Thread::requestExitAndWait is made virtual and this method is renamed to 1119 // "virtual status_t requestExitAndWait()", replace by "return Thread::requestExitAndWait();" 1120 requestExitAndWait(); 1121} 1122 1123status_t AudioFlinger::ThreadBase::setParameters(const String8& keyValuePairs) 1124{ 1125 status_t status; 1126 1127 ALOGV("ThreadBase::setParameters() %s", keyValuePairs.string()); 1128 Mutex::Autolock _l(mLock); 1129 1130 mNewParameters.add(keyValuePairs); 1131 mWaitWorkCV.signal(); 1132 // wait condition with timeout in case the thread loop has exited 1133 // before the request could be processed 1134 if (mParamCond.waitRelative(mLock, kSetParametersTimeoutNs) == NO_ERROR) { 1135 status = mParamStatus; 1136 mWaitWorkCV.signal(); 1137 } else { 1138 status = TIMED_OUT; 1139 } 1140 return status; 1141} 1142 1143void AudioFlinger::ThreadBase::sendConfigEvent(int event, int param) 1144{ 1145 Mutex::Autolock _l(mLock); 1146 sendConfigEvent_l(event, param); 1147} 1148 1149// sendConfigEvent_l() must be called with ThreadBase::mLock held 1150void AudioFlinger::ThreadBase::sendConfigEvent_l(int event, int param) 1151{ 1152 ConfigEvent configEvent; 1153 configEvent.mEvent = event; 1154 configEvent.mParam = param; 1155 mConfigEvents.add(configEvent); 1156 ALOGV("sendConfigEvent() num events %d event %d, param %d", mConfigEvents.size(), event, param); 1157 mWaitWorkCV.signal(); 1158} 1159 1160void AudioFlinger::ThreadBase::processConfigEvents() 1161{ 1162 mLock.lock(); 1163 while(!mConfigEvents.isEmpty()) { 1164 ALOGV("processConfigEvents() remaining events %d", mConfigEvents.size()); 1165 ConfigEvent configEvent = mConfigEvents[0]; 1166 mConfigEvents.removeAt(0); 1167 // release mLock before locking AudioFlinger mLock: lock order is always 1168 // AudioFlinger then ThreadBase to avoid cross deadlock 1169 mLock.unlock(); 1170 mAudioFlinger->mLock.lock(); 1171 audioConfigChanged_l(configEvent.mEvent, configEvent.mParam); 1172 mAudioFlinger->mLock.unlock(); 1173 mLock.lock(); 1174 } 1175 mLock.unlock(); 1176} 1177 1178status_t AudioFlinger::ThreadBase::dumpBase(int fd, const Vector<String16>& args) 1179{ 1180 const size_t SIZE = 256; 1181 char buffer[SIZE]; 1182 String8 result; 1183 1184 bool locked = tryLock(mLock); 1185 if (!locked) { 1186 snprintf(buffer, SIZE, "thread %p maybe dead locked\n", this); 1187 write(fd, buffer, strlen(buffer)); 1188 } 1189 1190 snprintf(buffer, SIZE, "standby: %d\n", mStandby); 1191 result.append(buffer); 1192 snprintf(buffer, SIZE, "Sample rate: %d\n", mSampleRate); 1193 result.append(buffer); 1194 snprintf(buffer, SIZE, "Frame count: %d\n", mFrameCount); 1195 result.append(buffer); 1196 snprintf(buffer, SIZE, "Channel Count: %d\n", mChannelCount); 1197 result.append(buffer); 1198 snprintf(buffer, SIZE, "Channel Mask: 0x%08x\n", mChannelMask); 1199 result.append(buffer); 1200 snprintf(buffer, SIZE, "Format: %d\n", mFormat); 1201 result.append(buffer); 1202 snprintf(buffer, SIZE, "Frame size: %u\n", mFrameSize); 1203 result.append(buffer); 1204 1205 snprintf(buffer, SIZE, "\nPending setParameters commands: \n"); 1206 result.append(buffer); 1207 result.append(" Index Command"); 1208 for (size_t i = 0; i < mNewParameters.size(); ++i) { 1209 snprintf(buffer, SIZE, "\n %02d ", i); 1210 result.append(buffer); 1211 result.append(mNewParameters[i]); 1212 } 1213 1214 snprintf(buffer, SIZE, "\n\nPending config events: \n"); 1215 result.append(buffer); 1216 snprintf(buffer, SIZE, " Index event param\n"); 1217 result.append(buffer); 1218 for (size_t i = 0; i < mConfigEvents.size(); i++) { 1219 snprintf(buffer, SIZE, " %02d %02d %d\n", i, mConfigEvents[i].mEvent, mConfigEvents[i].mParam); 1220 result.append(buffer); 1221 } 1222 result.append("\n"); 1223 1224 write(fd, result.string(), result.size()); 1225 1226 if (locked) { 1227 mLock.unlock(); 1228 } 1229 return NO_ERROR; 1230} 1231 1232status_t AudioFlinger::ThreadBase::dumpEffectChains(int fd, const Vector<String16>& args) 1233{ 1234 const size_t SIZE = 256; 1235 char buffer[SIZE]; 1236 String8 result; 1237 1238 snprintf(buffer, SIZE, "\n- %d Effect Chains:\n", mEffectChains.size()); 1239 write(fd, buffer, strlen(buffer)); 1240 1241 for (size_t i = 0; i < mEffectChains.size(); ++i) { 1242 sp<EffectChain> chain = mEffectChains[i]; 1243 if (chain != 0) { 1244 chain->dump(fd, args); 1245 } 1246 } 1247 return NO_ERROR; 1248} 1249 1250void AudioFlinger::ThreadBase::acquireWakeLock() 1251{ 1252 Mutex::Autolock _l(mLock); 1253 acquireWakeLock_l(); 1254} 1255 1256void AudioFlinger::ThreadBase::acquireWakeLock_l() 1257{ 1258 if (mPowerManager == 0) { 1259 // use checkService() to avoid blocking if power service is not up yet 1260 sp<IBinder> binder = 1261 defaultServiceManager()->checkService(String16("power")); 1262 if (binder == 0) { 1263 ALOGW("Thread %s cannot connect to the power manager service", mName); 1264 } else { 1265 mPowerManager = interface_cast<IPowerManager>(binder); 1266 binder->linkToDeath(mDeathRecipient); 1267 } 1268 } 1269 if (mPowerManager != 0) { 1270 sp<IBinder> binder = new BBinder(); 1271 status_t status = mPowerManager->acquireWakeLock(POWERMANAGER_PARTIAL_WAKE_LOCK, 1272 binder, 1273 String16(mName)); 1274 if (status == NO_ERROR) { 1275 mWakeLockToken = binder; 1276 } 1277 ALOGV("acquireWakeLock_l() %s status %d", mName, status); 1278 } 1279} 1280 1281void AudioFlinger::ThreadBase::releaseWakeLock() 1282{ 1283 Mutex::Autolock _l(mLock); 1284 releaseWakeLock_l(); 1285} 1286 1287void AudioFlinger::ThreadBase::releaseWakeLock_l() 1288{ 1289 if (mWakeLockToken != 0) { 1290 ALOGV("releaseWakeLock_l() %s", mName); 1291 if (mPowerManager != 0) { 1292 mPowerManager->releaseWakeLock(mWakeLockToken, 0); 1293 } 1294 mWakeLockToken.clear(); 1295 } 1296} 1297 1298void AudioFlinger::ThreadBase::clearPowerManager() 1299{ 1300 Mutex::Autolock _l(mLock); 1301 releaseWakeLock_l(); 1302 mPowerManager.clear(); 1303} 1304 1305void AudioFlinger::ThreadBase::PMDeathRecipient::binderDied(const wp<IBinder>& who) 1306{ 1307 sp<ThreadBase> thread = mThread.promote(); 1308 if (thread != 0) { 1309 thread->clearPowerManager(); 1310 } 1311 ALOGW("power manager service died !!!"); 1312} 1313 1314void AudioFlinger::ThreadBase::setEffectSuspended( 1315 const effect_uuid_t *type, bool suspend, int sessionId) 1316{ 1317 Mutex::Autolock _l(mLock); 1318 setEffectSuspended_l(type, suspend, sessionId); 1319} 1320 1321void AudioFlinger::ThreadBase::setEffectSuspended_l( 1322 const effect_uuid_t *type, bool suspend, int sessionId) 1323{ 1324 sp<EffectChain> chain = getEffectChain_l(sessionId); 1325 if (chain != 0) { 1326 if (type != NULL) { 1327 chain->setEffectSuspended_l(type, suspend); 1328 } else { 1329 chain->setEffectSuspendedAll_l(suspend); 1330 } 1331 } 1332 1333 updateSuspendedSessions_l(type, suspend, sessionId); 1334} 1335 1336void AudioFlinger::ThreadBase::checkSuspendOnAddEffectChain_l(const sp<EffectChain>& chain) 1337{ 1338 ssize_t index = mSuspendedSessions.indexOfKey(chain->sessionId()); 1339 if (index < 0) { 1340 return; 1341 } 1342 1343 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects = 1344 mSuspendedSessions.editValueAt(index); 1345 1346 for (size_t i = 0; i < sessionEffects.size(); i++) { 1347 sp <SuspendedSessionDesc> desc = sessionEffects.valueAt(i); 1348 for (int j = 0; j < desc->mRefCount; j++) { 1349 if (sessionEffects.keyAt(i) == EffectChain::kKeyForSuspendAll) { 1350 chain->setEffectSuspendedAll_l(true); 1351 } else { 1352 ALOGV("checkSuspendOnAddEffectChain_l() suspending effects %08x", 1353 desc->mType.timeLow); 1354 chain->setEffectSuspended_l(&desc->mType, true); 1355 } 1356 } 1357 } 1358} 1359 1360void AudioFlinger::ThreadBase::updateSuspendedSessions_l(const effect_uuid_t *type, 1361 bool suspend, 1362 int sessionId) 1363{ 1364 ssize_t index = mSuspendedSessions.indexOfKey(sessionId); 1365 1366 KeyedVector <int, sp<SuspendedSessionDesc> > sessionEffects; 1367 1368 if (suspend) { 1369 if (index >= 0) { 1370 sessionEffects = mSuspendedSessions.editValueAt(index); 1371 } else { 1372 mSuspendedSessions.add(sessionId, sessionEffects); 1373 } 1374 } else { 1375 if (index < 0) { 1376 return; 1377 } 1378 sessionEffects = mSuspendedSessions.editValueAt(index); 1379 } 1380 1381 1382 int key = EffectChain::kKeyForSuspendAll; 1383 if (type != NULL) { 1384 key = type->timeLow; 1385 } 1386 index = sessionEffects.indexOfKey(key); 1387 1388 sp <SuspendedSessionDesc> desc; 1389 if (suspend) { 1390 if (index >= 0) { 1391 desc = sessionEffects.valueAt(index); 1392 } else { 1393 desc = new SuspendedSessionDesc(); 1394 if (type != NULL) { 1395 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 1396 } 1397 sessionEffects.add(key, desc); 1398 ALOGV("updateSuspendedSessions_l() suspend adding effect %08x", key); 1399 } 1400 desc->mRefCount++; 1401 } else { 1402 if (index < 0) { 1403 return; 1404 } 1405 desc = sessionEffects.valueAt(index); 1406 if (--desc->mRefCount == 0) { 1407 ALOGV("updateSuspendedSessions_l() restore removing effect %08x", key); 1408 sessionEffects.removeItemsAt(index); 1409 if (sessionEffects.isEmpty()) { 1410 ALOGV("updateSuspendedSessions_l() restore removing session %d", 1411 sessionId); 1412 mSuspendedSessions.removeItem(sessionId); 1413 } 1414 } 1415 } 1416 if (!sessionEffects.isEmpty()) { 1417 mSuspendedSessions.replaceValueFor(sessionId, sessionEffects); 1418 } 1419} 1420 1421void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 1422 bool enabled, 1423 int sessionId) 1424{ 1425 Mutex::Autolock _l(mLock); 1426 checkSuspendOnEffectEnabled_l(effect, enabled, sessionId); 1427} 1428 1429void AudioFlinger::ThreadBase::checkSuspendOnEffectEnabled_l(const sp<EffectModule>& effect, 1430 bool enabled, 1431 int sessionId) 1432{ 1433 if (mType != RECORD) { 1434 // suspend all effects in AUDIO_SESSION_OUTPUT_MIX when enabling any effect on 1435 // another session. This gives the priority to well behaved effect control panels 1436 // and applications not using global effects. 1437 if (sessionId != AUDIO_SESSION_OUTPUT_MIX) { 1438 setEffectSuspended_l(NULL, enabled, AUDIO_SESSION_OUTPUT_MIX); 1439 } 1440 } 1441 1442 sp<EffectChain> chain = getEffectChain_l(sessionId); 1443 if (chain != 0) { 1444 chain->checkSuspendOnEffectEnabled(effect, enabled); 1445 } 1446} 1447 1448// ---------------------------------------------------------------------------- 1449 1450AudioFlinger::PlaybackThread::PlaybackThread(const sp<AudioFlinger>& audioFlinger, 1451 AudioStreamOut* output, 1452 audio_io_handle_t id, 1453 uint32_t device, 1454 type_t type) 1455 : ThreadBase(audioFlinger, id, device, type), 1456 mMixBuffer(NULL), mSuspended(0), mBytesWritten(0), 1457 // Assumes constructor is called by AudioFlinger with it's mLock held, 1458 // but it would be safer to explicitly pass initial masterMute as parameter 1459 mMasterMute(audioFlinger->masterMute_l()), 1460 // mStreamTypes[] initialized in constructor body 1461 mOutput(output), 1462 // Assumes constructor is called by AudioFlinger with it's mLock held, 1463 // but it would be safer to explicitly pass initial masterVolume as parameter 1464 mMasterVolume(audioFlinger->masterVolumeSW_l()), 1465 mLastWriteTime(0), mNumWrites(0), mNumDelayedWrites(0), mInWrite(false) 1466{ 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 2046 mixerStatus = MIXER_IDLE; 2047 { // scope for mLock 2048 2049 Mutex::Autolock _l(mLock); 2050 2051 if (checkForNewParameters_l()) { 2052 mixBufferSize = mFrameCount * mFrameSize; 2053 2054if (mType == MIXER) { 2055 // FIXME: Relaxed timing because of a certain device that can't meet latency 2056 // Should be reduced to 2x after the vendor fixes the driver issue 2057 // increase threshold again due to low power audio mode. The way this warning 2058 // threshold is calculated and its usefulness should be reconsidered anyway. 2059 maxPeriod = seconds(mFrameCount) / mSampleRate * 15; 2060} 2061 2062 updateWaitTime_l(); 2063 2064 activeSleepTime = activeSleepTimeUs(); 2065 idleSleepTime = idleSleepTimeUs(); 2066 2067if (mType == DIRECT) { 2068 standbyDelay = microseconds(activeSleepTime*2); 2069} 2070 2071 } 2072 2073 saveOutputTracks(); 2074 2075 // put audio hardware into standby after short delay 2076 if (CC_UNLIKELY((!mActiveTracks.size() && systemTime() > standbyTime) || 2077 mSuspended > 0)) { 2078 if (!mStandby) { 2079 2080 threadLoop_standby(); 2081 2082 mStandby = true; 2083 mBytesWritten = 0; 2084 } 2085 2086 if (!mActiveTracks.size() && mConfigEvents.isEmpty()) { 2087 // we're about to wait, flush the binder command buffer 2088 IPCThreadState::self()->flushCommands(); 2089 2090 clearOutputTracks(); 2091 2092 if (exitPending()) break; 2093 2094 releaseWakeLock_l(); 2095 // wait until we have something to do... 2096 ALOGV("Thread %p type %d TID %d going to sleep", this, mType, gettid()); 2097 mWaitWorkCV.wait(mLock); 2098 ALOGV("Thread %p type %d TID %d waking up", this, mType, gettid()); 2099 acquireWakeLock_l(); 2100 2101if (mType == MIXER || mType == DUPLICATING) { 2102 mPrevMixerStatus = MIXER_IDLE; 2103} 2104 2105 checkSilentMode_l(); 2106 2107if (mType == MIXER || mType == DUPLICATING) { 2108 standbyTime = systemTime() + mStandbyTimeInNsecs; 2109} 2110 2111if (mType == DIRECT) { 2112 standbyTime = systemTime() + standbyDelay; 2113} 2114 2115 sleepTime = idleSleepTime; 2116 2117if (mType == MIXER) { 2118 sleepTimeShift = 0; 2119} 2120 2121 continue; 2122 } 2123 } 2124 2125 mixerStatus = prepareTracks_l(&tracksToRemove); 2126 // see FIXME in AudioFlinger.h 2127 if (mixerStatus == MIXER_CONTINUE) { 2128 continue; 2129 } 2130 2131 // prevent any changes in effect chain list and in each effect chain 2132 // during mixing and effect process as the audio buffers could be deleted 2133 // or modified if an effect is created or deleted 2134 lockEffectChains_l(effectChains); 2135 } 2136 2137 if (CC_LIKELY(mixerStatus == MIXER_TRACKS_READY)) { 2138 threadLoop_mix(); 2139 } else { 2140 threadLoop_sleepTime(); 2141 } 2142 2143 if (mSuspended > 0) { 2144 sleepTime = suspendSleepTimeUs(); 2145 } 2146 2147 // only process effects if we're going to write 2148 if (sleepTime == 0) { 2149 for (size_t i = 0; i < effectChains.size(); i ++) { 2150 effectChains[i]->process_l(); 2151 } 2152 } 2153 2154 // enable changes in effect chain 2155 unlockEffectChains(effectChains); 2156 2157 // sleepTime == 0 means we must write to audio hardware 2158 if (sleepTime == 0) { 2159 2160 threadLoop_write(); 2161 2162if (mType == MIXER) { 2163 // write blocked detection 2164 nsecs_t now = systemTime(); 2165 nsecs_t delta = now - mLastWriteTime; 2166 if (!mStandby && delta > maxPeriod) { 2167 mNumDelayedWrites++; 2168 if ((now - lastWarning) > kWarningThrottleNs) { 2169 ALOGW("write blocked for %llu msecs, %d delayed writes, thread %p", 2170 ns2ms(delta), mNumDelayedWrites, this); 2171 lastWarning = now; 2172 } 2173 // FIXME this is broken: longStandbyExit should be handled out of the if() and with 2174 // a different threshold. Or completely removed for what it is worth anyway... 2175 if (mStandby) { 2176 longStandbyExit = true; 2177 } 2178 } 2179} 2180 2181 mStandby = false; 2182 } else { 2183 usleep(sleepTime); 2184 } 2185 2186 // finally let go of removed track(s), without the lock held 2187 // since we can't guarantee the destructors won't acquire that 2188 // same lock. 2189 tracksToRemove.clear(); 2190 2191 // FIXME I don't understand the need for this here; 2192 // it was in the original code but maybe the 2193 // assignment in saveOutputTracks() makes this unnecessary? 2194 clearOutputTracks(); 2195 2196 // Effect chains will be actually deleted here if they were removed from 2197 // mEffectChains list during mixing or effects processing 2198 effectChains.clear(); 2199 2200 // FIXME Note that the above .clear() is no longer necessary since effectChains 2201 // is now local to this block, but will keep it for now (at least until merge done). 2202 } 2203 2204if (mType == MIXER || mType == DIRECT) { 2205 // put output stream into standby mode 2206 if (!mStandby) { 2207 mOutput->stream->common.standby(&mOutput->stream->common); 2208 } 2209} 2210if (mType == DUPLICATING) { 2211 // for DuplicatingThread, standby mode is handled by the outputTracks 2212} 2213 2214 releaseWakeLock(); 2215 2216 ALOGV("Thread %p type %d exiting", this, mType); 2217 return false; 2218} 2219 2220// shared by MIXER and DIRECT, overridden by DUPLICATING 2221void AudioFlinger::PlaybackThread::threadLoop_write() 2222{ 2223 // FIXME rewrite to reduce number of system calls 2224 mLastWriteTime = systemTime(); 2225 mInWrite = true; 2226 mBytesWritten += mixBufferSize; 2227 int bytesWritten = (int)mOutput->stream->write(mOutput->stream, mMixBuffer, mixBufferSize); 2228 if (bytesWritten < 0) mBytesWritten -= mixBufferSize; 2229 mNumWrites++; 2230 mInWrite = false; 2231} 2232 2233// shared by MIXER and DIRECT, overridden by DUPLICATING 2234void AudioFlinger::PlaybackThread::threadLoop_standby() 2235{ 2236 ALOGV("Audio hardware entering standby, mixer %p, suspend count %u", this, mSuspended); 2237 mOutput->stream->common.standby(&mOutput->stream->common); 2238} 2239 2240void AudioFlinger::MixerThread::threadLoop_mix() 2241{ 2242 // obtain the presentation timestamp of the next output buffer 2243 int64_t pts; 2244 status_t status = INVALID_OPERATION; 2245 2246 if (NULL != mOutput->stream->get_next_write_timestamp) { 2247 status = mOutput->stream->get_next_write_timestamp( 2248 mOutput->stream, &pts); 2249 } 2250 2251 if (status != NO_ERROR) { 2252 pts = AudioBufferProvider::kInvalidPTS; 2253 } 2254 2255 // mix buffers... 2256 mAudioMixer->process(pts); 2257 // increase sleep time progressively when application underrun condition clears. 2258 // Only increase sleep time if the mixer is ready for two consecutive times to avoid 2259 // that a steady state of alternating ready/not ready conditions keeps the sleep time 2260 // such that we would underrun the audio HAL. 2261 if ((sleepTime == 0) && (sleepTimeShift > 0)) { 2262 sleepTimeShift--; 2263 } 2264 sleepTime = 0; 2265 standbyTime = systemTime() + mStandbyTimeInNsecs; 2266 //TODO: delay standby when effects have a tail 2267} 2268 2269void AudioFlinger::MixerThread::threadLoop_sleepTime() 2270{ 2271 // If no tracks are ready, sleep once for the duration of an output 2272 // buffer size, then write 0s to the output 2273 if (sleepTime == 0) { 2274 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2275 sleepTime = activeSleepTime >> sleepTimeShift; 2276 if (sleepTime < kMinThreadSleepTimeUs) { 2277 sleepTime = kMinThreadSleepTimeUs; 2278 } 2279 // reduce sleep time in case of consecutive application underruns to avoid 2280 // starving the audio HAL. As activeSleepTimeUs() is larger than a buffer 2281 // duration we would end up writing less data than needed by the audio HAL if 2282 // the condition persists. 2283 if (sleepTimeShift < kMaxThreadSleepTimeShift) { 2284 sleepTimeShift++; 2285 } 2286 } else { 2287 sleepTime = idleSleepTime; 2288 } 2289 } else if (mBytesWritten != 0 || 2290 (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)) { 2291 memset (mMixBuffer, 0, mixBufferSize); 2292 sleepTime = 0; 2293 ALOGV_IF((mBytesWritten == 0 && (mixerStatus == MIXER_TRACKS_ENABLED && longStandbyExit)), "anticipated start"); 2294 } 2295 // TODO add standby time extension fct of effect tail 2296} 2297 2298// prepareTracks_l() must be called with ThreadBase::mLock held 2299AudioFlinger::PlaybackThread::mixer_state AudioFlinger::MixerThread::prepareTracks_l( 2300 Vector< sp<Track> > *tracksToRemove) 2301{ 2302 2303 mixer_state mixerStatus = MIXER_IDLE; 2304 // find out which tracks need to be processed 2305 size_t count = mActiveTracks.size(); 2306 size_t mixedTracks = 0; 2307 size_t tracksWithEffect = 0; 2308 2309 float masterVolume = mMasterVolume; 2310 bool masterMute = mMasterMute; 2311 2312 if (masterMute) { 2313 masterVolume = 0; 2314 } 2315 // Delegate master volume control to effect in output mix effect chain if needed 2316 sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2317 if (chain != 0) { 2318 uint32_t v = (uint32_t)(masterVolume * (1 << 24)); 2319 chain->setVolume_l(&v, &v); 2320 masterVolume = (float)((v + (1 << 23)) >> 24); 2321 chain.clear(); 2322 } 2323 2324 for (size_t i=0 ; i<count ; i++) { 2325 sp<Track> t = mActiveTracks[i].promote(); 2326 if (t == 0) continue; 2327 2328 // this const just means the local variable doesn't change 2329 Track* const track = t.get(); 2330 audio_track_cblk_t* cblk = track->cblk(); 2331 2332 // The first time a track is added we wait 2333 // for all its buffers to be filled before processing it 2334 int name = track->name(); 2335 // make sure that we have enough frames to mix one full buffer. 2336 // enforce this condition only once to enable draining the buffer in case the client 2337 // app does not call stop() and relies on underrun to stop: 2338 // hence the test on (mPrevMixerStatus == MIXER_TRACKS_READY) meaning the track was mixed 2339 // during last round 2340 uint32_t minFrames = 1; 2341 if (!track->isStopped() && !track->isPausing() && 2342 (mPrevMixerStatus == MIXER_TRACKS_READY)) { 2343 if (t->sampleRate() == (int)mSampleRate) { 2344 minFrames = mFrameCount; 2345 } else { 2346 // +1 for rounding and +1 for additional sample needed for interpolation 2347 minFrames = (mFrameCount * t->sampleRate()) / mSampleRate + 1 + 1; 2348 // add frames already consumed but not yet released by the resampler 2349 // because cblk->framesReady() will include these frames 2350 minFrames += mAudioMixer->getUnreleasedFrames(track->name()); 2351 // the minimum track buffer size is normally twice the number of frames necessary 2352 // to fill one buffer and the resampler should not leave more than one buffer worth 2353 // of unreleased frames after each pass, but just in case... 2354 ALOG_ASSERT(minFrames <= cblk->frameCount); 2355 } 2356 } 2357 if ((track->framesReady() >= minFrames) && track->isReady() && 2358 !track->isPaused() && !track->isTerminated()) 2359 { 2360 //ALOGV("track %d u=%08x, s=%08x [OK] on thread %p", name, cblk->user, cblk->server, this); 2361 2362 mixedTracks++; 2363 2364 // track->mainBuffer() != mMixBuffer means there is an effect chain 2365 // connected to the track 2366 chain.clear(); 2367 if (track->mainBuffer() != mMixBuffer) { 2368 chain = getEffectChain_l(track->sessionId()); 2369 // Delegate volume control to effect in track effect chain if needed 2370 if (chain != 0) { 2371 tracksWithEffect++; 2372 } else { 2373 ALOGW("prepareTracks_l(): track %d attached to effect but no chain found on session %d", 2374 name, track->sessionId()); 2375 } 2376 } 2377 2378 2379 int param = AudioMixer::VOLUME; 2380 if (track->mFillingUpStatus == Track::FS_FILLED) { 2381 // no ramp for the first volume setting 2382 track->mFillingUpStatus = Track::FS_ACTIVE; 2383 if (track->mState == TrackBase::RESUMING) { 2384 track->mState = TrackBase::ACTIVE; 2385 param = AudioMixer::RAMP_VOLUME; 2386 } 2387 mAudioMixer->setParameter(name, AudioMixer::RESAMPLE, AudioMixer::RESET, NULL); 2388 } else if (cblk->server != 0) { 2389 // If the track is stopped before the first frame was mixed, 2390 // do not apply ramp 2391 param = AudioMixer::RAMP_VOLUME; 2392 } 2393 2394 // compute volume for this track 2395 uint32_t vl, vr, va; 2396 if (track->isMuted() || track->isPausing() || 2397 mStreamTypes[track->streamType()].mute) { 2398 vl = vr = va = 0; 2399 if (track->isPausing()) { 2400 track->setPaused(); 2401 } 2402 } else { 2403 2404 // read original volumes with volume control 2405 float typeVolume = mStreamTypes[track->streamType()].volume; 2406 float v = masterVolume * typeVolume; 2407 uint32_t vlr = cblk->getVolumeLR(); 2408 vl = vlr & 0xFFFF; 2409 vr = vlr >> 16; 2410 // track volumes come from shared memory, so can't be trusted and must be clamped 2411 if (vl > MAX_GAIN_INT) { 2412 ALOGV("Track left volume out of range: %04X", vl); 2413 vl = MAX_GAIN_INT; 2414 } 2415 if (vr > MAX_GAIN_INT) { 2416 ALOGV("Track right volume out of range: %04X", vr); 2417 vr = MAX_GAIN_INT; 2418 } 2419 // now apply the master volume and stream type volume 2420 vl = (uint32_t)(v * vl) << 12; 2421 vr = (uint32_t)(v * vr) << 12; 2422 // assuming master volume and stream type volume each go up to 1.0, 2423 // vl and vr are now in 8.24 format 2424 2425 uint16_t sendLevel = cblk->getSendLevel_U4_12(); 2426 // send level comes from shared memory and so may be corrupt 2427 if (sendLevel > MAX_GAIN_INT) { 2428 ALOGV("Track send level out of range: %04X", sendLevel); 2429 sendLevel = MAX_GAIN_INT; 2430 } 2431 va = (uint32_t)(v * sendLevel); 2432 } 2433 // Delegate volume control to effect in track effect chain if needed 2434 if (chain != 0 && chain->setVolume_l(&vl, &vr)) { 2435 // Do not ramp volume if volume is controlled by effect 2436 param = AudioMixer::VOLUME; 2437 track->mHasVolumeController = true; 2438 } else { 2439 // force no volume ramp when volume controller was just disabled or removed 2440 // from effect chain to avoid volume spike 2441 if (track->mHasVolumeController) { 2442 param = AudioMixer::VOLUME; 2443 } 2444 track->mHasVolumeController = false; 2445 } 2446 2447 // Convert volumes from 8.24 to 4.12 format 2448 // This additional clamping is needed in case chain->setVolume_l() overshot 2449 vl = (vl + (1 << 11)) >> 12; 2450 if (vl > MAX_GAIN_INT) vl = MAX_GAIN_INT; 2451 vr = (vr + (1 << 11)) >> 12; 2452 if (vr > MAX_GAIN_INT) vr = MAX_GAIN_INT; 2453 2454 if (va > MAX_GAIN_INT) va = MAX_GAIN_INT; // va is uint32_t, so no need to check for - 2455 2456 // XXX: these things DON'T need to be done each time 2457 mAudioMixer->setBufferProvider(name, track); 2458 mAudioMixer->enable(name); 2459 2460 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME0, (void *)vl); 2461 mAudioMixer->setParameter(name, param, AudioMixer::VOLUME1, (void *)vr); 2462 mAudioMixer->setParameter(name, param, AudioMixer::AUXLEVEL, (void *)va); 2463 mAudioMixer->setParameter( 2464 name, 2465 AudioMixer::TRACK, 2466 AudioMixer::FORMAT, (void *)track->format()); 2467 mAudioMixer->setParameter( 2468 name, 2469 AudioMixer::TRACK, 2470 AudioMixer::CHANNEL_MASK, (void *)track->channelMask()); 2471 mAudioMixer->setParameter( 2472 name, 2473 AudioMixer::RESAMPLE, 2474 AudioMixer::SAMPLE_RATE, 2475 (void *)(cblk->sampleRate)); 2476 mAudioMixer->setParameter( 2477 name, 2478 AudioMixer::TRACK, 2479 AudioMixer::MAIN_BUFFER, (void *)track->mainBuffer()); 2480 mAudioMixer->setParameter( 2481 name, 2482 AudioMixer::TRACK, 2483 AudioMixer::AUX_BUFFER, (void *)track->auxBuffer()); 2484 2485 // reset retry count 2486 track->mRetryCount = kMaxTrackRetries; 2487 // If one track is ready, set the mixer ready if: 2488 // - the mixer was not ready during previous round OR 2489 // - no other track is not ready 2490 if (mPrevMixerStatus != MIXER_TRACKS_READY || 2491 mixerStatus != MIXER_TRACKS_ENABLED) { 2492 mixerStatus = MIXER_TRACKS_READY; 2493 } 2494 } else { 2495 //ALOGV("track %d u=%08x, s=%08x [NOT READY] on thread %p", name, cblk->user, cblk->server, this); 2496 if (track->isStopped()) { 2497 track->reset(); 2498 } 2499 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2500 // We have consumed all the buffers of this track. 2501 // Remove it from the list of active tracks. 2502 tracksToRemove->add(track); 2503 } else { 2504 // No buffers for this track. Give it a few chances to 2505 // fill a buffer, then remove it from active list. 2506 if (--(track->mRetryCount) <= 0) { 2507 ALOGV("BUFFER TIMEOUT: remove(%d) from active list on thread %p", name, this); 2508 tracksToRemove->add(track); 2509 // indicate to client process that the track was disabled because of underrun 2510 android_atomic_or(CBLK_DISABLED_ON, &cblk->flags); 2511 // If one track is not ready, mark the mixer also not ready if: 2512 // - the mixer was ready during previous round OR 2513 // - no other track is ready 2514 } else if (mPrevMixerStatus == MIXER_TRACKS_READY || 2515 mixerStatus != MIXER_TRACKS_READY) { 2516 mixerStatus = MIXER_TRACKS_ENABLED; 2517 } 2518 } 2519 mAudioMixer->disable(name); 2520 } 2521 } 2522 2523 // remove all the tracks that need to be... 2524 count = tracksToRemove->size(); 2525 if (CC_UNLIKELY(count)) { 2526 for (size_t i=0 ; i<count ; i++) { 2527 const sp<Track>& track = tracksToRemove->itemAt(i); 2528 mActiveTracks.remove(track); 2529 if (track->mainBuffer() != mMixBuffer) { 2530 chain = getEffectChain_l(track->sessionId()); 2531 if (chain != 0) { 2532 ALOGV("stopping track on chain %p for session Id: %d", chain.get(), track->sessionId()); 2533 chain->decActiveTrackCnt(); 2534 } 2535 } 2536 if (track->isTerminated()) { 2537 removeTrack_l(track); 2538 } 2539 } 2540 } 2541 2542 // mix buffer must be cleared if all tracks are connected to an 2543 // effect chain as in this case the mixer will not write to 2544 // mix buffer and track effects will accumulate into it 2545 if (mixedTracks != 0 && mixedTracks == tracksWithEffect) { 2546 memset(mMixBuffer, 0, mFrameCount * mChannelCount * sizeof(int16_t)); 2547 } 2548 2549 mPrevMixerStatus = mixerStatus; 2550 return mixerStatus; 2551} 2552 2553void AudioFlinger::MixerThread::invalidateTracks(audio_stream_type_t streamType) 2554{ 2555 ALOGV ("MixerThread::invalidateTracks() mixer %p, streamType %d, mTracks.size %d", 2556 this, streamType, mTracks.size()); 2557 Mutex::Autolock _l(mLock); 2558 2559 size_t size = mTracks.size(); 2560 for (size_t i = 0; i < size; i++) { 2561 sp<Track> t = mTracks[i]; 2562 if (t->streamType() == streamType) { 2563 android_atomic_or(CBLK_INVALID_ON, &t->mCblk->flags); 2564 t->mCblk->cv.signal(); 2565 } 2566 } 2567} 2568 2569void AudioFlinger::PlaybackThread::setStreamValid(audio_stream_type_t streamType, bool valid) 2570{ 2571 ALOGV ("PlaybackThread::setStreamValid() thread %p, streamType %d, valid %d", 2572 this, streamType, valid); 2573 Mutex::Autolock _l(mLock); 2574 2575 mStreamTypes[streamType].valid = valid; 2576} 2577 2578// getTrackName_l() must be called with ThreadBase::mLock held 2579int AudioFlinger::MixerThread::getTrackName_l() 2580{ 2581 return mAudioMixer->getTrackName(); 2582} 2583 2584// deleteTrackName_l() must be called with ThreadBase::mLock held 2585void AudioFlinger::MixerThread::deleteTrackName_l(int name) 2586{ 2587 ALOGV("remove track (%d) and delete from mixer", name); 2588 mAudioMixer->deleteTrackName(name); 2589} 2590 2591// checkForNewParameters_l() must be called with ThreadBase::mLock held 2592bool AudioFlinger::MixerThread::checkForNewParameters_l() 2593{ 2594 bool reconfig = false; 2595 2596 while (!mNewParameters.isEmpty()) { 2597 status_t status = NO_ERROR; 2598 String8 keyValuePair = mNewParameters[0]; 2599 AudioParameter param = AudioParameter(keyValuePair); 2600 int value; 2601 2602 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 2603 reconfig = true; 2604 } 2605 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 2606 if ((audio_format_t) value != AUDIO_FORMAT_PCM_16_BIT) { 2607 status = BAD_VALUE; 2608 } else { 2609 reconfig = true; 2610 } 2611 } 2612 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 2613 if (value != AUDIO_CHANNEL_OUT_STEREO) { 2614 status = BAD_VALUE; 2615 } else { 2616 reconfig = true; 2617 } 2618 } 2619 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 2620 // do not accept frame count changes if tracks are open as the track buffer 2621 // size depends on frame count and correct behavior would not be guaranteed 2622 // if frame count is changed after track creation 2623 if (!mTracks.isEmpty()) { 2624 status = INVALID_OPERATION; 2625 } else { 2626 reconfig = true; 2627 } 2628 } 2629 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 2630 // when changing the audio output device, call addBatteryData to notify 2631 // the change 2632 if ((int)mDevice != value) { 2633 uint32_t params = 0; 2634 // check whether speaker is on 2635 if (value & AUDIO_DEVICE_OUT_SPEAKER) { 2636 params |= IMediaPlayerService::kBatteryDataSpeakerOn; 2637 } 2638 2639 int deviceWithoutSpeaker 2640 = AUDIO_DEVICE_OUT_ALL & ~AUDIO_DEVICE_OUT_SPEAKER; 2641 // check if any other device (except speaker) is on 2642 if (value & deviceWithoutSpeaker ) { 2643 params |= IMediaPlayerService::kBatteryDataOtherAudioDeviceOn; 2644 } 2645 2646 if (params != 0) { 2647 addBatteryData(params); 2648 } 2649 } 2650 2651 // forward device change to effects that have requested to be 2652 // aware of attached audio device. 2653 mDevice = (uint32_t)value; 2654 for (size_t i = 0; i < mEffectChains.size(); i++) { 2655 mEffectChains[i]->setDevice_l(mDevice); 2656 } 2657 } 2658 2659 if (status == NO_ERROR) { 2660 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2661 keyValuePair.string()); 2662 if (!mStandby && status == INVALID_OPERATION) { 2663 mOutput->stream->common.standby(&mOutput->stream->common); 2664 mStandby = true; 2665 mBytesWritten = 0; 2666 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 2667 keyValuePair.string()); 2668 } 2669 if (status == NO_ERROR && reconfig) { 2670 delete mAudioMixer; 2671 // for safety in case readOutputParameters() accesses mAudioMixer (it doesn't) 2672 mAudioMixer = NULL; 2673 readOutputParameters(); 2674 mAudioMixer = new AudioMixer(mFrameCount, mSampleRate); 2675 for (size_t i = 0; i < mTracks.size() ; i++) { 2676 int name = getTrackName_l(); 2677 if (name < 0) break; 2678 mTracks[i]->mName = name; 2679 // limit track sample rate to 2 x new output sample rate 2680 if (mTracks[i]->mCblk->sampleRate > 2 * sampleRate()) { 2681 mTracks[i]->mCblk->sampleRate = 2 * sampleRate(); 2682 } 2683 } 2684 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 2685 } 2686 } 2687 2688 mNewParameters.removeAt(0); 2689 2690 mParamStatus = status; 2691 mParamCond.signal(); 2692 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 2693 // already timed out waiting for the status and will never signal the condition. 2694 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 2695 } 2696 return reconfig; 2697} 2698 2699status_t AudioFlinger::MixerThread::dumpInternals(int fd, const Vector<String16>& args) 2700{ 2701 const size_t SIZE = 256; 2702 char buffer[SIZE]; 2703 String8 result; 2704 2705 PlaybackThread::dumpInternals(fd, args); 2706 2707 snprintf(buffer, SIZE, "AudioMixer tracks: %08x\n", mAudioMixer->trackNames()); 2708 result.append(buffer); 2709 write(fd, result.string(), result.size()); 2710 return NO_ERROR; 2711} 2712 2713uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() 2714{ 2715 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 2716} 2717 2718uint32_t AudioFlinger::MixerThread::suspendSleepTimeUs() 2719{ 2720 return (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 2721} 2722 2723// ---------------------------------------------------------------------------- 2724AudioFlinger::DirectOutputThread::DirectOutputThread(const sp<AudioFlinger>& audioFlinger, 2725 AudioStreamOut* output, audio_io_handle_t id, uint32_t device) 2726 : PlaybackThread(audioFlinger, output, id, device, DIRECT) 2727 // mLeftVolFloat, mRightVolFloat 2728 // mLeftVolShort, mRightVolShort 2729{ 2730} 2731 2732AudioFlinger::DirectOutputThread::~DirectOutputThread() 2733{ 2734} 2735 2736void AudioFlinger::DirectOutputThread::applyVolume() 2737{ 2738 // Do not apply volume on compressed audio 2739 if (!audio_is_linear_pcm(mFormat)) { 2740 return; 2741 } 2742 2743 // convert to signed 16 bit before volume calculation 2744 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2745 size_t count = mFrameCount * mChannelCount; 2746 uint8_t *src = (uint8_t *)mMixBuffer + count-1; 2747 int16_t *dst = mMixBuffer + count-1; 2748 while(count--) { 2749 *dst-- = (int16_t)(*src--^0x80) << 8; 2750 } 2751 } 2752 2753 size_t frameCount = mFrameCount; 2754 int16_t *out = mMixBuffer; 2755 if (rampVolume) { 2756 if (mChannelCount == 1) { 2757 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2758 int32_t vlInc = d / (int32_t)frameCount; 2759 int32_t vl = ((int32_t)mLeftVolShort << 16); 2760 do { 2761 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2762 out++; 2763 vl += vlInc; 2764 } while (--frameCount); 2765 2766 } else { 2767 int32_t d = ((int32_t)leftVol - (int32_t)mLeftVolShort) << 16; 2768 int32_t vlInc = d / (int32_t)frameCount; 2769 d = ((int32_t)rightVol - (int32_t)mRightVolShort) << 16; 2770 int32_t vrInc = d / (int32_t)frameCount; 2771 int32_t vl = ((int32_t)mLeftVolShort << 16); 2772 int32_t vr = ((int32_t)mRightVolShort << 16); 2773 do { 2774 out[0] = clamp16(mul(out[0], vl >> 16) >> 12); 2775 out[1] = clamp16(mul(out[1], vr >> 16) >> 12); 2776 out += 2; 2777 vl += vlInc; 2778 vr += vrInc; 2779 } while (--frameCount); 2780 } 2781 } else { 2782 if (mChannelCount == 1) { 2783 do { 2784 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2785 out++; 2786 } while (--frameCount); 2787 } else { 2788 do { 2789 out[0] = clamp16(mul(out[0], leftVol) >> 12); 2790 out[1] = clamp16(mul(out[1], rightVol) >> 12); 2791 out += 2; 2792 } while (--frameCount); 2793 } 2794 } 2795 2796 // convert back to unsigned 8 bit after volume calculation 2797 if (mFormat == AUDIO_FORMAT_PCM_8_BIT) { 2798 size_t count = mFrameCount * mChannelCount; 2799 int16_t *src = mMixBuffer; 2800 uint8_t *dst = (uint8_t *)mMixBuffer; 2801 while(count--) { 2802 *dst++ = (uint8_t)(((int32_t)*src++ + (1<<7)) >> 8)^0x80; 2803 } 2804 } 2805 2806 mLeftVolShort = leftVol; 2807 mRightVolShort = rightVol; 2808} 2809 2810AudioFlinger::PlaybackThread::mixer_state AudioFlinger::DirectOutputThread::prepareTracks_l( 2811 Vector< sp<Track> > *tracksToRemove 2812) 2813{ 2814 sp<Track> trackToRemove; 2815 2816 // FIXME Temporarily renamed to avoid confusion with the member "mixerStatus" 2817 mixer_state mixerStatus_ = MIXER_IDLE; 2818 2819 // find out which tracks need to be processed 2820 if (mActiveTracks.size() != 0) { 2821 sp<Track> t = mActiveTracks[0].promote(); 2822 // see FIXME in AudioFlinger.h, return MIXER_IDLE might also work 2823 if (t == 0) return MIXER_CONTINUE; 2824 //if (t == 0) continue; 2825 2826 Track* const track = t.get(); 2827 audio_track_cblk_t* cblk = track->cblk(); 2828 2829 // The first time a track is added we wait 2830 // for all its buffers to be filled before processing it 2831 if (cblk->framesReady() && track->isReady() && 2832 !track->isPaused() && !track->isTerminated()) 2833 { 2834 //ALOGV("track %d u=%08x, s=%08x [OK]", track->name(), cblk->user, cblk->server); 2835 2836 if (track->mFillingUpStatus == Track::FS_FILLED) { 2837 track->mFillingUpStatus = Track::FS_ACTIVE; 2838 mLeftVolFloat = mRightVolFloat = 0; 2839 mLeftVolShort = mRightVolShort = 0; 2840 if (track->mState == TrackBase::RESUMING) { 2841 track->mState = TrackBase::ACTIVE; 2842 rampVolume = true; 2843 } 2844 } else if (cblk->server != 0) { 2845 // If the track is stopped before the first frame was mixed, 2846 // do not apply ramp 2847 rampVolume = true; 2848 } 2849 // compute volume for this track 2850 float left, right; 2851 if (track->isMuted() || mMasterMute || track->isPausing() || 2852 mStreamTypes[track->streamType()].mute) { 2853 left = right = 0; 2854 if (track->isPausing()) { 2855 track->setPaused(); 2856 } 2857 } else { 2858 float typeVolume = mStreamTypes[track->streamType()].volume; 2859 float v = mMasterVolume * typeVolume; 2860 uint32_t vlr = cblk->getVolumeLR(); 2861 float v_clamped = v * (vlr & 0xFFFF); 2862 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2863 left = v_clamped/MAX_GAIN; 2864 v_clamped = v * (vlr >> 16); 2865 if (v_clamped > MAX_GAIN) v_clamped = MAX_GAIN; 2866 right = v_clamped/MAX_GAIN; 2867 } 2868 2869 if (left != mLeftVolFloat || right != mRightVolFloat) { 2870 mLeftVolFloat = left; 2871 mRightVolFloat = right; 2872 2873 // If audio HAL implements volume control, 2874 // force software volume to nominal value 2875 if (mOutput->stream->set_volume(mOutput->stream, left, right) == NO_ERROR) { 2876 left = 1.0f; 2877 right = 1.0f; 2878 } 2879 2880 // Convert volumes from float to 8.24 2881 uint32_t vl = (uint32_t)(left * (1 << 24)); 2882 uint32_t vr = (uint32_t)(right * (1 << 24)); 2883 2884 // Delegate volume control to effect in track effect chain if needed 2885 // only one effect chain can be present on DirectOutputThread, so if 2886 // there is one, the track is connected to it 2887 if (!mEffectChains.isEmpty()) { 2888 // Do not ramp volume if volume is controlled by effect 2889 if (mEffectChains[0]->setVolume_l(&vl, &vr)) { 2890 rampVolume = false; 2891 } 2892 } 2893 2894 // Convert volumes from 8.24 to 4.12 format 2895 uint32_t v_clamped = (vl + (1 << 11)) >> 12; 2896 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2897 leftVol = (uint16_t)v_clamped; 2898 v_clamped = (vr + (1 << 11)) >> 12; 2899 if (v_clamped > MAX_GAIN_INT) v_clamped = MAX_GAIN_INT; 2900 rightVol = (uint16_t)v_clamped; 2901 } else { 2902 leftVol = mLeftVolShort; 2903 rightVol = mRightVolShort; 2904 rampVolume = false; 2905 } 2906 2907 // reset retry count 2908 track->mRetryCount = kMaxTrackRetriesDirect; 2909 mActiveTrack = t; 2910 mixerStatus_ = MIXER_TRACKS_READY; 2911 } else { 2912 //ALOGV("track %d u=%08x, s=%08x [NOT READY]", track->name(), cblk->user, cblk->server); 2913 if (track->isStopped()) { 2914 track->reset(); 2915 } 2916 if (track->isTerminated() || track->isStopped() || track->isPaused()) { 2917 // We have consumed all the buffers of this track. 2918 // Remove it from the list of active tracks. 2919 trackToRemove = track; 2920 } else { 2921 // No buffers for this track. Give it a few chances to 2922 // fill a buffer, then remove it from active list. 2923 if (--(track->mRetryCount) <= 0) { 2924 ALOGV("BUFFER TIMEOUT: remove(%d) from active list", track->name()); 2925 trackToRemove = track; 2926 } else { 2927 mixerStatus_ = MIXER_TRACKS_ENABLED; 2928 } 2929 } 2930 } 2931 } 2932 2933 // FIXME merge this with similar code for removing multiple tracks 2934 // remove all the tracks that need to be... 2935 if (CC_UNLIKELY(trackToRemove != 0)) { 2936 tracksToRemove->add(trackToRemove); 2937 mActiveTracks.remove(trackToRemove); 2938 if (!mEffectChains.isEmpty()) { 2939 ALOGV("stopping track on chain %p for session Id: %d", effectChains[0].get(), 2940 trackToRemove->sessionId()); 2941 mEffectChains[0]->decActiveTrackCnt(); 2942 } 2943 if (trackToRemove->isTerminated()) { 2944 removeTrack_l(trackToRemove); 2945 } 2946 } 2947 2948 return mixerStatus_; 2949} 2950 2951void AudioFlinger::DirectOutputThread::threadLoop_mix() 2952{ 2953 AudioBufferProvider::Buffer buffer; 2954 size_t frameCount = mFrameCount; 2955 int8_t *curBuf = (int8_t *)mMixBuffer; 2956 // output audio to hardware 2957 while (frameCount) { 2958 buffer.frameCount = frameCount; 2959 mActiveTrack->getNextBuffer(&buffer); 2960 if (CC_UNLIKELY(buffer.raw == NULL)) { 2961 memset(curBuf, 0, frameCount * mFrameSize); 2962 break; 2963 } 2964 memcpy(curBuf, buffer.raw, buffer.frameCount * mFrameSize); 2965 frameCount -= buffer.frameCount; 2966 curBuf += buffer.frameCount * mFrameSize; 2967 mActiveTrack->releaseBuffer(&buffer); 2968 } 2969 sleepTime = 0; 2970 standbyTime = systemTime() + standbyDelay; 2971 mActiveTrack.clear(); 2972 applyVolume(); 2973} 2974 2975void AudioFlinger::DirectOutputThread::threadLoop_sleepTime() 2976{ 2977 if (sleepTime == 0) { 2978 if (mixerStatus == MIXER_TRACKS_ENABLED) { 2979 sleepTime = activeSleepTime; 2980 } else { 2981 sleepTime = idleSleepTime; 2982 } 2983 } else if (mBytesWritten != 0 && audio_is_linear_pcm(mFormat)) { 2984 memset (mMixBuffer, 0, mFrameCount * mFrameSize); 2985 sleepTime = 0; 2986 } 2987} 2988 2989// getTrackName_l() must be called with ThreadBase::mLock held 2990int AudioFlinger::DirectOutputThread::getTrackName_l() 2991{ 2992 return 0; 2993} 2994 2995// deleteTrackName_l() must be called with ThreadBase::mLock held 2996void AudioFlinger::DirectOutputThread::deleteTrackName_l(int name) 2997{ 2998} 2999 3000// checkForNewParameters_l() must be called with ThreadBase::mLock held 3001bool AudioFlinger::DirectOutputThread::checkForNewParameters_l() 3002{ 3003 bool reconfig = false; 3004 3005 while (!mNewParameters.isEmpty()) { 3006 status_t status = NO_ERROR; 3007 String8 keyValuePair = mNewParameters[0]; 3008 AudioParameter param = AudioParameter(keyValuePair); 3009 int value; 3010 3011 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 3012 // do not accept frame count changes if tracks are open as the track buffer 3013 // size depends on frame count and correct behavior would not be garantied 3014 // if frame count is changed after track creation 3015 if (!mTracks.isEmpty()) { 3016 status = INVALID_OPERATION; 3017 } else { 3018 reconfig = true; 3019 } 3020 } 3021 if (status == NO_ERROR) { 3022 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3023 keyValuePair.string()); 3024 if (!mStandby && status == INVALID_OPERATION) { 3025 mOutput->stream->common.standby(&mOutput->stream->common); 3026 mStandby = true; 3027 mBytesWritten = 0; 3028 status = mOutput->stream->common.set_parameters(&mOutput->stream->common, 3029 keyValuePair.string()); 3030 } 3031 if (status == NO_ERROR && reconfig) { 3032 readOutputParameters(); 3033 sendConfigEvent_l(AudioSystem::OUTPUT_CONFIG_CHANGED); 3034 } 3035 } 3036 3037 mNewParameters.removeAt(0); 3038 3039 mParamStatus = status; 3040 mParamCond.signal(); 3041 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 3042 // already timed out waiting for the status and will never signal the condition. 3043 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 3044 } 3045 return reconfig; 3046} 3047 3048uint32_t AudioFlinger::DirectOutputThread::activeSleepTimeUs() 3049{ 3050 uint32_t time; 3051 if (audio_is_linear_pcm(mFormat)) { 3052 time = PlaybackThread::activeSleepTimeUs(); 3053 } else { 3054 time = 10000; 3055 } 3056 return time; 3057} 3058 3059uint32_t AudioFlinger::DirectOutputThread::idleSleepTimeUs() 3060{ 3061 uint32_t time; 3062 if (audio_is_linear_pcm(mFormat)) { 3063 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000) / 2; 3064 } else { 3065 time = 10000; 3066 } 3067 return time; 3068} 3069 3070uint32_t AudioFlinger::DirectOutputThread::suspendSleepTimeUs() 3071{ 3072 uint32_t time; 3073 if (audio_is_linear_pcm(mFormat)) { 3074 time = (uint32_t)(((mFrameCount * 1000) / mSampleRate) * 1000); 3075 } else { 3076 time = 10000; 3077 } 3078 return time; 3079} 3080 3081 3082// ---------------------------------------------------------------------------- 3083 3084AudioFlinger::DuplicatingThread::DuplicatingThread(const sp<AudioFlinger>& audioFlinger, 3085 AudioFlinger::MixerThread* mainThread, audio_io_handle_t id) 3086 : MixerThread(audioFlinger, mainThread->getOutput(), id, mainThread->device(), DUPLICATING), 3087 mWaitTimeMs(UINT_MAX) 3088{ 3089 addOutputTrack(mainThread); 3090} 3091 3092AudioFlinger::DuplicatingThread::~DuplicatingThread() 3093{ 3094 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3095 mOutputTracks[i]->destroy(); 3096 } 3097} 3098 3099void AudioFlinger::DuplicatingThread::threadLoop_mix() 3100{ 3101 // mix buffers... 3102 if (outputsReady(outputTracks)) { 3103 mAudioMixer->process(AudioBufferProvider::kInvalidPTS); 3104 } else { 3105 memset(mMixBuffer, 0, mixBufferSize); 3106 } 3107 sleepTime = 0; 3108 writeFrames = mFrameCount; 3109} 3110 3111void AudioFlinger::DuplicatingThread::threadLoop_sleepTime() 3112{ 3113 if (sleepTime == 0) { 3114 if (mixerStatus == MIXER_TRACKS_ENABLED) { 3115 sleepTime = activeSleepTime; 3116 } else { 3117 sleepTime = idleSleepTime; 3118 } 3119 } else if (mBytesWritten != 0) { 3120 // flush remaining overflow buffers in output tracks 3121 for (size_t i = 0; i < outputTracks.size(); i++) { 3122 if (outputTracks[i]->isActive()) { 3123 sleepTime = 0; 3124 writeFrames = 0; 3125 memset(mMixBuffer, 0, mixBufferSize); 3126 break; 3127 } 3128 } 3129 } 3130} 3131 3132void AudioFlinger::DuplicatingThread::threadLoop_write() 3133{ 3134 standbyTime = systemTime() + mStandbyTimeInNsecs; 3135 for (size_t i = 0; i < outputTracks.size(); i++) { 3136 outputTracks[i]->write(mMixBuffer, writeFrames); 3137 } 3138 mBytesWritten += mixBufferSize; 3139} 3140 3141void AudioFlinger::DuplicatingThread::threadLoop_standby() 3142{ 3143 // DuplicatingThread implements standby by stopping all tracks 3144 for (size_t i = 0; i < outputTracks.size(); i++) { 3145 outputTracks[i]->stop(); 3146 } 3147} 3148 3149void AudioFlinger::DuplicatingThread::saveOutputTracks() 3150{ 3151 outputTracks = mOutputTracks; 3152} 3153 3154void AudioFlinger::DuplicatingThread::clearOutputTracks() 3155{ 3156 outputTracks.clear(); 3157} 3158 3159void AudioFlinger::DuplicatingThread::addOutputTrack(MixerThread *thread) 3160{ 3161 Mutex::Autolock _l(mLock); 3162 // FIXME explain this formula 3163 int frameCount = (3 * mFrameCount * mSampleRate) / thread->sampleRate(); 3164 OutputTrack *outputTrack = new OutputTrack(thread, 3165 this, 3166 mSampleRate, 3167 mFormat, 3168 mChannelMask, 3169 frameCount); 3170 if (outputTrack->cblk() != NULL) { 3171 thread->setStreamVolume(AUDIO_STREAM_CNT, 1.0f); 3172 mOutputTracks.add(outputTrack); 3173 ALOGV("addOutputTrack() track %p, on thread %p", outputTrack, thread); 3174 updateWaitTime_l(); 3175 } 3176} 3177 3178void AudioFlinger::DuplicatingThread::removeOutputTrack(MixerThread *thread) 3179{ 3180 Mutex::Autolock _l(mLock); 3181 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3182 if (mOutputTracks[i]->thread() == thread) { 3183 mOutputTracks[i]->destroy(); 3184 mOutputTracks.removeAt(i); 3185 updateWaitTime_l(); 3186 return; 3187 } 3188 } 3189 ALOGV("removeOutputTrack(): unkonwn thread: %p", thread); 3190} 3191 3192// caller must hold mLock 3193void AudioFlinger::DuplicatingThread::updateWaitTime_l() 3194{ 3195 mWaitTimeMs = UINT_MAX; 3196 for (size_t i = 0; i < mOutputTracks.size(); i++) { 3197 sp<ThreadBase> strong = mOutputTracks[i]->thread().promote(); 3198 if (strong != 0) { 3199 uint32_t waitTimeMs = (strong->frameCount() * 2 * 1000) / strong->sampleRate(); 3200 if (waitTimeMs < mWaitTimeMs) { 3201 mWaitTimeMs = waitTimeMs; 3202 } 3203 } 3204 } 3205} 3206 3207 3208bool AudioFlinger::DuplicatingThread::outputsReady(const SortedVector< sp<OutputTrack> > &outputTracks) 3209{ 3210 for (size_t i = 0; i < outputTracks.size(); i++) { 3211 sp <ThreadBase> thread = outputTracks[i]->thread().promote(); 3212 if (thread == 0) { 3213 ALOGW("DuplicatingThread::outputsReady() could not promote thread on output track %p", outputTracks[i].get()); 3214 return false; 3215 } 3216 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3217 if (playbackThread->standby() && !playbackThread->isSuspended()) { 3218 ALOGV("DuplicatingThread output track %p on thread %p Not Ready", outputTracks[i].get(), thread.get()); 3219 return false; 3220 } 3221 } 3222 return true; 3223} 3224 3225uint32_t AudioFlinger::DuplicatingThread::activeSleepTimeUs() 3226{ 3227 return (mWaitTimeMs * 1000) / 2; 3228} 3229 3230// ---------------------------------------------------------------------------- 3231 3232// TrackBase constructor must be called with AudioFlinger::mLock held 3233AudioFlinger::ThreadBase::TrackBase::TrackBase( 3234 ThreadBase *thread, 3235 const sp<Client>& client, 3236 uint32_t sampleRate, 3237 audio_format_t format, 3238 uint32_t channelMask, 3239 int frameCount, 3240 const sp<IMemory>& sharedBuffer, 3241 int sessionId) 3242 : RefBase(), 3243 mThread(thread), 3244 mClient(client), 3245 mCblk(NULL), 3246 // mBuffer 3247 // mBufferEnd 3248 mFrameCount(0), 3249 mState(IDLE), 3250 mFormat(format), 3251 mStepServerFailed(false), 3252 mSessionId(sessionId) 3253 // mChannelCount 3254 // mChannelMask 3255{ 3256 ALOGV_IF(sharedBuffer != 0, "sharedBuffer: %p, size: %d", sharedBuffer->pointer(), sharedBuffer->size()); 3257 3258 // ALOGD("Creating track with %d buffers @ %d bytes", bufferCount, bufferSize); 3259 size_t size = sizeof(audio_track_cblk_t); 3260 uint8_t channelCount = popcount(channelMask); 3261 size_t bufferSize = frameCount*channelCount*sizeof(int16_t); 3262 if (sharedBuffer == 0) { 3263 size += bufferSize; 3264 } 3265 3266 if (client != NULL) { 3267 mCblkMemory = client->heap()->allocate(size); 3268 if (mCblkMemory != 0) { 3269 mCblk = static_cast<audio_track_cblk_t *>(mCblkMemory->pointer()); 3270 if (mCblk != NULL) { // construct the shared structure in-place. 3271 new(mCblk) audio_track_cblk_t(); 3272 // clear all buffers 3273 mCblk->frameCount = frameCount; 3274 mCblk->sampleRate = sampleRate; 3275 mChannelCount = channelCount; 3276 mChannelMask = channelMask; 3277 if (sharedBuffer == 0) { 3278 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3279 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3280 // Force underrun condition to avoid false underrun callback until first data is 3281 // written to buffer (other flags are cleared) 3282 mCblk->flags = CBLK_UNDERRUN_ON; 3283 } else { 3284 mBuffer = sharedBuffer->pointer(); 3285 } 3286 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3287 } 3288 } else { 3289 ALOGE("not enough memory for AudioTrack size=%u", size); 3290 client->heap()->dump("AudioTrack"); 3291 return; 3292 } 3293 } else { 3294 mCblk = (audio_track_cblk_t *)(new uint8_t[size]); 3295 // construct the shared structure in-place. 3296 new(mCblk) audio_track_cblk_t(); 3297 // clear all buffers 3298 mCblk->frameCount = frameCount; 3299 mCblk->sampleRate = sampleRate; 3300 mChannelCount = channelCount; 3301 mChannelMask = channelMask; 3302 mBuffer = (char*)mCblk + sizeof(audio_track_cblk_t); 3303 memset(mBuffer, 0, frameCount*channelCount*sizeof(int16_t)); 3304 // Force underrun condition to avoid false underrun callback until first data is 3305 // written to buffer (other flags are cleared) 3306 mCblk->flags = CBLK_UNDERRUN_ON; 3307 mBufferEnd = (uint8_t *)mBuffer + bufferSize; 3308 } 3309} 3310 3311AudioFlinger::ThreadBase::TrackBase::~TrackBase() 3312{ 3313 if (mCblk != NULL) { 3314 if (mClient == 0) { 3315 delete mCblk; 3316 } else { 3317 mCblk->~audio_track_cblk_t(); // destroy our shared-structure. 3318 } 3319 } 3320 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 3321 if (mClient != 0) { 3322 // Client destructor must run with AudioFlinger mutex locked 3323 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 3324 // If the client's reference count drops to zero, the associated destructor 3325 // must run with AudioFlinger lock held. Thus the explicit clear() rather than 3326 // relying on the automatic clear() at end of scope. 3327 mClient.clear(); 3328 } 3329} 3330 3331// AudioBufferProvider interface 3332// getNextBuffer() = 0; 3333// This implementation of releaseBuffer() is used by Track and RecordTrack, but not TimedTrack 3334void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer) 3335{ 3336 buffer->raw = NULL; 3337 mFrameCount = buffer->frameCount; 3338 (void) step(); // ignore return value of step() 3339 buffer->frameCount = 0; 3340} 3341 3342bool AudioFlinger::ThreadBase::TrackBase::step() { 3343 bool result; 3344 audio_track_cblk_t* cblk = this->cblk(); 3345 3346 result = cblk->stepServer(mFrameCount); 3347 if (!result) { 3348 ALOGV("stepServer failed acquiring cblk mutex"); 3349 mStepServerFailed = true; 3350 } 3351 return result; 3352} 3353 3354void AudioFlinger::ThreadBase::TrackBase::reset() { 3355 audio_track_cblk_t* cblk = this->cblk(); 3356 3357 cblk->user = 0; 3358 cblk->server = 0; 3359 cblk->userBase = 0; 3360 cblk->serverBase = 0; 3361 mStepServerFailed = false; 3362 ALOGV("TrackBase::reset"); 3363} 3364 3365int AudioFlinger::ThreadBase::TrackBase::sampleRate() const { 3366 return (int)mCblk->sampleRate; 3367} 3368 3369void* AudioFlinger::ThreadBase::TrackBase::getBuffer(uint32_t offset, uint32_t frames) const { 3370 audio_track_cblk_t* cblk = this->cblk(); 3371 size_t frameSize = cblk->frameSize; 3372 int8_t *bufferStart = (int8_t *)mBuffer + (offset-cblk->serverBase)*frameSize; 3373 int8_t *bufferEnd = bufferStart + frames * frameSize; 3374 3375 // Check validity of returned pointer in case the track control block would have been corrupted. 3376 if (bufferStart < mBuffer || bufferStart > bufferEnd || bufferEnd > mBufferEnd || 3377 ((unsigned long)bufferStart & (unsigned long)(frameSize - 1))) { 3378 ALOGE("TrackBase::getBuffer buffer out of range:\n start: %p, end %p , mBuffer %p mBufferEnd %p\n \ 3379 server %d, serverBase %d, user %d, userBase %d", 3380 bufferStart, bufferEnd, mBuffer, mBufferEnd, 3381 cblk->server, cblk->serverBase, cblk->user, cblk->userBase); 3382 return NULL; 3383 } 3384 3385 return bufferStart; 3386} 3387 3388// ---------------------------------------------------------------------------- 3389 3390// Track constructor must be called with AudioFlinger::mLock and ThreadBase::mLock held 3391AudioFlinger::PlaybackThread::Track::Track( 3392 PlaybackThread *thread, 3393 const sp<Client>& client, 3394 audio_stream_type_t streamType, 3395 uint32_t sampleRate, 3396 audio_format_t format, 3397 uint32_t channelMask, 3398 int frameCount, 3399 const sp<IMemory>& sharedBuffer, 3400 int sessionId) 3401 : TrackBase(thread, client, sampleRate, format, channelMask, frameCount, sharedBuffer, sessionId), 3402 mMute(false), mSharedBuffer(sharedBuffer), mName(-1), mMainBuffer(NULL), mAuxBuffer(NULL), 3403 mAuxEffectId(0), mHasVolumeController(false) 3404{ 3405 if (mCblk != NULL) { 3406 if (thread != NULL) { 3407 mName = thread->getTrackName_l(); 3408 mMainBuffer = thread->mixBuffer(); 3409 } 3410 ALOGV("Track constructor name %d, calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3411 if (mName < 0) { 3412 ALOGE("no more track names available"); 3413 } 3414 mStreamType = streamType; 3415 // NOTE: audio_track_cblk_t::frameSize for 8 bit PCM data is based on a sample size of 3416 // 16 bit because data is converted to 16 bit before being stored in buffer by AudioTrack 3417 mCblk->frameSize = audio_is_linear_pcm(format) ? mChannelCount * sizeof(int16_t) : sizeof(uint8_t); 3418 } 3419} 3420 3421AudioFlinger::PlaybackThread::Track::~Track() 3422{ 3423 ALOGV("PlaybackThread::Track destructor"); 3424 sp<ThreadBase> thread = mThread.promote(); 3425 if (thread != 0) { 3426 Mutex::Autolock _l(thread->mLock); 3427 mState = TERMINATED; 3428 } 3429} 3430 3431void AudioFlinger::PlaybackThread::Track::destroy() 3432{ 3433 // NOTE: destroyTrack_l() can remove a strong reference to this Track 3434 // by removing it from mTracks vector, so there is a risk that this Tracks's 3435 // destructor is called. As the destructor needs to lock mLock, 3436 // we must acquire a strong reference on this Track before locking mLock 3437 // here so that the destructor is called only when exiting this function. 3438 // On the other hand, as long as Track::destroy() is only called by 3439 // TrackHandle destructor, the TrackHandle still holds a strong ref on 3440 // this Track with its member mTrack. 3441 sp<Track> keep(this); 3442 { // scope for mLock 3443 sp<ThreadBase> thread = mThread.promote(); 3444 if (thread != 0) { 3445 if (!isOutputTrack()) { 3446 if (mState == ACTIVE || mState == RESUMING) { 3447 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3448 3449 // to track the speaker usage 3450 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3451 } 3452 AudioSystem::releaseOutput(thread->id()); 3453 } 3454 Mutex::Autolock _l(thread->mLock); 3455 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3456 playbackThread->destroyTrack_l(this); 3457 } 3458 } 3459} 3460 3461void AudioFlinger::PlaybackThread::Track::dump(char* buffer, size_t size) 3462{ 3463 uint32_t vlr = mCblk->getVolumeLR(); 3464 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", 3465 mName - AudioMixer::TRACK0, 3466 (mClient == 0) ? getpid_cached : mClient->pid(), 3467 mStreamType, 3468 mFormat, 3469 mChannelMask, 3470 mSessionId, 3471 mFrameCount, 3472 mState, 3473 mMute, 3474 mFillingUpStatus, 3475 mCblk->sampleRate, 3476 vlr & 0xFFFF, 3477 vlr >> 16, 3478 mCblk->server, 3479 mCblk->user, 3480 (int)mMainBuffer, 3481 (int)mAuxBuffer); 3482} 3483 3484// AudioBufferProvider interface 3485status_t AudioFlinger::PlaybackThread::Track::getNextBuffer( 3486 AudioBufferProvider::Buffer* buffer, int64_t pts) 3487{ 3488 audio_track_cblk_t* cblk = this->cblk(); 3489 uint32_t framesReady; 3490 uint32_t framesReq = buffer->frameCount; 3491 3492 // Check if last stepServer failed, try to step now 3493 if (mStepServerFailed) { 3494 if (!step()) goto getNextBuffer_exit; 3495 ALOGV("stepServer recovered"); 3496 mStepServerFailed = false; 3497 } 3498 3499 framesReady = cblk->framesReady(); 3500 3501 if (CC_LIKELY(framesReady)) { 3502 uint32_t s = cblk->server; 3503 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 3504 3505 bufferEnd = (cblk->loopEnd < bufferEnd) ? cblk->loopEnd : bufferEnd; 3506 if (framesReq > framesReady) { 3507 framesReq = framesReady; 3508 } 3509 if (s + framesReq > bufferEnd) { 3510 framesReq = bufferEnd - s; 3511 } 3512 3513 buffer->raw = getBuffer(s, framesReq); 3514 if (buffer->raw == NULL) goto getNextBuffer_exit; 3515 3516 buffer->frameCount = framesReq; 3517 return NO_ERROR; 3518 } 3519 3520getNextBuffer_exit: 3521 buffer->raw = NULL; 3522 buffer->frameCount = 0; 3523 ALOGV("getNextBuffer() no more data for track %d on thread %p", mName, mThread.unsafe_get()); 3524 return NOT_ENOUGH_DATA; 3525} 3526 3527uint32_t AudioFlinger::PlaybackThread::Track::framesReady() const{ 3528 return mCblk->framesReady(); 3529} 3530 3531bool AudioFlinger::PlaybackThread::Track::isReady() const { 3532 if (mFillingUpStatus != FS_FILLING || isStopped() || isPausing()) return true; 3533 3534 if (framesReady() >= mCblk->frameCount || 3535 (mCblk->flags & CBLK_FORCEREADY_MSK)) { 3536 mFillingUpStatus = FS_FILLED; 3537 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3538 return true; 3539 } 3540 return false; 3541} 3542 3543status_t AudioFlinger::PlaybackThread::Track::start(pid_t tid) 3544{ 3545 status_t status = NO_ERROR; 3546 ALOGV("start(%d), calling pid %d session %d tid %d", 3547 mName, IPCThreadState::self()->getCallingPid(), mSessionId, tid); 3548 sp<ThreadBase> thread = mThread.promote(); 3549 if (thread != 0) { 3550 Mutex::Autolock _l(thread->mLock); 3551 track_state state = mState; 3552 // here the track could be either new, or restarted 3553 // in both cases "unstop" the track 3554 if (mState == PAUSED) { 3555 mState = TrackBase::RESUMING; 3556 ALOGV("PAUSED => RESUMING (%d) on thread %p", mName, this); 3557 } else { 3558 mState = TrackBase::ACTIVE; 3559 ALOGV("? => ACTIVE (%d) on thread %p", mName, this); 3560 } 3561 3562 if (!isOutputTrack() && state != ACTIVE && state != RESUMING) { 3563 thread->mLock.unlock(); 3564 status = AudioSystem::startOutput(thread->id(), mStreamType, mSessionId); 3565 thread->mLock.lock(); 3566 3567 // to track the speaker usage 3568 if (status == NO_ERROR) { 3569 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStart); 3570 } 3571 } 3572 if (status == NO_ERROR) { 3573 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3574 playbackThread->addTrack_l(this); 3575 } else { 3576 mState = state; 3577 } 3578 } else { 3579 status = BAD_VALUE; 3580 } 3581 return status; 3582} 3583 3584void AudioFlinger::PlaybackThread::Track::stop() 3585{ 3586 ALOGV("stop(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3587 sp<ThreadBase> thread = mThread.promote(); 3588 if (thread != 0) { 3589 Mutex::Autolock _l(thread->mLock); 3590 track_state state = mState; 3591 if (mState > STOPPED) { 3592 mState = STOPPED; 3593 // If the track is not active (PAUSED and buffers full), flush buffers 3594 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3595 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3596 reset(); 3597 } 3598 ALOGV("(> STOPPED) => STOPPED (%d) on thread %p", mName, playbackThread); 3599 } 3600 if (!isOutputTrack() && (state == ACTIVE || state == RESUMING)) { 3601 thread->mLock.unlock(); 3602 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3603 thread->mLock.lock(); 3604 3605 // to track the speaker usage 3606 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3607 } 3608 } 3609} 3610 3611void AudioFlinger::PlaybackThread::Track::pause() 3612{ 3613 ALOGV("pause(%d), calling pid %d", mName, IPCThreadState::self()->getCallingPid()); 3614 sp<ThreadBase> thread = mThread.promote(); 3615 if (thread != 0) { 3616 Mutex::Autolock _l(thread->mLock); 3617 if (mState == ACTIVE || mState == RESUMING) { 3618 mState = PAUSING; 3619 ALOGV("ACTIVE/RESUMING => PAUSING (%d) on thread %p", mName, thread.get()); 3620 if (!isOutputTrack()) { 3621 thread->mLock.unlock(); 3622 AudioSystem::stopOutput(thread->id(), mStreamType, mSessionId); 3623 thread->mLock.lock(); 3624 3625 // to track the speaker usage 3626 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop); 3627 } 3628 } 3629 } 3630} 3631 3632void AudioFlinger::PlaybackThread::Track::flush() 3633{ 3634 ALOGV("flush(%d)", mName); 3635 sp<ThreadBase> thread = mThread.promote(); 3636 if (thread != 0) { 3637 Mutex::Autolock _l(thread->mLock); 3638 if (mState != STOPPED && mState != PAUSED && mState != PAUSING) { 3639 return; 3640 } 3641 // No point remaining in PAUSED state after a flush => go to 3642 // STOPPED state 3643 mState = STOPPED; 3644 3645 // do not reset the track if it is still in the process of being stopped or paused. 3646 // this will be done by prepareTracks_l() when the track is stopped. 3647 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3648 if (playbackThread->mActiveTracks.indexOf(this) < 0) { 3649 reset(); 3650 } 3651 } 3652} 3653 3654void AudioFlinger::PlaybackThread::Track::reset() 3655{ 3656 // Do not reset twice to avoid discarding data written just after a flush and before 3657 // the audioflinger thread detects the track is stopped. 3658 if (!mResetDone) { 3659 TrackBase::reset(); 3660 // Force underrun condition to avoid false underrun callback until first data is 3661 // written to buffer 3662 android_atomic_and(~CBLK_FORCEREADY_MSK, &mCblk->flags); 3663 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 3664 mFillingUpStatus = FS_FILLING; 3665 mResetDone = true; 3666 } 3667} 3668 3669void AudioFlinger::PlaybackThread::Track::mute(bool muted) 3670{ 3671 mMute = muted; 3672} 3673 3674status_t AudioFlinger::PlaybackThread::Track::attachAuxEffect(int EffectId) 3675{ 3676 status_t status = DEAD_OBJECT; 3677 sp<ThreadBase> thread = mThread.promote(); 3678 if (thread != 0) { 3679 PlaybackThread *playbackThread = (PlaybackThread *)thread.get(); 3680 status = playbackThread->attachAuxEffect(this, EffectId); 3681 } 3682 return status; 3683} 3684 3685void AudioFlinger::PlaybackThread::Track::setAuxBuffer(int EffectId, int32_t *buffer) 3686{ 3687 mAuxEffectId = EffectId; 3688 mAuxBuffer = buffer; 3689} 3690 3691// timed audio tracks 3692 3693sp<AudioFlinger::PlaybackThread::TimedTrack> 3694AudioFlinger::PlaybackThread::TimedTrack::create( 3695 PlaybackThread *thread, 3696 const sp<Client>& client, 3697 audio_stream_type_t streamType, 3698 uint32_t sampleRate, 3699 audio_format_t format, 3700 uint32_t channelMask, 3701 int frameCount, 3702 const sp<IMemory>& sharedBuffer, 3703 int sessionId) { 3704 if (!client->reserveTimedTrack()) 3705 return NULL; 3706 3707 sp<TimedTrack> track = new TimedTrack( 3708 thread, client, streamType, sampleRate, format, channelMask, frameCount, 3709 sharedBuffer, sessionId); 3710 3711 if (track == NULL) { 3712 client->releaseTimedTrack(); 3713 return NULL; 3714 } 3715 3716 return track; 3717} 3718 3719AudioFlinger::PlaybackThread::TimedTrack::TimedTrack( 3720 PlaybackThread *thread, 3721 const sp<Client>& client, 3722 audio_stream_type_t streamType, 3723 uint32_t sampleRate, 3724 audio_format_t format, 3725 uint32_t channelMask, 3726 int frameCount, 3727 const sp<IMemory>& sharedBuffer, 3728 int sessionId) 3729 : Track(thread, client, streamType, sampleRate, format, channelMask, 3730 frameCount, sharedBuffer, sessionId), 3731 mTimedSilenceBuffer(NULL), 3732 mTimedSilenceBufferSize(0), 3733 mTimedAudioOutputOnTime(false), 3734 mMediaTimeTransformValid(false) 3735{ 3736 LocalClock lc; 3737 mLocalTimeFreq = lc.getLocalFreq(); 3738 3739 mLocalTimeToSampleTransform.a_zero = 0; 3740 mLocalTimeToSampleTransform.b_zero = 0; 3741 mLocalTimeToSampleTransform.a_to_b_numer = sampleRate; 3742 mLocalTimeToSampleTransform.a_to_b_denom = mLocalTimeFreq; 3743 LinearTransform::reduce(&mLocalTimeToSampleTransform.a_to_b_numer, 3744 &mLocalTimeToSampleTransform.a_to_b_denom); 3745} 3746 3747AudioFlinger::PlaybackThread::TimedTrack::~TimedTrack() { 3748 mClient->releaseTimedTrack(); 3749 delete [] mTimedSilenceBuffer; 3750} 3751 3752status_t AudioFlinger::PlaybackThread::TimedTrack::allocateTimedBuffer( 3753 size_t size, sp<IMemory>* buffer) { 3754 3755 Mutex::Autolock _l(mTimedBufferQueueLock); 3756 3757 trimTimedBufferQueue_l(); 3758 3759 // lazily initialize the shared memory heap for timed buffers 3760 if (mTimedMemoryDealer == NULL) { 3761 const int kTimedBufferHeapSize = 512 << 10; 3762 3763 mTimedMemoryDealer = new MemoryDealer(kTimedBufferHeapSize, 3764 "AudioFlingerTimed"); 3765 if (mTimedMemoryDealer == NULL) 3766 return NO_MEMORY; 3767 } 3768 3769 sp<IMemory> newBuffer = mTimedMemoryDealer->allocate(size); 3770 if (newBuffer == NULL) { 3771 newBuffer = mTimedMemoryDealer->allocate(size); 3772 if (newBuffer == NULL) 3773 return NO_MEMORY; 3774 } 3775 3776 *buffer = newBuffer; 3777 return NO_ERROR; 3778} 3779 3780// caller must hold mTimedBufferQueueLock 3781void AudioFlinger::PlaybackThread::TimedTrack::trimTimedBufferQueue_l() { 3782 int64_t mediaTimeNow; 3783 { 3784 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3785 if (!mMediaTimeTransformValid) 3786 return; 3787 3788 int64_t targetTimeNow; 3789 status_t res = (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) 3790 ? mCCHelper.getCommonTime(&targetTimeNow) 3791 : mCCHelper.getLocalTime(&targetTimeNow); 3792 3793 if (OK != res) 3794 return; 3795 3796 if (!mMediaTimeTransform.doReverseTransform(targetTimeNow, 3797 &mediaTimeNow)) { 3798 return; 3799 } 3800 } 3801 3802 size_t trimIndex; 3803 for (trimIndex = 0; trimIndex < mTimedBufferQueue.size(); trimIndex++) { 3804 if (mTimedBufferQueue[trimIndex].pts() > mediaTimeNow) 3805 break; 3806 } 3807 3808 if (trimIndex) { 3809 mTimedBufferQueue.removeItemsAt(0, trimIndex); 3810 } 3811} 3812 3813status_t AudioFlinger::PlaybackThread::TimedTrack::queueTimedBuffer( 3814 const sp<IMemory>& buffer, int64_t pts) { 3815 3816 { 3817 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3818 if (!mMediaTimeTransformValid) 3819 return INVALID_OPERATION; 3820 } 3821 3822 Mutex::Autolock _l(mTimedBufferQueueLock); 3823 3824 mTimedBufferQueue.add(TimedBuffer(buffer, pts)); 3825 3826 return NO_ERROR; 3827} 3828 3829status_t AudioFlinger::PlaybackThread::TimedTrack::setMediaTimeTransform( 3830 const LinearTransform& xform, TimedAudioTrack::TargetTimeline target) { 3831 3832 ALOGV("%s az=%lld bz=%lld n=%d d=%u tgt=%d", __PRETTY_FUNCTION__, 3833 xform.a_zero, xform.b_zero, xform.a_to_b_numer, xform.a_to_b_denom, 3834 target); 3835 3836 if (!(target == TimedAudioTrack::LOCAL_TIME || 3837 target == TimedAudioTrack::COMMON_TIME)) { 3838 return BAD_VALUE; 3839 } 3840 3841 Mutex::Autolock lock(mMediaTimeTransformLock); 3842 mMediaTimeTransform = xform; 3843 mMediaTimeTransformTarget = target; 3844 mMediaTimeTransformValid = true; 3845 3846 return NO_ERROR; 3847} 3848 3849#define min(a, b) ((a) < (b) ? (a) : (b)) 3850 3851// implementation of getNextBuffer for tracks whose buffers have timestamps 3852status_t AudioFlinger::PlaybackThread::TimedTrack::getNextBuffer( 3853 AudioBufferProvider::Buffer* buffer, int64_t pts) 3854{ 3855 if (pts == AudioBufferProvider::kInvalidPTS) { 3856 buffer->raw = 0; 3857 buffer->frameCount = 0; 3858 return INVALID_OPERATION; 3859 } 3860 3861 Mutex::Autolock _l(mTimedBufferQueueLock); 3862 3863 while (true) { 3864 3865 // if we have no timed buffers, then fail 3866 if (mTimedBufferQueue.isEmpty()) { 3867 buffer->raw = 0; 3868 buffer->frameCount = 0; 3869 return NOT_ENOUGH_DATA; 3870 } 3871 3872 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 3873 3874 // calculate the PTS of the head of the timed buffer queue expressed in 3875 // local time 3876 int64_t headLocalPTS; 3877 { 3878 Mutex::Autolock mttLock(mMediaTimeTransformLock); 3879 3880 assert(mMediaTimeTransformValid); 3881 3882 if (mMediaTimeTransform.a_to_b_denom == 0) { 3883 // the transform represents a pause, so yield silence 3884 timedYieldSilence(buffer->frameCount, buffer); 3885 return NO_ERROR; 3886 } 3887 3888 int64_t transformedPTS; 3889 if (!mMediaTimeTransform.doForwardTransform(head.pts(), 3890 &transformedPTS)) { 3891 // the transform failed. this shouldn't happen, but if it does 3892 // then just drop this buffer 3893 ALOGW("timedGetNextBuffer transform failed"); 3894 buffer->raw = 0; 3895 buffer->frameCount = 0; 3896 mTimedBufferQueue.removeAt(0); 3897 return NO_ERROR; 3898 } 3899 3900 if (mMediaTimeTransformTarget == TimedAudioTrack::COMMON_TIME) { 3901 if (OK != mCCHelper.commonTimeToLocalTime(transformedPTS, 3902 &headLocalPTS)) { 3903 buffer->raw = 0; 3904 buffer->frameCount = 0; 3905 return INVALID_OPERATION; 3906 } 3907 } else { 3908 headLocalPTS = transformedPTS; 3909 } 3910 } 3911 3912 // adjust the head buffer's PTS to reflect the portion of the head buffer 3913 // that has already been consumed 3914 int64_t effectivePTS = headLocalPTS + 3915 ((head.position() / mCblk->frameSize) * mLocalTimeFreq / sampleRate()); 3916 3917 // Calculate the delta in samples between the head of the input buffer 3918 // queue and the start of the next output buffer that will be written. 3919 // If the transformation fails because of over or underflow, it means 3920 // that the sample's position in the output stream is so far out of 3921 // whack that it should just be dropped. 3922 int64_t sampleDelta; 3923 if (llabs(effectivePTS - pts) >= (static_cast<int64_t>(1) << 31)) { 3924 ALOGV("*** head buffer is too far from PTS: dropped buffer"); 3925 mTimedBufferQueue.removeAt(0); 3926 continue; 3927 } 3928 if (!mLocalTimeToSampleTransform.doForwardTransform( 3929 (effectivePTS - pts) << 32, &sampleDelta)) { 3930 ALOGV("*** too late during sample rate transform: dropped buffer"); 3931 mTimedBufferQueue.removeAt(0); 3932 continue; 3933 } 3934 3935 ALOGV("*** %s head.pts=%lld head.pos=%d pts=%lld sampleDelta=[%d.%08x]", 3936 __PRETTY_FUNCTION__, head.pts(), head.position(), pts, 3937 static_cast<int32_t>((sampleDelta >= 0 ? 0 : 1) + (sampleDelta >> 32)), 3938 static_cast<uint32_t>(sampleDelta & 0xFFFFFFFF)); 3939 3940 // if the delta between the ideal placement for the next input sample and 3941 // the current output position is within this threshold, then we will 3942 // concatenate the next input samples to the previous output 3943 const int64_t kSampleContinuityThreshold = 3944 (static_cast<int64_t>(sampleRate()) << 32) / 10; 3945 3946 // if this is the first buffer of audio that we're emitting from this track 3947 // then it should be almost exactly on time. 3948 const int64_t kSampleStartupThreshold = 1LL << 32; 3949 3950 if ((mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleContinuityThreshold) || 3951 (!mTimedAudioOutputOnTime && llabs(sampleDelta) <= kSampleStartupThreshold)) { 3952 // the next input is close enough to being on time, so concatenate it 3953 // with the last output 3954 timedYieldSamples(buffer); 3955 3956 ALOGV("*** on time: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3957 return NO_ERROR; 3958 } else if (sampleDelta > 0) { 3959 // the gap between the current output position and the proper start of 3960 // the next input sample is too big, so fill it with silence 3961 uint32_t framesUntilNextInput = (sampleDelta + 0x80000000) >> 32; 3962 3963 timedYieldSilence(framesUntilNextInput, buffer); 3964 ALOGV("*** silence: frameCount=%u", buffer->frameCount); 3965 return NO_ERROR; 3966 } else { 3967 // the next input sample is late 3968 uint32_t lateFrames = static_cast<uint32_t>(-((sampleDelta + 0x80000000) >> 32)); 3969 size_t onTimeSamplePosition = 3970 head.position() + lateFrames * mCblk->frameSize; 3971 3972 if (onTimeSamplePosition > head.buffer()->size()) { 3973 // all the remaining samples in the head are too late, so 3974 // drop it and move on 3975 ALOGV("*** too late: dropped buffer"); 3976 mTimedBufferQueue.removeAt(0); 3977 continue; 3978 } else { 3979 // skip over the late samples 3980 head.setPosition(onTimeSamplePosition); 3981 3982 // yield the available samples 3983 timedYieldSamples(buffer); 3984 3985 ALOGV("*** late: head.pos=%d frameCount=%u", head.position(), buffer->frameCount); 3986 return NO_ERROR; 3987 } 3988 } 3989 } 3990} 3991 3992// Yield samples from the timed buffer queue head up to the given output 3993// buffer's capacity. 3994// 3995// Caller must hold mTimedBufferQueueLock 3996void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSamples( 3997 AudioBufferProvider::Buffer* buffer) { 3998 3999 const TimedBuffer& head = mTimedBufferQueue[0]; 4000 4001 buffer->raw = (static_cast<uint8_t*>(head.buffer()->pointer()) + 4002 head.position()); 4003 4004 uint32_t framesLeftInHead = ((head.buffer()->size() - head.position()) / 4005 mCblk->frameSize); 4006 size_t framesRequested = buffer->frameCount; 4007 buffer->frameCount = min(framesLeftInHead, framesRequested); 4008 4009 mTimedAudioOutputOnTime = true; 4010} 4011 4012// Yield samples of silence up to the given output buffer's capacity 4013// 4014// Caller must hold mTimedBufferQueueLock 4015void AudioFlinger::PlaybackThread::TimedTrack::timedYieldSilence( 4016 uint32_t numFrames, AudioBufferProvider::Buffer* buffer) { 4017 4018 // lazily allocate a buffer filled with silence 4019 if (mTimedSilenceBufferSize < numFrames * mCblk->frameSize) { 4020 delete [] mTimedSilenceBuffer; 4021 mTimedSilenceBufferSize = numFrames * mCblk->frameSize; 4022 mTimedSilenceBuffer = new uint8_t[mTimedSilenceBufferSize]; 4023 memset(mTimedSilenceBuffer, 0, mTimedSilenceBufferSize); 4024 } 4025 4026 buffer->raw = mTimedSilenceBuffer; 4027 size_t framesRequested = buffer->frameCount; 4028 buffer->frameCount = min(numFrames, framesRequested); 4029 4030 mTimedAudioOutputOnTime = false; 4031} 4032 4033// AudioBufferProvider interface 4034void AudioFlinger::PlaybackThread::TimedTrack::releaseBuffer( 4035 AudioBufferProvider::Buffer* buffer) { 4036 4037 Mutex::Autolock _l(mTimedBufferQueueLock); 4038 4039 // If the buffer which was just released is part of the buffer at the head 4040 // of the queue, be sure to update the amt of the buffer which has been 4041 // consumed. If the buffer being returned is not part of the head of the 4042 // queue, its either because the buffer is part of the silence buffer, or 4043 // because the head of the timed queue was trimmed after the mixer called 4044 // getNextBuffer but before the mixer called releaseBuffer. 4045 if ((buffer->raw != mTimedSilenceBuffer) && mTimedBufferQueue.size()) { 4046 TimedBuffer& head = mTimedBufferQueue.editItemAt(0); 4047 4048 void* start = head.buffer()->pointer(); 4049 void* end = (char *) head.buffer()->pointer() + head.buffer()->size(); 4050 4051 if ((buffer->raw >= start) && (buffer->raw <= end)) { 4052 head.setPosition(head.position() + 4053 (buffer->frameCount * mCblk->frameSize)); 4054 if (static_cast<size_t>(head.position()) >= head.buffer()->size()) { 4055 mTimedBufferQueue.removeAt(0); 4056 } 4057 } 4058 } 4059 4060 buffer->raw = 0; 4061 buffer->frameCount = 0; 4062} 4063 4064uint32_t AudioFlinger::PlaybackThread::TimedTrack::framesReady() const { 4065 Mutex::Autolock _l(mTimedBufferQueueLock); 4066 4067 uint32_t frames = 0; 4068 for (size_t i = 0; i < mTimedBufferQueue.size(); i++) { 4069 const TimedBuffer& tb = mTimedBufferQueue[i]; 4070 frames += (tb.buffer()->size() - tb.position()) / mCblk->frameSize; 4071 } 4072 4073 return frames; 4074} 4075 4076AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer() 4077 : mPTS(0), mPosition(0) {} 4078 4079AudioFlinger::PlaybackThread::TimedTrack::TimedBuffer::TimedBuffer( 4080 const sp<IMemory>& buffer, int64_t pts) 4081 : mBuffer(buffer), mPTS(pts), mPosition(0) {} 4082 4083// ---------------------------------------------------------------------------- 4084 4085// RecordTrack constructor must be called with AudioFlinger::mLock held 4086AudioFlinger::RecordThread::RecordTrack::RecordTrack( 4087 RecordThread *thread, 4088 const sp<Client>& client, 4089 uint32_t sampleRate, 4090 audio_format_t format, 4091 uint32_t channelMask, 4092 int frameCount, 4093 int sessionId) 4094 : TrackBase(thread, client, sampleRate, format, 4095 channelMask, frameCount, 0 /*sharedBuffer*/, sessionId), 4096 mOverflow(false) 4097{ 4098 if (mCblk != NULL) { 4099 ALOGV("RecordTrack constructor, size %d", (int)mBufferEnd - (int)mBuffer); 4100 if (format == AUDIO_FORMAT_PCM_16_BIT) { 4101 mCblk->frameSize = mChannelCount * sizeof(int16_t); 4102 } else if (format == AUDIO_FORMAT_PCM_8_BIT) { 4103 mCblk->frameSize = mChannelCount * sizeof(int8_t); 4104 } else { 4105 mCblk->frameSize = sizeof(int8_t); 4106 } 4107 } 4108} 4109 4110AudioFlinger::RecordThread::RecordTrack::~RecordTrack() 4111{ 4112 sp<ThreadBase> thread = mThread.promote(); 4113 if (thread != 0) { 4114 AudioSystem::releaseInput(thread->id()); 4115 } 4116} 4117 4118// AudioBufferProvider interface 4119status_t AudioFlinger::RecordThread::RecordTrack::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 4120{ 4121 audio_track_cblk_t* cblk = this->cblk(); 4122 uint32_t framesAvail; 4123 uint32_t framesReq = buffer->frameCount; 4124 4125 // Check if last stepServer failed, try to step now 4126 if (mStepServerFailed) { 4127 if (!step()) goto getNextBuffer_exit; 4128 ALOGV("stepServer recovered"); 4129 mStepServerFailed = false; 4130 } 4131 4132 framesAvail = cblk->framesAvailable_l(); 4133 4134 if (CC_LIKELY(framesAvail)) { 4135 uint32_t s = cblk->server; 4136 uint32_t bufferEnd = cblk->serverBase + cblk->frameCount; 4137 4138 if (framesReq > framesAvail) { 4139 framesReq = framesAvail; 4140 } 4141 if (s + framesReq > bufferEnd) { 4142 framesReq = bufferEnd - s; 4143 } 4144 4145 buffer->raw = getBuffer(s, framesReq); 4146 if (buffer->raw == NULL) goto getNextBuffer_exit; 4147 4148 buffer->frameCount = framesReq; 4149 return NO_ERROR; 4150 } 4151 4152getNextBuffer_exit: 4153 buffer->raw = NULL; 4154 buffer->frameCount = 0; 4155 return NOT_ENOUGH_DATA; 4156} 4157 4158status_t AudioFlinger::RecordThread::RecordTrack::start(pid_t tid) 4159{ 4160 sp<ThreadBase> thread = mThread.promote(); 4161 if (thread != 0) { 4162 RecordThread *recordThread = (RecordThread *)thread.get(); 4163 return recordThread->start(this, tid); 4164 } else { 4165 return BAD_VALUE; 4166 } 4167} 4168 4169void AudioFlinger::RecordThread::RecordTrack::stop() 4170{ 4171 sp<ThreadBase> thread = mThread.promote(); 4172 if (thread != 0) { 4173 RecordThread *recordThread = (RecordThread *)thread.get(); 4174 recordThread->stop(this); 4175 TrackBase::reset(); 4176 // Force overerrun condition to avoid false overrun callback until first data is 4177 // read from buffer 4178 android_atomic_or(CBLK_UNDERRUN_ON, &mCblk->flags); 4179 } 4180} 4181 4182void AudioFlinger::RecordThread::RecordTrack::dump(char* buffer, size_t size) 4183{ 4184 snprintf(buffer, size, " %05d %03u 0x%08x %05d %04u %01d %05u %08x %08x\n", 4185 (mClient == 0) ? getpid_cached : mClient->pid(), 4186 mFormat, 4187 mChannelMask, 4188 mSessionId, 4189 mFrameCount, 4190 mState, 4191 mCblk->sampleRate, 4192 mCblk->server, 4193 mCblk->user); 4194} 4195 4196 4197// ---------------------------------------------------------------------------- 4198 4199AudioFlinger::PlaybackThread::OutputTrack::OutputTrack( 4200 PlaybackThread *playbackThread, 4201 DuplicatingThread *sourceThread, 4202 uint32_t sampleRate, 4203 audio_format_t format, 4204 uint32_t channelMask, 4205 int frameCount) 4206 : Track(playbackThread, NULL, AUDIO_STREAM_CNT, sampleRate, format, channelMask, frameCount, NULL, 0), 4207 mActive(false), mSourceThread(sourceThread) 4208{ 4209 4210 if (mCblk != NULL) { 4211 mCblk->flags |= CBLK_DIRECTION_OUT; 4212 mCblk->buffers = (char*)mCblk + sizeof(audio_track_cblk_t); 4213 mOutBuffer.frameCount = 0; 4214 playbackThread->mTracks.add(this); 4215 ALOGV("OutputTrack constructor mCblk %p, mBuffer %p, mCblk->buffers %p, " \ 4216 "mCblk->frameCount %d, mCblk->sampleRate %d, mChannelMask 0x%08x mBufferEnd %p", 4217 mCblk, mBuffer, mCblk->buffers, 4218 mCblk->frameCount, mCblk->sampleRate, mChannelMask, mBufferEnd); 4219 } else { 4220 ALOGW("Error creating output track on thread %p", playbackThread); 4221 } 4222} 4223 4224AudioFlinger::PlaybackThread::OutputTrack::~OutputTrack() 4225{ 4226 clearBufferQueue(); 4227} 4228 4229status_t AudioFlinger::PlaybackThread::OutputTrack::start(pid_t tid) 4230{ 4231 status_t status = Track::start(tid); 4232 if (status != NO_ERROR) { 4233 return status; 4234 } 4235 4236 mActive = true; 4237 mRetryCount = 127; 4238 return status; 4239} 4240 4241void AudioFlinger::PlaybackThread::OutputTrack::stop() 4242{ 4243 Track::stop(); 4244 clearBufferQueue(); 4245 mOutBuffer.frameCount = 0; 4246 mActive = false; 4247} 4248 4249bool AudioFlinger::PlaybackThread::OutputTrack::write(int16_t* data, uint32_t frames) 4250{ 4251 Buffer *pInBuffer; 4252 Buffer inBuffer; 4253 uint32_t channelCount = mChannelCount; 4254 bool outputBufferFull = false; 4255 inBuffer.frameCount = frames; 4256 inBuffer.i16 = data; 4257 4258 uint32_t waitTimeLeftMs = mSourceThread->waitTimeMs(); 4259 4260 if (!mActive && frames != 0) { 4261 start(0); 4262 sp<ThreadBase> thread = mThread.promote(); 4263 if (thread != 0) { 4264 MixerThread *mixerThread = (MixerThread *)thread.get(); 4265 if (mCblk->frameCount > frames){ 4266 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4267 uint32_t startFrames = (mCblk->frameCount - frames); 4268 pInBuffer = new Buffer; 4269 pInBuffer->mBuffer = new int16_t[startFrames * channelCount]; 4270 pInBuffer->frameCount = startFrames; 4271 pInBuffer->i16 = pInBuffer->mBuffer; 4272 memset(pInBuffer->raw, 0, startFrames * channelCount * sizeof(int16_t)); 4273 mBufferQueue.add(pInBuffer); 4274 } else { 4275 ALOGW ("OutputTrack::write() %p no more buffers in queue", this); 4276 } 4277 } 4278 } 4279 } 4280 4281 while (waitTimeLeftMs) { 4282 // First write pending buffers, then new data 4283 if (mBufferQueue.size()) { 4284 pInBuffer = mBufferQueue.itemAt(0); 4285 } else { 4286 pInBuffer = &inBuffer; 4287 } 4288 4289 if (pInBuffer->frameCount == 0) { 4290 break; 4291 } 4292 4293 if (mOutBuffer.frameCount == 0) { 4294 mOutBuffer.frameCount = pInBuffer->frameCount; 4295 nsecs_t startTime = systemTime(); 4296 if (obtainBuffer(&mOutBuffer, waitTimeLeftMs) == (status_t)NO_MORE_BUFFERS) { 4297 ALOGV ("OutputTrack::write() %p thread %p no more output buffers", this, mThread.unsafe_get()); 4298 outputBufferFull = true; 4299 break; 4300 } 4301 uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime); 4302 if (waitTimeLeftMs >= waitTimeMs) { 4303 waitTimeLeftMs -= waitTimeMs; 4304 } else { 4305 waitTimeLeftMs = 0; 4306 } 4307 } 4308 4309 uint32_t outFrames = pInBuffer->frameCount > mOutBuffer.frameCount ? mOutBuffer.frameCount : pInBuffer->frameCount; 4310 memcpy(mOutBuffer.raw, pInBuffer->raw, outFrames * channelCount * sizeof(int16_t)); 4311 mCblk->stepUser(outFrames); 4312 pInBuffer->frameCount -= outFrames; 4313 pInBuffer->i16 += outFrames * channelCount; 4314 mOutBuffer.frameCount -= outFrames; 4315 mOutBuffer.i16 += outFrames * channelCount; 4316 4317 if (pInBuffer->frameCount == 0) { 4318 if (mBufferQueue.size()) { 4319 mBufferQueue.removeAt(0); 4320 delete [] pInBuffer->mBuffer; 4321 delete pInBuffer; 4322 ALOGV("OutputTrack::write() %p thread %p released overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4323 } else { 4324 break; 4325 } 4326 } 4327 } 4328 4329 // If we could not write all frames, allocate a buffer and queue it for next time. 4330 if (inBuffer.frameCount) { 4331 sp<ThreadBase> thread = mThread.promote(); 4332 if (thread != 0 && !thread->standby()) { 4333 if (mBufferQueue.size() < kMaxOverFlowBuffers) { 4334 pInBuffer = new Buffer; 4335 pInBuffer->mBuffer = new int16_t[inBuffer.frameCount * channelCount]; 4336 pInBuffer->frameCount = inBuffer.frameCount; 4337 pInBuffer->i16 = pInBuffer->mBuffer; 4338 memcpy(pInBuffer->raw, inBuffer.raw, inBuffer.frameCount * channelCount * sizeof(int16_t)); 4339 mBufferQueue.add(pInBuffer); 4340 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %d", this, mThread.unsafe_get(), mBufferQueue.size()); 4341 } else { 4342 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers", mThread.unsafe_get(), this); 4343 } 4344 } 4345 } 4346 4347 // Calling write() with a 0 length buffer, means that no more data will be written: 4348 // If no more buffers are pending, fill output track buffer to make sure it is started 4349 // by output mixer. 4350 if (frames == 0 && mBufferQueue.size() == 0) { 4351 if (mCblk->user < mCblk->frameCount) { 4352 frames = mCblk->frameCount - mCblk->user; 4353 pInBuffer = new Buffer; 4354 pInBuffer->mBuffer = new int16_t[frames * channelCount]; 4355 pInBuffer->frameCount = frames; 4356 pInBuffer->i16 = pInBuffer->mBuffer; 4357 memset(pInBuffer->raw, 0, frames * channelCount * sizeof(int16_t)); 4358 mBufferQueue.add(pInBuffer); 4359 } else if (mActive) { 4360 stop(); 4361 } 4362 } 4363 4364 return outputBufferFull; 4365} 4366 4367status_t AudioFlinger::PlaybackThread::OutputTrack::obtainBuffer(AudioBufferProvider::Buffer* buffer, uint32_t waitTimeMs) 4368{ 4369 int active; 4370 status_t result; 4371 audio_track_cblk_t* cblk = mCblk; 4372 uint32_t framesReq = buffer->frameCount; 4373 4374// ALOGV("OutputTrack::obtainBuffer user %d, server %d", cblk->user, cblk->server); 4375 buffer->frameCount = 0; 4376 4377 uint32_t framesAvail = cblk->framesAvailable(); 4378 4379 4380 if (framesAvail == 0) { 4381 Mutex::Autolock _l(cblk->lock); 4382 goto start_loop_here; 4383 while (framesAvail == 0) { 4384 active = mActive; 4385 if (CC_UNLIKELY(!active)) { 4386 ALOGV("Not active and NO_MORE_BUFFERS"); 4387 return NO_MORE_BUFFERS; 4388 } 4389 result = cblk->cv.waitRelative(cblk->lock, milliseconds(waitTimeMs)); 4390 if (result != NO_ERROR) { 4391 return NO_MORE_BUFFERS; 4392 } 4393 // read the server count again 4394 start_loop_here: 4395 framesAvail = cblk->framesAvailable_l(); 4396 } 4397 } 4398 4399// if (framesAvail < framesReq) { 4400// return NO_MORE_BUFFERS; 4401// } 4402 4403 if (framesReq > framesAvail) { 4404 framesReq = framesAvail; 4405 } 4406 4407 uint32_t u = cblk->user; 4408 uint32_t bufferEnd = cblk->userBase + cblk->frameCount; 4409 4410 if (u + framesReq > bufferEnd) { 4411 framesReq = bufferEnd - u; 4412 } 4413 4414 buffer->frameCount = framesReq; 4415 buffer->raw = (void *)cblk->buffer(u); 4416 return NO_ERROR; 4417} 4418 4419 4420void AudioFlinger::PlaybackThread::OutputTrack::clearBufferQueue() 4421{ 4422 size_t size = mBufferQueue.size(); 4423 4424 for (size_t i = 0; i < size; i++) { 4425 Buffer *pBuffer = mBufferQueue.itemAt(i); 4426 delete [] pBuffer->mBuffer; 4427 delete pBuffer; 4428 } 4429 mBufferQueue.clear(); 4430} 4431 4432// ---------------------------------------------------------------------------- 4433 4434AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 4435 : RefBase(), 4436 mAudioFlinger(audioFlinger), 4437 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 4438 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 4439 mPid(pid), 4440 mTimedTrackCount(0) 4441{ 4442 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 4443} 4444 4445// Client destructor must be called with AudioFlinger::mLock held 4446AudioFlinger::Client::~Client() 4447{ 4448 mAudioFlinger->removeClient_l(mPid); 4449} 4450 4451sp<MemoryDealer> AudioFlinger::Client::heap() const 4452{ 4453 return mMemoryDealer; 4454} 4455 4456// Reserve one of the limited slots for a timed audio track associated 4457// with this client 4458bool AudioFlinger::Client::reserveTimedTrack() 4459{ 4460 const int kMaxTimedTracksPerClient = 4; 4461 4462 Mutex::Autolock _l(mTimedTrackLock); 4463 4464 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 4465 ALOGW("can not create timed track - pid %d has exceeded the limit", 4466 mPid); 4467 return false; 4468 } 4469 4470 mTimedTrackCount++; 4471 return true; 4472} 4473 4474// Release a slot for a timed audio track 4475void AudioFlinger::Client::releaseTimedTrack() 4476{ 4477 Mutex::Autolock _l(mTimedTrackLock); 4478 mTimedTrackCount--; 4479} 4480 4481// ---------------------------------------------------------------------------- 4482 4483AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 4484 const sp<IAudioFlingerClient>& client, 4485 pid_t pid) 4486 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 4487{ 4488} 4489 4490AudioFlinger::NotificationClient::~NotificationClient() 4491{ 4492} 4493 4494void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who) 4495{ 4496 sp<NotificationClient> keep(this); 4497 mAudioFlinger->removeNotificationClient(mPid); 4498} 4499 4500// ---------------------------------------------------------------------------- 4501 4502AudioFlinger::TrackHandle::TrackHandle(const sp<AudioFlinger::PlaybackThread::Track>& track) 4503 : BnAudioTrack(), 4504 mTrack(track) 4505{ 4506} 4507 4508AudioFlinger::TrackHandle::~TrackHandle() { 4509 // just stop the track on deletion, associated resources 4510 // will be freed from the main thread once all pending buffers have 4511 // been played. Unless it's not in the active track list, in which 4512 // case we free everything now... 4513 mTrack->destroy(); 4514} 4515 4516sp<IMemory> AudioFlinger::TrackHandle::getCblk() const { 4517 return mTrack->getCblk(); 4518} 4519 4520status_t AudioFlinger::TrackHandle::start(pid_t tid) { 4521 return mTrack->start(tid); 4522} 4523 4524void AudioFlinger::TrackHandle::stop() { 4525 mTrack->stop(); 4526} 4527 4528void AudioFlinger::TrackHandle::flush() { 4529 mTrack->flush(); 4530} 4531 4532void AudioFlinger::TrackHandle::mute(bool e) { 4533 mTrack->mute(e); 4534} 4535 4536void AudioFlinger::TrackHandle::pause() { 4537 mTrack->pause(); 4538} 4539 4540status_t AudioFlinger::TrackHandle::attachAuxEffect(int EffectId) 4541{ 4542 return mTrack->attachAuxEffect(EffectId); 4543} 4544 4545status_t AudioFlinger::TrackHandle::allocateTimedBuffer(size_t size, 4546 sp<IMemory>* buffer) { 4547 if (!mTrack->isTimedTrack()) 4548 return INVALID_OPERATION; 4549 4550 PlaybackThread::TimedTrack* tt = 4551 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4552 return tt->allocateTimedBuffer(size, buffer); 4553} 4554 4555status_t AudioFlinger::TrackHandle::queueTimedBuffer(const sp<IMemory>& buffer, 4556 int64_t pts) { 4557 if (!mTrack->isTimedTrack()) 4558 return INVALID_OPERATION; 4559 4560 PlaybackThread::TimedTrack* tt = 4561 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4562 return tt->queueTimedBuffer(buffer, pts); 4563} 4564 4565status_t AudioFlinger::TrackHandle::setMediaTimeTransform( 4566 const LinearTransform& xform, int target) { 4567 4568 if (!mTrack->isTimedTrack()) 4569 return INVALID_OPERATION; 4570 4571 PlaybackThread::TimedTrack* tt = 4572 reinterpret_cast<PlaybackThread::TimedTrack*>(mTrack.get()); 4573 return tt->setMediaTimeTransform( 4574 xform, static_cast<TimedAudioTrack::TargetTimeline>(target)); 4575} 4576 4577status_t AudioFlinger::TrackHandle::onTransact( 4578 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4579{ 4580 return BnAudioTrack::onTransact(code, data, reply, flags); 4581} 4582 4583// ---------------------------------------------------------------------------- 4584 4585sp<IAudioRecord> AudioFlinger::openRecord( 4586 pid_t pid, 4587 audio_io_handle_t input, 4588 uint32_t sampleRate, 4589 audio_format_t format, 4590 uint32_t channelMask, 4591 int frameCount, 4592 // FIXME dead, remove from IAudioFlinger 4593 uint32_t flags, 4594 int *sessionId, 4595 status_t *status) 4596{ 4597 sp<RecordThread::RecordTrack> recordTrack; 4598 sp<RecordHandle> recordHandle; 4599 sp<Client> client; 4600 status_t lStatus; 4601 RecordThread *thread; 4602 size_t inFrameCount; 4603 int lSessionId; 4604 4605 // check calling permissions 4606 if (!recordingAllowed()) { 4607 lStatus = PERMISSION_DENIED; 4608 goto Exit; 4609 } 4610 4611 // add client to list 4612 { // scope for mLock 4613 Mutex::Autolock _l(mLock); 4614 thread = checkRecordThread_l(input); 4615 if (thread == NULL) { 4616 lStatus = BAD_VALUE; 4617 goto Exit; 4618 } 4619 4620 client = registerPid_l(pid); 4621 4622 // If no audio session id is provided, create one here 4623 if (sessionId != NULL && *sessionId != AUDIO_SESSION_OUTPUT_MIX) { 4624 lSessionId = *sessionId; 4625 } else { 4626 lSessionId = nextUniqueId(); 4627 if (sessionId != NULL) { 4628 *sessionId = lSessionId; 4629 } 4630 } 4631 // create new record track. The record track uses one track in mHardwareMixerThread by convention. 4632 recordTrack = thread->createRecordTrack_l(client, 4633 sampleRate, 4634 format, 4635 channelMask, 4636 frameCount, 4637 lSessionId, 4638 &lStatus); 4639 } 4640 if (lStatus != NO_ERROR) { 4641 // remove local strong reference to Client before deleting the RecordTrack so that the Client 4642 // destructor is called by the TrackBase destructor with mLock held 4643 client.clear(); 4644 recordTrack.clear(); 4645 goto Exit; 4646 } 4647 4648 // return to handle to client 4649 recordHandle = new RecordHandle(recordTrack); 4650 lStatus = NO_ERROR; 4651 4652Exit: 4653 if (status) { 4654 *status = lStatus; 4655 } 4656 return recordHandle; 4657} 4658 4659// ---------------------------------------------------------------------------- 4660 4661AudioFlinger::RecordHandle::RecordHandle(const sp<AudioFlinger::RecordThread::RecordTrack>& recordTrack) 4662 : BnAudioRecord(), 4663 mRecordTrack(recordTrack) 4664{ 4665} 4666 4667AudioFlinger::RecordHandle::~RecordHandle() { 4668 stop(); 4669} 4670 4671sp<IMemory> AudioFlinger::RecordHandle::getCblk() const { 4672 return mRecordTrack->getCblk(); 4673} 4674 4675status_t AudioFlinger::RecordHandle::start(pid_t tid) { 4676 ALOGV("RecordHandle::start()"); 4677 return mRecordTrack->start(tid); 4678} 4679 4680void AudioFlinger::RecordHandle::stop() { 4681 ALOGV("RecordHandle::stop()"); 4682 mRecordTrack->stop(); 4683} 4684 4685status_t AudioFlinger::RecordHandle::onTransact( 4686 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 4687{ 4688 return BnAudioRecord::onTransact(code, data, reply, flags); 4689} 4690 4691// ---------------------------------------------------------------------------- 4692 4693AudioFlinger::RecordThread::RecordThread(const sp<AudioFlinger>& audioFlinger, 4694 AudioStreamIn *input, 4695 uint32_t sampleRate, 4696 uint32_t channels, 4697 audio_io_handle_t id, 4698 uint32_t device) : 4699 ThreadBase(audioFlinger, id, device, RECORD), 4700 mInput(input), mTrack(NULL), mResampler(NULL), mRsmpOutBuffer(NULL), mRsmpInBuffer(NULL), 4701 // mRsmpInIndex and mInputBytes set by readInputParameters() 4702 mReqChannelCount(popcount(channels)), 4703 mReqSampleRate(sampleRate) 4704 // mBytesRead is only meaningful while active, and so is cleared in start() 4705 // (but might be better to also clear here for dump?) 4706{ 4707 snprintf(mName, kNameLength, "AudioIn_%X", id); 4708 4709 readInputParameters(); 4710} 4711 4712 4713AudioFlinger::RecordThread::~RecordThread() 4714{ 4715 delete[] mRsmpInBuffer; 4716 delete mResampler; 4717 delete[] mRsmpOutBuffer; 4718} 4719 4720void AudioFlinger::RecordThread::onFirstRef() 4721{ 4722 run(mName, PRIORITY_URGENT_AUDIO); 4723} 4724 4725status_t AudioFlinger::RecordThread::readyToRun() 4726{ 4727 status_t status = initCheck(); 4728 ALOGW_IF(status != NO_ERROR,"RecordThread %p could not initialize", this); 4729 return status; 4730} 4731 4732bool AudioFlinger::RecordThread::threadLoop() 4733{ 4734 AudioBufferProvider::Buffer buffer; 4735 sp<RecordTrack> activeTrack; 4736 Vector< sp<EffectChain> > effectChains; 4737 4738 nsecs_t lastWarning = 0; 4739 4740 acquireWakeLock(); 4741 4742 // start recording 4743 while (!exitPending()) { 4744 4745 processConfigEvents(); 4746 4747 { // scope for mLock 4748 Mutex::Autolock _l(mLock); 4749 checkForNewParameters_l(); 4750 if (mActiveTrack == 0 && mConfigEvents.isEmpty()) { 4751 if (!mStandby) { 4752 mInput->stream->common.standby(&mInput->stream->common); 4753 mStandby = true; 4754 } 4755 4756 if (exitPending()) break; 4757 4758 releaseWakeLock_l(); 4759 ALOGV("RecordThread: loop stopping"); 4760 // go to sleep 4761 mWaitWorkCV.wait(mLock); 4762 ALOGV("RecordThread: loop starting"); 4763 acquireWakeLock_l(); 4764 continue; 4765 } 4766 if (mActiveTrack != 0) { 4767 if (mActiveTrack->mState == TrackBase::PAUSING) { 4768 if (!mStandby) { 4769 mInput->stream->common.standby(&mInput->stream->common); 4770 mStandby = true; 4771 } 4772 mActiveTrack.clear(); 4773 mStartStopCond.broadcast(); 4774 } else if (mActiveTrack->mState == TrackBase::RESUMING) { 4775 if (mReqChannelCount != mActiveTrack->channelCount()) { 4776 mActiveTrack.clear(); 4777 mStartStopCond.broadcast(); 4778 } else if (mBytesRead != 0) { 4779 // record start succeeds only if first read from audio input 4780 // succeeds 4781 if (mBytesRead > 0) { 4782 mActiveTrack->mState = TrackBase::ACTIVE; 4783 } else { 4784 mActiveTrack.clear(); 4785 } 4786 mStartStopCond.broadcast(); 4787 } 4788 mStandby = false; 4789 } 4790 } 4791 lockEffectChains_l(effectChains); 4792 } 4793 4794 if (mActiveTrack != 0) { 4795 if (mActiveTrack->mState != TrackBase::ACTIVE && 4796 mActiveTrack->mState != TrackBase::RESUMING) { 4797 unlockEffectChains(effectChains); 4798 usleep(kRecordThreadSleepUs); 4799 continue; 4800 } 4801 for (size_t i = 0; i < effectChains.size(); i ++) { 4802 effectChains[i]->process_l(); 4803 } 4804 4805 buffer.frameCount = mFrameCount; 4806 if (CC_LIKELY(mActiveTrack->getNextBuffer(&buffer) == NO_ERROR)) { 4807 size_t framesOut = buffer.frameCount; 4808 if (mResampler == NULL) { 4809 // no resampling 4810 while (framesOut) { 4811 size_t framesIn = mFrameCount - mRsmpInIndex; 4812 if (framesIn) { 4813 int8_t *src = (int8_t *)mRsmpInBuffer + mRsmpInIndex * mFrameSize; 4814 int8_t *dst = buffer.i8 + (buffer.frameCount - framesOut) * mActiveTrack->mCblk->frameSize; 4815 if (framesIn > framesOut) 4816 framesIn = framesOut; 4817 mRsmpInIndex += framesIn; 4818 framesOut -= framesIn; 4819 if ((int)mChannelCount == mReqChannelCount || 4820 mFormat != AUDIO_FORMAT_PCM_16_BIT) { 4821 memcpy(dst, src, framesIn * mFrameSize); 4822 } else { 4823 int16_t *src16 = (int16_t *)src; 4824 int16_t *dst16 = (int16_t *)dst; 4825 if (mChannelCount == 1) { 4826 while (framesIn--) { 4827 *dst16++ = *src16; 4828 *dst16++ = *src16++; 4829 } 4830 } else { 4831 while (framesIn--) { 4832 *dst16++ = (int16_t)(((int32_t)*src16 + (int32_t)*(src16 + 1)) >> 1); 4833 src16 += 2; 4834 } 4835 } 4836 } 4837 } 4838 if (framesOut && mFrameCount == mRsmpInIndex) { 4839 if (framesOut == mFrameCount && 4840 ((int)mChannelCount == mReqChannelCount || mFormat != AUDIO_FORMAT_PCM_16_BIT)) { 4841 mBytesRead = mInput->stream->read(mInput->stream, buffer.raw, mInputBytes); 4842 framesOut = 0; 4843 } else { 4844 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 4845 mRsmpInIndex = 0; 4846 } 4847 if (mBytesRead < 0) { 4848 ALOGE("Error reading audio input"); 4849 if (mActiveTrack->mState == TrackBase::ACTIVE) { 4850 // Force input into standby so that it tries to 4851 // recover at next read attempt 4852 mInput->stream->common.standby(&mInput->stream->common); 4853 usleep(kRecordThreadSleepUs); 4854 } 4855 mRsmpInIndex = mFrameCount; 4856 framesOut = 0; 4857 buffer.frameCount = 0; 4858 } 4859 } 4860 } 4861 } else { 4862 // resampling 4863 4864 memset(mRsmpOutBuffer, 0, framesOut * 2 * sizeof(int32_t)); 4865 // alter output frame count as if we were expecting stereo samples 4866 if (mChannelCount == 1 && mReqChannelCount == 1) { 4867 framesOut >>= 1; 4868 } 4869 mResampler->resample(mRsmpOutBuffer, framesOut, this); 4870 // ditherAndClamp() works as long as all buffers returned by mActiveTrack->getNextBuffer() 4871 // are 32 bit aligned which should be always true. 4872 if (mChannelCount == 2 && mReqChannelCount == 1) { 4873 ditherAndClamp(mRsmpOutBuffer, mRsmpOutBuffer, framesOut); 4874 // the resampler always outputs stereo samples: do post stereo to mono conversion 4875 int16_t *src = (int16_t *)mRsmpOutBuffer; 4876 int16_t *dst = buffer.i16; 4877 while (framesOut--) { 4878 *dst++ = (int16_t)(((int32_t)*src + (int32_t)*(src + 1)) >> 1); 4879 src += 2; 4880 } 4881 } else { 4882 ditherAndClamp((int32_t *)buffer.raw, mRsmpOutBuffer, framesOut); 4883 } 4884 4885 } 4886 mActiveTrack->releaseBuffer(&buffer); 4887 mActiveTrack->overflow(); 4888 } 4889 // client isn't retrieving buffers fast enough 4890 else { 4891 if (!mActiveTrack->setOverflow()) { 4892 nsecs_t now = systemTime(); 4893 if ((now - lastWarning) > kWarningThrottleNs) { 4894 ALOGW("RecordThread: buffer overflow"); 4895 lastWarning = now; 4896 } 4897 } 4898 // Release the processor for a while before asking for a new buffer. 4899 // This will give the application more chance to read from the buffer and 4900 // clear the overflow. 4901 usleep(kRecordThreadSleepUs); 4902 } 4903 } 4904 // enable changes in effect chain 4905 unlockEffectChains(effectChains); 4906 effectChains.clear(); 4907 } 4908 4909 if (!mStandby) { 4910 mInput->stream->common.standby(&mInput->stream->common); 4911 } 4912 mActiveTrack.clear(); 4913 4914 mStartStopCond.broadcast(); 4915 4916 releaseWakeLock(); 4917 4918 ALOGV("RecordThread %p exiting", this); 4919 return false; 4920} 4921 4922 4923sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRecordTrack_l( 4924 const sp<AudioFlinger::Client>& client, 4925 uint32_t sampleRate, 4926 audio_format_t format, 4927 int channelMask, 4928 int frameCount, 4929 int sessionId, 4930 status_t *status) 4931{ 4932 sp<RecordTrack> track; 4933 status_t lStatus; 4934 4935 lStatus = initCheck(); 4936 if (lStatus != NO_ERROR) { 4937 ALOGE("Audio driver not initialized."); 4938 goto Exit; 4939 } 4940 4941 { // scope for mLock 4942 Mutex::Autolock _l(mLock); 4943 4944 track = new RecordTrack(this, client, sampleRate, 4945 format, channelMask, frameCount, sessionId); 4946 4947 if (track->getCblk() == 0) { 4948 lStatus = NO_MEMORY; 4949 goto Exit; 4950 } 4951 4952 mTrack = track.get(); 4953 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 4954 bool suspend = audio_is_bluetooth_sco_device( 4955 (audio_devices_t)(mDevice & AUDIO_DEVICE_IN_ALL)) && mAudioFlinger->btNrecIsOff(); 4956 setEffectSuspended_l(FX_IID_AEC, suspend, sessionId); 4957 setEffectSuspended_l(FX_IID_NS, suspend, sessionId); 4958 } 4959 lStatus = NO_ERROR; 4960 4961Exit: 4962 if (status) { 4963 *status = lStatus; 4964 } 4965 return track; 4966} 4967 4968status_t AudioFlinger::RecordThread::start(RecordThread::RecordTrack* recordTrack, pid_t tid) 4969{ 4970 ALOGV("RecordThread::start tid=%d", tid); 4971 sp <ThreadBase> strongMe = this; 4972 status_t status = NO_ERROR; 4973 { 4974 AutoMutex lock(mLock); 4975 if (mActiveTrack != 0) { 4976 if (recordTrack != mActiveTrack.get()) { 4977 status = -EBUSY; 4978 } else if (mActiveTrack->mState == TrackBase::PAUSING) { 4979 mActiveTrack->mState = TrackBase::ACTIVE; 4980 } 4981 return status; 4982 } 4983 4984 recordTrack->mState = TrackBase::IDLE; 4985 mActiveTrack = recordTrack; 4986 mLock.unlock(); 4987 status_t status = AudioSystem::startInput(mId); 4988 mLock.lock(); 4989 if (status != NO_ERROR) { 4990 mActiveTrack.clear(); 4991 return status; 4992 } 4993 mRsmpInIndex = mFrameCount; 4994 mBytesRead = 0; 4995 if (mResampler != NULL) { 4996 mResampler->reset(); 4997 } 4998 mActiveTrack->mState = TrackBase::RESUMING; 4999 // signal thread to start 5000 ALOGV("Signal record thread"); 5001 mWaitWorkCV.signal(); 5002 // do not wait for mStartStopCond if exiting 5003 if (exitPending()) { 5004 mActiveTrack.clear(); 5005 status = INVALID_OPERATION; 5006 goto startError; 5007 } 5008 mStartStopCond.wait(mLock); 5009 if (mActiveTrack == 0) { 5010 ALOGV("Record failed to start"); 5011 status = BAD_VALUE; 5012 goto startError; 5013 } 5014 ALOGV("Record started OK"); 5015 return status; 5016 } 5017startError: 5018 AudioSystem::stopInput(mId); 5019 return status; 5020} 5021 5022void AudioFlinger::RecordThread::stop(RecordThread::RecordTrack* recordTrack) { 5023 ALOGV("RecordThread::stop"); 5024 sp <ThreadBase> strongMe = this; 5025 { 5026 AutoMutex lock(mLock); 5027 if (mActiveTrack != 0 && recordTrack == mActiveTrack.get()) { 5028 mActiveTrack->mState = TrackBase::PAUSING; 5029 // do not wait for mStartStopCond if exiting 5030 if (exitPending()) { 5031 return; 5032 } 5033 mStartStopCond.wait(mLock); 5034 // if we have been restarted, recordTrack == mActiveTrack.get() here 5035 if (mActiveTrack == 0 || recordTrack != mActiveTrack.get()) { 5036 mLock.unlock(); 5037 AudioSystem::stopInput(mId); 5038 mLock.lock(); 5039 ALOGV("Record stopped OK"); 5040 } 5041 } 5042 } 5043} 5044 5045status_t AudioFlinger::RecordThread::dump(int fd, const Vector<String16>& args) 5046{ 5047 const size_t SIZE = 256; 5048 char buffer[SIZE]; 5049 String8 result; 5050 5051 snprintf(buffer, SIZE, "\nInput thread %p internals\n", this); 5052 result.append(buffer); 5053 5054 if (mActiveTrack != 0) { 5055 result.append("Active Track:\n"); 5056 result.append(" Clien Fmt Chn mask Session Buf S SRate Serv User\n"); 5057 mActiveTrack->dump(buffer, SIZE); 5058 result.append(buffer); 5059 5060 snprintf(buffer, SIZE, "In index: %d\n", mRsmpInIndex); 5061 result.append(buffer); 5062 snprintf(buffer, SIZE, "In size: %d\n", mInputBytes); 5063 result.append(buffer); 5064 snprintf(buffer, SIZE, "Resampling: %d\n", (mResampler != NULL)); 5065 result.append(buffer); 5066 snprintf(buffer, SIZE, "Out channel count: %d\n", mReqChannelCount); 5067 result.append(buffer); 5068 snprintf(buffer, SIZE, "Out sample rate: %d\n", mReqSampleRate); 5069 result.append(buffer); 5070 5071 5072 } else { 5073 result.append("No record client\n"); 5074 } 5075 write(fd, result.string(), result.size()); 5076 5077 dumpBase(fd, args); 5078 dumpEffectChains(fd, args); 5079 5080 return NO_ERROR; 5081} 5082 5083// AudioBufferProvider interface 5084status_t AudioFlinger::RecordThread::getNextBuffer(AudioBufferProvider::Buffer* buffer, int64_t pts) 5085{ 5086 size_t framesReq = buffer->frameCount; 5087 size_t framesReady = mFrameCount - mRsmpInIndex; 5088 int channelCount; 5089 5090 if (framesReady == 0) { 5091 mBytesRead = mInput->stream->read(mInput->stream, mRsmpInBuffer, mInputBytes); 5092 if (mBytesRead < 0) { 5093 ALOGE("RecordThread::getNextBuffer() Error reading audio input"); 5094 if (mActiveTrack->mState == TrackBase::ACTIVE) { 5095 // Force input into standby so that it tries to 5096 // recover at next read attempt 5097 mInput->stream->common.standby(&mInput->stream->common); 5098 usleep(kRecordThreadSleepUs); 5099 } 5100 buffer->raw = NULL; 5101 buffer->frameCount = 0; 5102 return NOT_ENOUGH_DATA; 5103 } 5104 mRsmpInIndex = 0; 5105 framesReady = mFrameCount; 5106 } 5107 5108 if (framesReq > framesReady) { 5109 framesReq = framesReady; 5110 } 5111 5112 if (mChannelCount == 1 && mReqChannelCount == 2) { 5113 channelCount = 1; 5114 } else { 5115 channelCount = 2; 5116 } 5117 buffer->raw = mRsmpInBuffer + mRsmpInIndex * channelCount; 5118 buffer->frameCount = framesReq; 5119 return NO_ERROR; 5120} 5121 5122// AudioBufferProvider interface 5123void AudioFlinger::RecordThread::releaseBuffer(AudioBufferProvider::Buffer* buffer) 5124{ 5125 mRsmpInIndex += buffer->frameCount; 5126 buffer->frameCount = 0; 5127} 5128 5129bool AudioFlinger::RecordThread::checkForNewParameters_l() 5130{ 5131 bool reconfig = false; 5132 5133 while (!mNewParameters.isEmpty()) { 5134 status_t status = NO_ERROR; 5135 String8 keyValuePair = mNewParameters[0]; 5136 AudioParameter param = AudioParameter(keyValuePair); 5137 int value; 5138 audio_format_t reqFormat = mFormat; 5139 int reqSamplingRate = mReqSampleRate; 5140 int reqChannelCount = mReqChannelCount; 5141 5142 if (param.getInt(String8(AudioParameter::keySamplingRate), value) == NO_ERROR) { 5143 reqSamplingRate = value; 5144 reconfig = true; 5145 } 5146 if (param.getInt(String8(AudioParameter::keyFormat), value) == NO_ERROR) { 5147 reqFormat = (audio_format_t) value; 5148 reconfig = true; 5149 } 5150 if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) { 5151 reqChannelCount = popcount(value); 5152 reconfig = true; 5153 } 5154 if (param.getInt(String8(AudioParameter::keyFrameCount), value) == NO_ERROR) { 5155 // do not accept frame count changes if tracks are open as the track buffer 5156 // size depends on frame count and correct behavior would not be guaranteed 5157 // if frame count is changed after track creation 5158 if (mActiveTrack != 0) { 5159 status = INVALID_OPERATION; 5160 } else { 5161 reconfig = true; 5162 } 5163 } 5164 if (param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) { 5165 // forward device change to effects that have requested to be 5166 // aware of attached audio device. 5167 for (size_t i = 0; i < mEffectChains.size(); i++) { 5168 mEffectChains[i]->setDevice_l(value); 5169 } 5170 // store input device and output device but do not forward output device to audio HAL. 5171 // Note that status is ignored by the caller for output device 5172 // (see AudioFlinger::setParameters() 5173 if (value & AUDIO_DEVICE_OUT_ALL) { 5174 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_OUT_ALL); 5175 status = BAD_VALUE; 5176 } else { 5177 mDevice &= (uint32_t)~(value & AUDIO_DEVICE_IN_ALL); 5178 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 5179 if (mTrack != NULL) { 5180 bool suspend = audio_is_bluetooth_sco_device( 5181 (audio_devices_t)value) && mAudioFlinger->btNrecIsOff(); 5182 setEffectSuspended_l(FX_IID_AEC, suspend, mTrack->sessionId()); 5183 setEffectSuspended_l(FX_IID_NS, suspend, mTrack->sessionId()); 5184 } 5185 } 5186 mDevice |= (uint32_t)value; 5187 } 5188 if (status == NO_ERROR) { 5189 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5190 if (status == INVALID_OPERATION) { 5191 mInput->stream->common.standby(&mInput->stream->common); 5192 status = mInput->stream->common.set_parameters(&mInput->stream->common, keyValuePair.string()); 5193 } 5194 if (reconfig) { 5195 if (status == BAD_VALUE && 5196 reqFormat == mInput->stream->common.get_format(&mInput->stream->common) && 5197 reqFormat == AUDIO_FORMAT_PCM_16_BIT && 5198 ((int)mInput->stream->common.get_sample_rate(&mInput->stream->common) <= (2 * reqSamplingRate)) && 5199 (popcount(mInput->stream->common.get_channels(&mInput->stream->common)) < 3) && 5200 (reqChannelCount < 3)) { 5201 status = NO_ERROR; 5202 } 5203 if (status == NO_ERROR) { 5204 readInputParameters(); 5205 sendConfigEvent_l(AudioSystem::INPUT_CONFIG_CHANGED); 5206 } 5207 } 5208 } 5209 5210 mNewParameters.removeAt(0); 5211 5212 mParamStatus = status; 5213 mParamCond.signal(); 5214 // wait for condition with time out in case the thread calling ThreadBase::setParameters() 5215 // already timed out waiting for the status and will never signal the condition. 5216 mWaitWorkCV.waitRelative(mLock, kSetParametersTimeoutNs); 5217 } 5218 return reconfig; 5219} 5220 5221String8 AudioFlinger::RecordThread::getParameters(const String8& keys) 5222{ 5223 char *s; 5224 String8 out_s8 = String8(); 5225 5226 Mutex::Autolock _l(mLock); 5227 if (initCheck() != NO_ERROR) { 5228 return out_s8; 5229 } 5230 5231 s = mInput->stream->common.get_parameters(&mInput->stream->common, keys.string()); 5232 out_s8 = String8(s); 5233 free(s); 5234 return out_s8; 5235} 5236 5237void AudioFlinger::RecordThread::audioConfigChanged_l(int event, int param) { 5238 AudioSystem::OutputDescriptor desc; 5239 void *param2 = NULL; 5240 5241 switch (event) { 5242 case AudioSystem::INPUT_OPENED: 5243 case AudioSystem::INPUT_CONFIG_CHANGED: 5244 desc.channels = mChannelMask; 5245 desc.samplingRate = mSampleRate; 5246 desc.format = mFormat; 5247 desc.frameCount = mFrameCount; 5248 desc.latency = 0; 5249 param2 = &desc; 5250 break; 5251 5252 case AudioSystem::INPUT_CLOSED: 5253 default: 5254 break; 5255 } 5256 mAudioFlinger->audioConfigChanged_l(event, mId, param2); 5257} 5258 5259void AudioFlinger::RecordThread::readInputParameters() 5260{ 5261 delete mRsmpInBuffer; 5262 // mRsmpInBuffer is always assigned a new[] below 5263 delete mRsmpOutBuffer; 5264 mRsmpOutBuffer = NULL; 5265 delete mResampler; 5266 mResampler = NULL; 5267 5268 mSampleRate = mInput->stream->common.get_sample_rate(&mInput->stream->common); 5269 mChannelMask = mInput->stream->common.get_channels(&mInput->stream->common); 5270 mChannelCount = (uint16_t)popcount(mChannelMask); 5271 mFormat = mInput->stream->common.get_format(&mInput->stream->common); 5272 mFrameSize = audio_stream_frame_size(&mInput->stream->common); 5273 mInputBytes = mInput->stream->common.get_buffer_size(&mInput->stream->common); 5274 mFrameCount = mInputBytes / mFrameSize; 5275 mRsmpInBuffer = new int16_t[mFrameCount * mChannelCount]; 5276 5277 if (mSampleRate != mReqSampleRate && mChannelCount < 3 && mReqChannelCount < 3) 5278 { 5279 int channelCount; 5280 // optmization: if mono to mono, use the resampler in stereo to stereo mode to avoid 5281 // stereo to mono post process as the resampler always outputs stereo. 5282 if (mChannelCount == 1 && mReqChannelCount == 2) { 5283 channelCount = 1; 5284 } else { 5285 channelCount = 2; 5286 } 5287 mResampler = AudioResampler::create(16, channelCount, mReqSampleRate); 5288 mResampler->setSampleRate(mSampleRate); 5289 mResampler->setVolume(AudioMixer::UNITY_GAIN, AudioMixer::UNITY_GAIN); 5290 mRsmpOutBuffer = new int32_t[mFrameCount * 2]; 5291 5292 // optmization: if mono to mono, alter input frame count as if we were inputing stereo samples 5293 if (mChannelCount == 1 && mReqChannelCount == 1) { 5294 mFrameCount >>= 1; 5295 } 5296 5297 } 5298 mRsmpInIndex = mFrameCount; 5299} 5300 5301unsigned int AudioFlinger::RecordThread::getInputFramesLost() 5302{ 5303 Mutex::Autolock _l(mLock); 5304 if (initCheck() != NO_ERROR) { 5305 return 0; 5306 } 5307 5308 return mInput->stream->get_input_frames_lost(mInput->stream); 5309} 5310 5311uint32_t AudioFlinger::RecordThread::hasAudioSession(int sessionId) 5312{ 5313 Mutex::Autolock _l(mLock); 5314 uint32_t result = 0; 5315 if (getEffectChain_l(sessionId) != 0) { 5316 result = EFFECT_SESSION; 5317 } 5318 5319 if (mTrack != NULL && sessionId == mTrack->sessionId()) { 5320 result |= TRACK_SESSION; 5321 } 5322 5323 return result; 5324} 5325 5326AudioFlinger::RecordThread::RecordTrack* AudioFlinger::RecordThread::track() 5327{ 5328 Mutex::Autolock _l(mLock); 5329 return mTrack; 5330} 5331 5332AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::getInput() const 5333{ 5334 Mutex::Autolock _l(mLock); 5335 return mInput; 5336} 5337 5338AudioFlinger::AudioStreamIn* AudioFlinger::RecordThread::clearInput() 5339{ 5340 Mutex::Autolock _l(mLock); 5341 AudioStreamIn *input = mInput; 5342 mInput = NULL; 5343 return input; 5344} 5345 5346// this method must always be called either with ThreadBase mLock held or inside the thread loop 5347audio_stream_t* AudioFlinger::RecordThread::stream() 5348{ 5349 if (mInput == NULL) { 5350 return NULL; 5351 } 5352 return &mInput->stream->common; 5353} 5354 5355 5356// ---------------------------------------------------------------------------- 5357 5358audio_io_handle_t AudioFlinger::openOutput(uint32_t *pDevices, 5359 uint32_t *pSamplingRate, 5360 audio_format_t *pFormat, 5361 uint32_t *pChannels, 5362 uint32_t *pLatencyMs, 5363 uint32_t flags) 5364{ 5365 status_t status; 5366 PlaybackThread *thread = NULL; 5367 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5368 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5369 uint32_t channels = pChannels ? *pChannels : 0; 5370 uint32_t latency = pLatencyMs ? *pLatencyMs : 0; 5371 audio_stream_out_t *outStream; 5372 audio_hw_device_t *outHwDev; 5373 5374 ALOGV("openOutput(), Device %x, SamplingRate %d, Format %d, Channels %x, flags %x", 5375 pDevices ? *pDevices : 0, 5376 samplingRate, 5377 format, 5378 channels, 5379 flags); 5380 5381 if (pDevices == NULL || *pDevices == 0) { 5382 return 0; 5383 } 5384 5385 Mutex::Autolock _l(mLock); 5386 5387 outHwDev = findSuitableHwDev_l(*pDevices); 5388 if (outHwDev == NULL) 5389 return 0; 5390 5391 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 5392 status = outHwDev->open_output_stream(outHwDev, *pDevices, &format, 5393 &channels, &samplingRate, &outStream); 5394 mHardwareStatus = AUDIO_HW_IDLE; 5395 ALOGV("openOutput() openOutputStream returned output %p, SamplingRate %d, Format %d, Channels %x, status %d", 5396 outStream, 5397 samplingRate, 5398 format, 5399 channels, 5400 status); 5401 5402 if (outStream != NULL) { 5403 AudioStreamOut *output = new AudioStreamOut(outHwDev, outStream); 5404 audio_io_handle_t id = nextUniqueId(); 5405 5406 if ((flags & AUDIO_POLICY_OUTPUT_FLAG_DIRECT) || 5407 (format != AUDIO_FORMAT_PCM_16_BIT) || 5408 (channels != AUDIO_CHANNEL_OUT_STEREO)) { 5409 thread = new DirectOutputThread(this, output, id, *pDevices); 5410 ALOGV("openOutput() created direct output: ID %d thread %p", id, thread); 5411 } else { 5412 thread = new MixerThread(this, output, id, *pDevices); 5413 ALOGV("openOutput() created mixer output: ID %d thread %p", id, thread); 5414 } 5415 mPlaybackThreads.add(id, thread); 5416 5417 if (pSamplingRate != NULL) *pSamplingRate = samplingRate; 5418 if (pFormat != NULL) *pFormat = format; 5419 if (pChannels != NULL) *pChannels = channels; 5420 if (pLatencyMs != NULL) *pLatencyMs = thread->latency(); 5421 5422 // notify client processes of the new output creation 5423 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5424 return id; 5425 } 5426 5427 return 0; 5428} 5429 5430audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 5431 audio_io_handle_t output2) 5432{ 5433 Mutex::Autolock _l(mLock); 5434 MixerThread *thread1 = checkMixerThread_l(output1); 5435 MixerThread *thread2 = checkMixerThread_l(output2); 5436 5437 if (thread1 == NULL || thread2 == NULL) { 5438 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, output2); 5439 return 0; 5440 } 5441 5442 audio_io_handle_t id = nextUniqueId(); 5443 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 5444 thread->addOutputTrack(thread2); 5445 mPlaybackThreads.add(id, thread); 5446 // notify client processes of the new output creation 5447 thread->audioConfigChanged_l(AudioSystem::OUTPUT_OPENED); 5448 return id; 5449} 5450 5451status_t AudioFlinger::closeOutput(audio_io_handle_t output) 5452{ 5453 // keep strong reference on the playback thread so that 5454 // it is not destroyed while exit() is executed 5455 sp <PlaybackThread> thread; 5456 { 5457 Mutex::Autolock _l(mLock); 5458 thread = checkPlaybackThread_l(output); 5459 if (thread == NULL) { 5460 return BAD_VALUE; 5461 } 5462 5463 ALOGV("closeOutput() %d", output); 5464 5465 if (thread->type() == ThreadBase::MIXER) { 5466 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5467 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 5468 DuplicatingThread *dupThread = (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 5469 dupThread->removeOutputTrack((MixerThread *)thread.get()); 5470 } 5471 } 5472 } 5473 audioConfigChanged_l(AudioSystem::OUTPUT_CLOSED, output, NULL); 5474 mPlaybackThreads.removeItem(output); 5475 } 5476 thread->exit(); 5477 // The thread entity (active unit of execution) is no longer running here, 5478 // but the ThreadBase container still exists. 5479 5480 if (thread->type() != ThreadBase::DUPLICATING) { 5481 AudioStreamOut *out = thread->clearOutput(); 5482 assert(out != NULL); 5483 // from now on thread->mOutput is NULL 5484 out->hwDev->close_output_stream(out->hwDev, out->stream); 5485 delete out; 5486 } 5487 return NO_ERROR; 5488} 5489 5490status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 5491{ 5492 Mutex::Autolock _l(mLock); 5493 PlaybackThread *thread = checkPlaybackThread_l(output); 5494 5495 if (thread == NULL) { 5496 return BAD_VALUE; 5497 } 5498 5499 ALOGV("suspendOutput() %d", output); 5500 thread->suspend(); 5501 5502 return NO_ERROR; 5503} 5504 5505status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 5506{ 5507 Mutex::Autolock _l(mLock); 5508 PlaybackThread *thread = checkPlaybackThread_l(output); 5509 5510 if (thread == NULL) { 5511 return BAD_VALUE; 5512 } 5513 5514 ALOGV("restoreOutput() %d", output); 5515 5516 thread->restore(); 5517 5518 return NO_ERROR; 5519} 5520 5521audio_io_handle_t AudioFlinger::openInput(uint32_t *pDevices, 5522 uint32_t *pSamplingRate, 5523 audio_format_t *pFormat, 5524 uint32_t *pChannels, 5525 audio_in_acoustics_t acoustics) 5526{ 5527 status_t status; 5528 RecordThread *thread = NULL; 5529 uint32_t samplingRate = pSamplingRate ? *pSamplingRate : 0; 5530 audio_format_t format = pFormat ? *pFormat : AUDIO_FORMAT_DEFAULT; 5531 uint32_t channels = pChannels ? *pChannels : 0; 5532 uint32_t reqSamplingRate = samplingRate; 5533 audio_format_t reqFormat = format; 5534 uint32_t reqChannels = channels; 5535 audio_stream_in_t *inStream; 5536 audio_hw_device_t *inHwDev; 5537 5538 if (pDevices == NULL || *pDevices == 0) { 5539 return 0; 5540 } 5541 5542 Mutex::Autolock _l(mLock); 5543 5544 inHwDev = findSuitableHwDev_l(*pDevices); 5545 if (inHwDev == NULL) 5546 return 0; 5547 5548 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5549 &channels, &samplingRate, 5550 acoustics, 5551 &inStream); 5552 ALOGV("openInput() openInputStream returned input %p, SamplingRate %d, Format %d, Channels %x, acoustics %x, status %d", 5553 inStream, 5554 samplingRate, 5555 format, 5556 channels, 5557 acoustics, 5558 status); 5559 5560 // If the input could not be opened with the requested parameters and we can handle the conversion internally, 5561 // try to open again with the proposed parameters. The AudioFlinger can resample the input and do mono to stereo 5562 // or stereo to mono conversions on 16 bit PCM inputs. 5563 if (inStream == NULL && status == BAD_VALUE && 5564 reqFormat == format && format == AUDIO_FORMAT_PCM_16_BIT && 5565 (samplingRate <= 2 * reqSamplingRate) && 5566 (popcount(channels) < 3) && (popcount(reqChannels) < 3)) { 5567 ALOGV("openInput() reopening with proposed sampling rate and channels"); 5568 status = inHwDev->open_input_stream(inHwDev, *pDevices, &format, 5569 &channels, &samplingRate, 5570 acoustics, 5571 &inStream); 5572 } 5573 5574 if (inStream != NULL) { 5575 AudioStreamIn *input = new AudioStreamIn(inHwDev, inStream); 5576 5577 audio_io_handle_t id = nextUniqueId(); 5578 // Start record thread 5579 // RecorThread require both input and output device indication to forward to audio 5580 // pre processing modules 5581 uint32_t device = (*pDevices) | primaryOutputDevice_l(); 5582 thread = new RecordThread(this, 5583 input, 5584 reqSamplingRate, 5585 reqChannels, 5586 id, 5587 device); 5588 mRecordThreads.add(id, thread); 5589 ALOGV("openInput() created record thread: ID %d thread %p", id, thread); 5590 if (pSamplingRate != NULL) *pSamplingRate = reqSamplingRate; 5591 if (pFormat != NULL) *pFormat = format; 5592 if (pChannels != NULL) *pChannels = reqChannels; 5593 5594 input->stream->common.standby(&input->stream->common); 5595 5596 // notify client processes of the new input creation 5597 thread->audioConfigChanged_l(AudioSystem::INPUT_OPENED); 5598 return id; 5599 } 5600 5601 return 0; 5602} 5603 5604status_t AudioFlinger::closeInput(audio_io_handle_t input) 5605{ 5606 // keep strong reference on the record thread so that 5607 // it is not destroyed while exit() is executed 5608 sp <RecordThread> thread; 5609 { 5610 Mutex::Autolock _l(mLock); 5611 thread = checkRecordThread_l(input); 5612 if (thread == NULL) { 5613 return BAD_VALUE; 5614 } 5615 5616 ALOGV("closeInput() %d", input); 5617 audioConfigChanged_l(AudioSystem::INPUT_CLOSED, input, NULL); 5618 mRecordThreads.removeItem(input); 5619 } 5620 thread->exit(); 5621 // The thread entity (active unit of execution) is no longer running here, 5622 // but the ThreadBase container still exists. 5623 5624 AudioStreamIn *in = thread->clearInput(); 5625 assert(in != NULL); 5626 // from now on thread->mInput is NULL 5627 in->hwDev->close_input_stream(in->hwDev, in->stream); 5628 delete in; 5629 5630 return NO_ERROR; 5631} 5632 5633status_t AudioFlinger::setStreamOutput(audio_stream_type_t stream, audio_io_handle_t output) 5634{ 5635 Mutex::Autolock _l(mLock); 5636 MixerThread *dstThread = checkMixerThread_l(output); 5637 if (dstThread == NULL) { 5638 ALOGW("setStreamOutput() bad output id %d", output); 5639 return BAD_VALUE; 5640 } 5641 5642 ALOGV("setStreamOutput() stream %d to output %d", stream, output); 5643 audioConfigChanged_l(AudioSystem::STREAM_CONFIG_CHANGED, output, &stream); 5644 5645 dstThread->setStreamValid(stream, true); 5646 5647 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5648 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5649 if (thread != dstThread && thread->type() != ThreadBase::DIRECT) { 5650 MixerThread *srcThread = (MixerThread *)thread; 5651 srcThread->setStreamValid(stream, false); 5652 srcThread->invalidateTracks(stream); 5653 } 5654 } 5655 5656 return NO_ERROR; 5657} 5658 5659 5660int AudioFlinger::newAudioSessionId() 5661{ 5662 return nextUniqueId(); 5663} 5664 5665void AudioFlinger::acquireAudioSessionId(int audioSession) 5666{ 5667 Mutex::Autolock _l(mLock); 5668 pid_t caller = IPCThreadState::self()->getCallingPid(); 5669 ALOGV("acquiring %d from %d", audioSession, caller); 5670 size_t num = mAudioSessionRefs.size(); 5671 for (size_t i = 0; i< num; i++) { 5672 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 5673 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5674 ref->mCnt++; 5675 ALOGV(" incremented refcount to %d", ref->mCnt); 5676 return; 5677 } 5678 } 5679 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 5680 ALOGV(" added new entry for %d", audioSession); 5681} 5682 5683void AudioFlinger::releaseAudioSessionId(int audioSession) 5684{ 5685 Mutex::Autolock _l(mLock); 5686 pid_t caller = IPCThreadState::self()->getCallingPid(); 5687 ALOGV("releasing %d from %d", audioSession, caller); 5688 size_t num = mAudioSessionRefs.size(); 5689 for (size_t i = 0; i< num; i++) { 5690 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 5691 if (ref->mSessionid == audioSession && ref->mPid == caller) { 5692 ref->mCnt--; 5693 ALOGV(" decremented refcount to %d", ref->mCnt); 5694 if (ref->mCnt == 0) { 5695 mAudioSessionRefs.removeAt(i); 5696 delete ref; 5697 purgeStaleEffects_l(); 5698 } 5699 return; 5700 } 5701 } 5702 ALOGW("session id %d not found for pid %d", audioSession, caller); 5703} 5704 5705void AudioFlinger::purgeStaleEffects_l() { 5706 5707 ALOGV("purging stale effects"); 5708 5709 Vector< sp<EffectChain> > chains; 5710 5711 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5712 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 5713 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5714 sp<EffectChain> ec = t->mEffectChains[j]; 5715 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 5716 chains.push(ec); 5717 } 5718 } 5719 } 5720 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5721 sp<RecordThread> t = mRecordThreads.valueAt(i); 5722 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 5723 sp<EffectChain> ec = t->mEffectChains[j]; 5724 chains.push(ec); 5725 } 5726 } 5727 5728 for (size_t i = 0; i < chains.size(); i++) { 5729 sp<EffectChain> ec = chains[i]; 5730 int sessionid = ec->sessionId(); 5731 sp<ThreadBase> t = ec->mThread.promote(); 5732 if (t == 0) { 5733 continue; 5734 } 5735 size_t numsessionrefs = mAudioSessionRefs.size(); 5736 bool found = false; 5737 for (size_t k = 0; k < numsessionrefs; k++) { 5738 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 5739 if (ref->mSessionid == sessionid) { 5740 ALOGV(" session %d still exists for %d with %d refs", 5741 sessionid, ref->mPid, ref->mCnt); 5742 found = true; 5743 break; 5744 } 5745 } 5746 if (!found) { 5747 // remove all effects from the chain 5748 while (ec->mEffects.size()) { 5749 sp<EffectModule> effect = ec->mEffects[0]; 5750 effect->unPin(); 5751 Mutex::Autolock _l (t->mLock); 5752 t->removeEffect_l(effect); 5753 for (size_t j = 0; j < effect->mHandles.size(); j++) { 5754 sp<EffectHandle> handle = effect->mHandles[j].promote(); 5755 if (handle != 0) { 5756 handle->mEffect.clear(); 5757 if (handle->mHasControl && handle->mEnabled) { 5758 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 5759 } 5760 } 5761 } 5762 AudioSystem::unregisterEffect(effect->id()); 5763 } 5764 } 5765 } 5766 return; 5767} 5768 5769// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 5770AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 5771{ 5772 return mPlaybackThreads.valueFor(output).get(); 5773} 5774 5775// checkMixerThread_l() must be called with AudioFlinger::mLock held 5776AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 5777{ 5778 PlaybackThread *thread = checkPlaybackThread_l(output); 5779 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 5780} 5781 5782// checkRecordThread_l() must be called with AudioFlinger::mLock held 5783AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 5784{ 5785 return mRecordThreads.valueFor(input).get(); 5786} 5787 5788uint32_t AudioFlinger::nextUniqueId() 5789{ 5790 return android_atomic_inc(&mNextUniqueId); 5791} 5792 5793AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 5794{ 5795 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5796 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 5797 AudioStreamOut *output = thread->getOutput(); 5798 if (output != NULL && output->hwDev == mPrimaryHardwareDev) { 5799 return thread; 5800 } 5801 } 5802 return NULL; 5803} 5804 5805uint32_t AudioFlinger::primaryOutputDevice_l() const 5806{ 5807 PlaybackThread *thread = primaryPlaybackThread_l(); 5808 5809 if (thread == NULL) { 5810 return 0; 5811 } 5812 5813 return thread->device(); 5814} 5815 5816 5817// ---------------------------------------------------------------------------- 5818// Effect management 5819// ---------------------------------------------------------------------------- 5820 5821 5822status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 5823{ 5824 Mutex::Autolock _l(mLock); 5825 return EffectQueryNumberEffects(numEffects); 5826} 5827 5828status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 5829{ 5830 Mutex::Autolock _l(mLock); 5831 return EffectQueryEffect(index, descriptor); 5832} 5833 5834status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 5835 effect_descriptor_t *descriptor) const 5836{ 5837 Mutex::Autolock _l(mLock); 5838 return EffectGetDescriptor(pUuid, descriptor); 5839} 5840 5841 5842sp<IEffect> AudioFlinger::createEffect(pid_t pid, 5843 effect_descriptor_t *pDesc, 5844 const sp<IEffectClient>& effectClient, 5845 int32_t priority, 5846 audio_io_handle_t io, 5847 int sessionId, 5848 status_t *status, 5849 int *id, 5850 int *enabled) 5851{ 5852 status_t lStatus = NO_ERROR; 5853 sp<EffectHandle> handle; 5854 effect_descriptor_t desc; 5855 5856 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 5857 pid, effectClient.get(), priority, sessionId, io); 5858 5859 if (pDesc == NULL) { 5860 lStatus = BAD_VALUE; 5861 goto Exit; 5862 } 5863 5864 // check audio settings permission for global effects 5865 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 5866 lStatus = PERMISSION_DENIED; 5867 goto Exit; 5868 } 5869 5870 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 5871 // that can only be created by audio policy manager (running in same process) 5872 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 5873 lStatus = PERMISSION_DENIED; 5874 goto Exit; 5875 } 5876 5877 if (io == 0) { 5878 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 5879 // output must be specified by AudioPolicyManager when using session 5880 // AUDIO_SESSION_OUTPUT_STAGE 5881 lStatus = BAD_VALUE; 5882 goto Exit; 5883 } else if (sessionId == AUDIO_SESSION_OUTPUT_MIX) { 5884 // if the output returned by getOutputForEffect() is removed before we lock the 5885 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 5886 // and we will exit safely 5887 io = AudioSystem::getOutputForEffect(&desc); 5888 } 5889 } 5890 5891 { 5892 Mutex::Autolock _l(mLock); 5893 5894 5895 if (!EffectIsNullUuid(&pDesc->uuid)) { 5896 // if uuid is specified, request effect descriptor 5897 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 5898 if (lStatus < 0) { 5899 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 5900 goto Exit; 5901 } 5902 } else { 5903 // if uuid is not specified, look for an available implementation 5904 // of the required type in effect factory 5905 if (EffectIsNullUuid(&pDesc->type)) { 5906 ALOGW("createEffect() no effect type"); 5907 lStatus = BAD_VALUE; 5908 goto Exit; 5909 } 5910 uint32_t numEffects = 0; 5911 effect_descriptor_t d; 5912 d.flags = 0; // prevent compiler warning 5913 bool found = false; 5914 5915 lStatus = EffectQueryNumberEffects(&numEffects); 5916 if (lStatus < 0) { 5917 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 5918 goto Exit; 5919 } 5920 for (uint32_t i = 0; i < numEffects; i++) { 5921 lStatus = EffectQueryEffect(i, &desc); 5922 if (lStatus < 0) { 5923 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 5924 continue; 5925 } 5926 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 5927 // If matching type found save effect descriptor. If the session is 5928 // 0 and the effect is not auxiliary, continue enumeration in case 5929 // an auxiliary version of this effect type is available 5930 found = true; 5931 memcpy(&d, &desc, sizeof(effect_descriptor_t)); 5932 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 5933 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5934 break; 5935 } 5936 } 5937 } 5938 if (!found) { 5939 lStatus = BAD_VALUE; 5940 ALOGW("createEffect() effect not found"); 5941 goto Exit; 5942 } 5943 // For same effect type, chose auxiliary version over insert version if 5944 // connect to output mix (Compliance to OpenSL ES) 5945 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 5946 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 5947 memcpy(&desc, &d, sizeof(effect_descriptor_t)); 5948 } 5949 } 5950 5951 // Do not allow auxiliary effects on a session different from 0 (output mix) 5952 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 5953 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 5954 lStatus = INVALID_OPERATION; 5955 goto Exit; 5956 } 5957 5958 // check recording permission for visualizer 5959 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 5960 !recordingAllowed()) { 5961 lStatus = PERMISSION_DENIED; 5962 goto Exit; 5963 } 5964 5965 // return effect descriptor 5966 memcpy(pDesc, &desc, sizeof(effect_descriptor_t)); 5967 5968 // If output is not specified try to find a matching audio session ID in one of the 5969 // output threads. 5970 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 5971 // because of code checking output when entering the function. 5972 // Note: io is never 0 when creating an effect on an input 5973 if (io == 0) { 5974 // look for the thread where the specified audio session is present 5975 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 5976 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5977 io = mPlaybackThreads.keyAt(i); 5978 break; 5979 } 5980 } 5981 if (io == 0) { 5982 for (size_t i = 0; i < mRecordThreads.size(); i++) { 5983 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 5984 io = mRecordThreads.keyAt(i); 5985 break; 5986 } 5987 } 5988 } 5989 // If no output thread contains the requested session ID, default to 5990 // first output. The effect chain will be moved to the correct output 5991 // thread when a track with the same session ID is created 5992 if (io == 0 && mPlaybackThreads.size()) { 5993 io = mPlaybackThreads.keyAt(0); 5994 } 5995 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 5996 } 5997 ThreadBase *thread = checkRecordThread_l(io); 5998 if (thread == NULL) { 5999 thread = checkPlaybackThread_l(io); 6000 if (thread == NULL) { 6001 ALOGE("createEffect() unknown output thread"); 6002 lStatus = BAD_VALUE; 6003 goto Exit; 6004 } 6005 } 6006 6007 sp<Client> client = registerPid_l(pid); 6008 6009 // create effect on selected output thread 6010 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 6011 &desc, enabled, &lStatus); 6012 if (handle != 0 && id != NULL) { 6013 *id = handle->id(); 6014 } 6015 } 6016 6017Exit: 6018 if(status) { 6019 *status = lStatus; 6020 } 6021 return handle; 6022} 6023 6024status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 6025 audio_io_handle_t dstOutput) 6026{ 6027 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 6028 sessionId, srcOutput, dstOutput); 6029 Mutex::Autolock _l(mLock); 6030 if (srcOutput == dstOutput) { 6031 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 6032 return NO_ERROR; 6033 } 6034 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 6035 if (srcThread == NULL) { 6036 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 6037 return BAD_VALUE; 6038 } 6039 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 6040 if (dstThread == NULL) { 6041 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 6042 return BAD_VALUE; 6043 } 6044 6045 Mutex::Autolock _dl(dstThread->mLock); 6046 Mutex::Autolock _sl(srcThread->mLock); 6047 moveEffectChain_l(sessionId, srcThread, dstThread, false); 6048 6049 return NO_ERROR; 6050} 6051 6052// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 6053status_t AudioFlinger::moveEffectChain_l(int sessionId, 6054 AudioFlinger::PlaybackThread *srcThread, 6055 AudioFlinger::PlaybackThread *dstThread, 6056 bool reRegister) 6057{ 6058 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 6059 sessionId, srcThread, dstThread); 6060 6061 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 6062 if (chain == 0) { 6063 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 6064 sessionId, srcThread); 6065 return INVALID_OPERATION; 6066 } 6067 6068 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 6069 // so that a new chain is created with correct parameters when first effect is added. This is 6070 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 6071 // removed. 6072 srcThread->removeEffectChain_l(chain); 6073 6074 // transfer all effects one by one so that new effect chain is created on new thread with 6075 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 6076 audio_io_handle_t dstOutput = dstThread->id(); 6077 sp<EffectChain> dstChain; 6078 uint32_t strategy = 0; // prevent compiler warning 6079 sp<EffectModule> effect = chain->getEffectFromId_l(0); 6080 while (effect != 0) { 6081 srcThread->removeEffect_l(effect); 6082 dstThread->addEffect_l(effect); 6083 // removeEffect_l() has stopped the effect if it was active so it must be restarted 6084 if (effect->state() == EffectModule::ACTIVE || 6085 effect->state() == EffectModule::STOPPING) { 6086 effect->start(); 6087 } 6088 // if the move request is not received from audio policy manager, the effect must be 6089 // re-registered with the new strategy and output 6090 if (dstChain == 0) { 6091 dstChain = effect->chain().promote(); 6092 if (dstChain == 0) { 6093 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 6094 srcThread->addEffect_l(effect); 6095 return NO_INIT; 6096 } 6097 strategy = dstChain->strategy(); 6098 } 6099 if (reRegister) { 6100 AudioSystem::unregisterEffect(effect->id()); 6101 AudioSystem::registerEffect(&effect->desc(), 6102 dstOutput, 6103 strategy, 6104 sessionId, 6105 effect->id()); 6106 } 6107 effect = chain->getEffectFromId_l(0); 6108 } 6109 6110 return NO_ERROR; 6111} 6112 6113 6114// PlaybackThread::createEffect_l() must be called with AudioFlinger::mLock held 6115sp<AudioFlinger::EffectHandle> AudioFlinger::ThreadBase::createEffect_l( 6116 const sp<AudioFlinger::Client>& client, 6117 const sp<IEffectClient>& effectClient, 6118 int32_t priority, 6119 int sessionId, 6120 effect_descriptor_t *desc, 6121 int *enabled, 6122 status_t *status 6123 ) 6124{ 6125 sp<EffectModule> effect; 6126 sp<EffectHandle> handle; 6127 status_t lStatus; 6128 sp<EffectChain> chain; 6129 bool chainCreated = false; 6130 bool effectCreated = false; 6131 bool effectRegistered = false; 6132 6133 lStatus = initCheck(); 6134 if (lStatus != NO_ERROR) { 6135 ALOGW("createEffect_l() Audio driver not initialized."); 6136 goto Exit; 6137 } 6138 6139 // Do not allow effects with session ID 0 on direct output or duplicating threads 6140 // TODO: add rule for hw accelerated effects on direct outputs with non PCM format 6141 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && mType != MIXER) { 6142 ALOGW("createEffect_l() Cannot add auxiliary effect %s to session %d", 6143 desc->name, sessionId); 6144 lStatus = BAD_VALUE; 6145 goto Exit; 6146 } 6147 // Only Pre processor effects are allowed on input threads and only on input threads 6148 if ((mType == RECORD) != ((desc->flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC)) { 6149 ALOGW("createEffect_l() effect %s (flags %08x) created on wrong thread type %d", 6150 desc->name, desc->flags, mType); 6151 lStatus = BAD_VALUE; 6152 goto Exit; 6153 } 6154 6155 ALOGV("createEffect_l() thread %p effect %s on session %d", this, desc->name, sessionId); 6156 6157 { // scope for mLock 6158 Mutex::Autolock _l(mLock); 6159 6160 // check for existing effect chain with the requested audio session 6161 chain = getEffectChain_l(sessionId); 6162 if (chain == 0) { 6163 // create a new chain for this session 6164 ALOGV("createEffect_l() new effect chain for session %d", sessionId); 6165 chain = new EffectChain(this, sessionId); 6166 addEffectChain_l(chain); 6167 chain->setStrategy(getStrategyForSession_l(sessionId)); 6168 chainCreated = true; 6169 } else { 6170 effect = chain->getEffectFromDesc_l(desc); 6171 } 6172 6173 ALOGV("createEffect_l() got effect %p on chain %p", effect.get(), chain.get()); 6174 6175 if (effect == 0) { 6176 int id = mAudioFlinger->nextUniqueId(); 6177 // Check CPU and memory usage 6178 lStatus = AudioSystem::registerEffect(desc, mId, chain->strategy(), sessionId, id); 6179 if (lStatus != NO_ERROR) { 6180 goto Exit; 6181 } 6182 effectRegistered = true; 6183 // create a new effect module if none present in the chain 6184 effect = new EffectModule(this, chain, desc, id, sessionId); 6185 lStatus = effect->status(); 6186 if (lStatus != NO_ERROR) { 6187 goto Exit; 6188 } 6189 lStatus = chain->addEffect_l(effect); 6190 if (lStatus != NO_ERROR) { 6191 goto Exit; 6192 } 6193 effectCreated = true; 6194 6195 effect->setDevice(mDevice); 6196 effect->setMode(mAudioFlinger->getMode()); 6197 } 6198 // create effect handle and connect it to effect module 6199 handle = new EffectHandle(effect, client, effectClient, priority); 6200 lStatus = effect->addHandle(handle); 6201 if (enabled != NULL) { 6202 *enabled = (int)effect->isEnabled(); 6203 } 6204 } 6205 6206Exit: 6207 if (lStatus != NO_ERROR && lStatus != ALREADY_EXISTS) { 6208 Mutex::Autolock _l(mLock); 6209 if (effectCreated) { 6210 chain->removeEffect_l(effect); 6211 } 6212 if (effectRegistered) { 6213 AudioSystem::unregisterEffect(effect->id()); 6214 } 6215 if (chainCreated) { 6216 removeEffectChain_l(chain); 6217 } 6218 handle.clear(); 6219 } 6220 6221 if(status) { 6222 *status = lStatus; 6223 } 6224 return handle; 6225} 6226 6227sp<AudioFlinger::EffectModule> AudioFlinger::ThreadBase::getEffect_l(int sessionId, int effectId) 6228{ 6229 sp<EffectChain> chain = getEffectChain_l(sessionId); 6230 return chain != 0 ? chain->getEffectFromId_l(effectId) : 0; 6231} 6232 6233// PlaybackThread::addEffect_l() must be called with AudioFlinger::mLock and 6234// PlaybackThread::mLock held 6235status_t AudioFlinger::ThreadBase::addEffect_l(const sp<EffectModule>& effect) 6236{ 6237 // check for existing effect chain with the requested audio session 6238 int sessionId = effect->sessionId(); 6239 sp<EffectChain> chain = getEffectChain_l(sessionId); 6240 bool chainCreated = false; 6241 6242 if (chain == 0) { 6243 // create a new chain for this session 6244 ALOGV("addEffect_l() new effect chain for session %d", sessionId); 6245 chain = new EffectChain(this, sessionId); 6246 addEffectChain_l(chain); 6247 chain->setStrategy(getStrategyForSession_l(sessionId)); 6248 chainCreated = true; 6249 } 6250 ALOGV("addEffect_l() %p chain %p effect %p", this, chain.get(), effect.get()); 6251 6252 if (chain->getEffectFromId_l(effect->id()) != 0) { 6253 ALOGW("addEffect_l() %p effect %s already present in chain %p", 6254 this, effect->desc().name, chain.get()); 6255 return BAD_VALUE; 6256 } 6257 6258 status_t status = chain->addEffect_l(effect); 6259 if (status != NO_ERROR) { 6260 if (chainCreated) { 6261 removeEffectChain_l(chain); 6262 } 6263 return status; 6264 } 6265 6266 effect->setDevice(mDevice); 6267 effect->setMode(mAudioFlinger->getMode()); 6268 return NO_ERROR; 6269} 6270 6271void AudioFlinger::ThreadBase::removeEffect_l(const sp<EffectModule>& effect) { 6272 6273 ALOGV("removeEffect_l() %p effect %p", this, effect.get()); 6274 effect_descriptor_t desc = effect->desc(); 6275 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6276 detachAuxEffect_l(effect->id()); 6277 } 6278 6279 sp<EffectChain> chain = effect->chain().promote(); 6280 if (chain != 0) { 6281 // remove effect chain if removing last effect 6282 if (chain->removeEffect_l(effect) == 0) { 6283 removeEffectChain_l(chain); 6284 } 6285 } else { 6286 ALOGW("removeEffect_l() %p cannot promote chain for effect %p", this, effect.get()); 6287 } 6288} 6289 6290void AudioFlinger::ThreadBase::lockEffectChains_l( 6291 Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6292{ 6293 effectChains = mEffectChains; 6294 for (size_t i = 0; i < mEffectChains.size(); i++) { 6295 mEffectChains[i]->lock(); 6296 } 6297} 6298 6299void AudioFlinger::ThreadBase::unlockEffectChains( 6300 const Vector<sp <AudioFlinger::EffectChain> >& effectChains) 6301{ 6302 for (size_t i = 0; i < effectChains.size(); i++) { 6303 effectChains[i]->unlock(); 6304 } 6305} 6306 6307sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain(int sessionId) 6308{ 6309 Mutex::Autolock _l(mLock); 6310 return getEffectChain_l(sessionId); 6311} 6312 6313sp<AudioFlinger::EffectChain> AudioFlinger::ThreadBase::getEffectChain_l(int sessionId) 6314{ 6315 size_t size = mEffectChains.size(); 6316 for (size_t i = 0; i < size; i++) { 6317 if (mEffectChains[i]->sessionId() == sessionId) { 6318 return mEffectChains[i]; 6319 } 6320 } 6321 return 0; 6322} 6323 6324void AudioFlinger::ThreadBase::setMode(audio_mode_t mode) 6325{ 6326 Mutex::Autolock _l(mLock); 6327 size_t size = mEffectChains.size(); 6328 for (size_t i = 0; i < size; i++) { 6329 mEffectChains[i]->setMode_l(mode); 6330 } 6331} 6332 6333void AudioFlinger::ThreadBase::disconnectEffect(const sp<EffectModule>& effect, 6334 const wp<EffectHandle>& handle, 6335 bool unpinIfLast) { 6336 6337 Mutex::Autolock _l(mLock); 6338 ALOGV("disconnectEffect() %p effect %p", this, effect.get()); 6339 // delete the effect module if removing last handle on it 6340 if (effect->removeHandle(handle) == 0) { 6341 if (!effect->isPinned() || unpinIfLast) { 6342 removeEffect_l(effect); 6343 AudioSystem::unregisterEffect(effect->id()); 6344 } 6345 } 6346} 6347 6348status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain) 6349{ 6350 int session = chain->sessionId(); 6351 int16_t *buffer = mMixBuffer; 6352 bool ownsBuffer = false; 6353 6354 ALOGV("addEffectChain_l() %p on thread %p for session %d", chain.get(), this, session); 6355 if (session > 0) { 6356 // Only one effect chain can be present in direct output thread and it uses 6357 // the mix buffer as input 6358 if (mType != DIRECT) { 6359 size_t numSamples = mFrameCount * mChannelCount; 6360 buffer = new int16_t[numSamples]; 6361 memset(buffer, 0, numSamples * sizeof(int16_t)); 6362 ALOGV("addEffectChain_l() creating new input buffer %p session %d", buffer, session); 6363 ownsBuffer = true; 6364 } 6365 6366 // Attach all tracks with same session ID to this chain. 6367 for (size_t i = 0; i < mTracks.size(); ++i) { 6368 sp<Track> track = mTracks[i]; 6369 if (session == track->sessionId()) { 6370 ALOGV("addEffectChain_l() track->setMainBuffer track %p buffer %p", track.get(), buffer); 6371 track->setMainBuffer(buffer); 6372 chain->incTrackCnt(); 6373 } 6374 } 6375 6376 // indicate all active tracks in the chain 6377 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6378 sp<Track> track = mActiveTracks[i].promote(); 6379 if (track == 0) continue; 6380 if (session == track->sessionId()) { 6381 ALOGV("addEffectChain_l() activating track %p on session %d", track.get(), session); 6382 chain->incActiveTrackCnt(); 6383 } 6384 } 6385 } 6386 6387 chain->setInBuffer(buffer, ownsBuffer); 6388 chain->setOutBuffer(mMixBuffer); 6389 // Effect chain for session AUDIO_SESSION_OUTPUT_STAGE is inserted at end of effect 6390 // chains list in order to be processed last as it contains output stage effects 6391 // Effect chain for session AUDIO_SESSION_OUTPUT_MIX is inserted before 6392 // session AUDIO_SESSION_OUTPUT_STAGE to be processed 6393 // after track specific effects and before output stage 6394 // It is therefore mandatory that AUDIO_SESSION_OUTPUT_MIX == 0 and 6395 // that AUDIO_SESSION_OUTPUT_STAGE < AUDIO_SESSION_OUTPUT_MIX 6396 // Effect chain for other sessions are inserted at beginning of effect 6397 // chains list to be processed before output mix effects. Relative order between other 6398 // sessions is not important 6399 size_t size = mEffectChains.size(); 6400 size_t i = 0; 6401 for (i = 0; i < size; i++) { 6402 if (mEffectChains[i]->sessionId() < session) break; 6403 } 6404 mEffectChains.insertAt(chain, i); 6405 checkSuspendOnAddEffectChain_l(chain); 6406 6407 return NO_ERROR; 6408} 6409 6410size_t AudioFlinger::PlaybackThread::removeEffectChain_l(const sp<EffectChain>& chain) 6411{ 6412 int session = chain->sessionId(); 6413 6414 ALOGV("removeEffectChain_l() %p from thread %p for session %d", chain.get(), this, session); 6415 6416 for (size_t i = 0; i < mEffectChains.size(); i++) { 6417 if (chain == mEffectChains[i]) { 6418 mEffectChains.removeAt(i); 6419 // detach all active tracks from the chain 6420 for (size_t i = 0 ; i < mActiveTracks.size() ; ++i) { 6421 sp<Track> track = mActiveTracks[i].promote(); 6422 if (track == 0) continue; 6423 if (session == track->sessionId()) { 6424 ALOGV("removeEffectChain_l(): stopping track on chain %p for session Id: %d", 6425 chain.get(), session); 6426 chain->decActiveTrackCnt(); 6427 } 6428 } 6429 6430 // detach all tracks with same session ID from this chain 6431 for (size_t i = 0; i < mTracks.size(); ++i) { 6432 sp<Track> track = mTracks[i]; 6433 if (session == track->sessionId()) { 6434 track->setMainBuffer(mMixBuffer); 6435 chain->decTrackCnt(); 6436 } 6437 } 6438 break; 6439 } 6440 } 6441 return mEffectChains.size(); 6442} 6443 6444status_t AudioFlinger::PlaybackThread::attachAuxEffect( 6445 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6446{ 6447 Mutex::Autolock _l(mLock); 6448 return attachAuxEffect_l(track, EffectId); 6449} 6450 6451status_t AudioFlinger::PlaybackThread::attachAuxEffect_l( 6452 const sp<AudioFlinger::PlaybackThread::Track> track, int EffectId) 6453{ 6454 status_t status = NO_ERROR; 6455 6456 if (EffectId == 0) { 6457 track->setAuxBuffer(0, NULL); 6458 } else { 6459 // Auxiliary effects are always in audio session AUDIO_SESSION_OUTPUT_MIX 6460 sp<EffectModule> effect = getEffect_l(AUDIO_SESSION_OUTPUT_MIX, EffectId); 6461 if (effect != 0) { 6462 if ((effect->desc().flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6463 track->setAuxBuffer(EffectId, (int32_t *)effect->inBuffer()); 6464 } else { 6465 status = INVALID_OPERATION; 6466 } 6467 } else { 6468 status = BAD_VALUE; 6469 } 6470 } 6471 return status; 6472} 6473 6474void AudioFlinger::PlaybackThread::detachAuxEffect_l(int effectId) 6475{ 6476 for (size_t i = 0; i < mTracks.size(); ++i) { 6477 sp<Track> track = mTracks[i]; 6478 if (track->auxEffectId() == effectId) { 6479 attachAuxEffect_l(track, 0); 6480 } 6481 } 6482} 6483 6484status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain) 6485{ 6486 // only one chain per input thread 6487 if (mEffectChains.size() != 0) { 6488 return INVALID_OPERATION; 6489 } 6490 ALOGV("addEffectChain_l() %p on thread %p", chain.get(), this); 6491 6492 chain->setInBuffer(NULL); 6493 chain->setOutBuffer(NULL); 6494 6495 checkSuspendOnAddEffectChain_l(chain); 6496 6497 mEffectChains.add(chain); 6498 6499 return NO_ERROR; 6500} 6501 6502size_t AudioFlinger::RecordThread::removeEffectChain_l(const sp<EffectChain>& chain) 6503{ 6504 ALOGV("removeEffectChain_l() %p from thread %p", chain.get(), this); 6505 ALOGW_IF(mEffectChains.size() != 1, 6506 "removeEffectChain_l() %p invalid chain size %d on thread %p", 6507 chain.get(), mEffectChains.size(), this); 6508 if (mEffectChains.size() == 1) { 6509 mEffectChains.removeAt(0); 6510 } 6511 return 0; 6512} 6513 6514// ---------------------------------------------------------------------------- 6515// EffectModule implementation 6516// ---------------------------------------------------------------------------- 6517 6518#undef LOG_TAG 6519#define LOG_TAG "AudioFlinger::EffectModule" 6520 6521AudioFlinger::EffectModule::EffectModule(ThreadBase *thread, 6522 const wp<AudioFlinger::EffectChain>& chain, 6523 effect_descriptor_t *desc, 6524 int id, 6525 int sessionId) 6526 : mThread(thread), mChain(chain), mId(id), mSessionId(sessionId), mEffectInterface(NULL), 6527 mStatus(NO_INIT), mState(IDLE), mSuspended(false) 6528{ 6529 ALOGV("Constructor %p", this); 6530 int lStatus; 6531 if (thread == NULL) { 6532 return; 6533 } 6534 6535 memcpy(&mDescriptor, desc, sizeof(effect_descriptor_t)); 6536 6537 // create effect engine from effect factory 6538 mStatus = EffectCreate(&desc->uuid, sessionId, thread->id(), &mEffectInterface); 6539 6540 if (mStatus != NO_ERROR) { 6541 return; 6542 } 6543 lStatus = init(); 6544 if (lStatus < 0) { 6545 mStatus = lStatus; 6546 goto Error; 6547 } 6548 6549 if (mSessionId > AUDIO_SESSION_OUTPUT_MIX) { 6550 mPinned = true; 6551 } 6552 ALOGV("Constructor success name %s, Interface %p", mDescriptor.name, mEffectInterface); 6553 return; 6554Error: 6555 EffectRelease(mEffectInterface); 6556 mEffectInterface = NULL; 6557 ALOGV("Constructor Error %d", mStatus); 6558} 6559 6560AudioFlinger::EffectModule::~EffectModule() 6561{ 6562 ALOGV("Destructor %p", this); 6563 if (mEffectInterface != NULL) { 6564 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6565 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) { 6566 sp<ThreadBase> thread = mThread.promote(); 6567 if (thread != 0) { 6568 audio_stream_t *stream = thread->stream(); 6569 if (stream != NULL) { 6570 stream->remove_audio_effect(stream, mEffectInterface); 6571 } 6572 } 6573 } 6574 // release effect engine 6575 EffectRelease(mEffectInterface); 6576 } 6577} 6578 6579status_t AudioFlinger::EffectModule::addHandle(const sp<EffectHandle>& handle) 6580{ 6581 status_t status; 6582 6583 Mutex::Autolock _l(mLock); 6584 int priority = handle->priority(); 6585 size_t size = mHandles.size(); 6586 sp<EffectHandle> h; 6587 size_t i; 6588 for (i = 0; i < size; i++) { 6589 h = mHandles[i].promote(); 6590 if (h == 0) continue; 6591 if (h->priority() <= priority) break; 6592 } 6593 // if inserted in first place, move effect control from previous owner to this handle 6594 if (i == 0) { 6595 bool enabled = false; 6596 if (h != 0) { 6597 enabled = h->enabled(); 6598 h->setControl(false/*hasControl*/, true /*signal*/, enabled /*enabled*/); 6599 } 6600 handle->setControl(true /*hasControl*/, false /*signal*/, enabled /*enabled*/); 6601 status = NO_ERROR; 6602 } else { 6603 status = ALREADY_EXISTS; 6604 } 6605 ALOGV("addHandle() %p added handle %p in position %d", this, handle.get(), i); 6606 mHandles.insertAt(handle, i); 6607 return status; 6608} 6609 6610size_t AudioFlinger::EffectModule::removeHandle(const wp<EffectHandle>& handle) 6611{ 6612 Mutex::Autolock _l(mLock); 6613 size_t size = mHandles.size(); 6614 size_t i; 6615 for (i = 0; i < size; i++) { 6616 if (mHandles[i] == handle) break; 6617 } 6618 if (i == size) { 6619 return size; 6620 } 6621 ALOGV("removeHandle() %p removed handle %p in position %d", this, handle.unsafe_get(), i); 6622 6623 bool enabled = false; 6624 EffectHandle *hdl = handle.unsafe_get(); 6625 if (hdl != NULL) { 6626 ALOGV("removeHandle() unsafe_get OK"); 6627 enabled = hdl->enabled(); 6628 } 6629 mHandles.removeAt(i); 6630 size = mHandles.size(); 6631 // if removed from first place, move effect control from this handle to next in line 6632 if (i == 0 && size != 0) { 6633 sp<EffectHandle> h = mHandles[0].promote(); 6634 if (h != 0) { 6635 h->setControl(true /*hasControl*/, true /*signal*/ , enabled /*enabled*/); 6636 } 6637 } 6638 6639 // Prevent calls to process() and other functions on effect interface from now on. 6640 // The effect engine will be released by the destructor when the last strong reference on 6641 // this object is released which can happen after next process is called. 6642 if (size == 0 && !mPinned) { 6643 mState = DESTROYED; 6644 } 6645 6646 return size; 6647} 6648 6649sp<AudioFlinger::EffectHandle> AudioFlinger::EffectModule::controlHandle() 6650{ 6651 Mutex::Autolock _l(mLock); 6652 return mHandles.size() != 0 ? mHandles[0].promote() : 0; 6653} 6654 6655void AudioFlinger::EffectModule::disconnect(const wp<EffectHandle>& handle, bool unpinIfLast) 6656{ 6657 ALOGV("disconnect() %p handle %p", this, handle.unsafe_get()); 6658 // keep a strong reference on this EffectModule to avoid calling the 6659 // destructor before we exit 6660 sp<EffectModule> keep(this); 6661 { 6662 sp<ThreadBase> thread = mThread.promote(); 6663 if (thread != 0) { 6664 thread->disconnectEffect(keep, handle, unpinIfLast); 6665 } 6666 } 6667} 6668 6669void AudioFlinger::EffectModule::updateState() { 6670 Mutex::Autolock _l(mLock); 6671 6672 switch (mState) { 6673 case RESTART: 6674 reset_l(); 6675 // FALL THROUGH 6676 6677 case STARTING: 6678 // clear auxiliary effect input buffer for next accumulation 6679 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6680 memset(mConfig.inputCfg.buffer.raw, 6681 0, 6682 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6683 } 6684 start_l(); 6685 mState = ACTIVE; 6686 break; 6687 case STOPPING: 6688 stop_l(); 6689 mDisableWaitCnt = mMaxDisableWaitCnt; 6690 mState = STOPPED; 6691 break; 6692 case STOPPED: 6693 // mDisableWaitCnt is forced to 1 by process() when the engine indicates the end of the 6694 // turn off sequence. 6695 if (--mDisableWaitCnt == 0) { 6696 reset_l(); 6697 mState = IDLE; 6698 } 6699 break; 6700 default: //IDLE , ACTIVE, DESTROYED 6701 break; 6702 } 6703} 6704 6705void AudioFlinger::EffectModule::process() 6706{ 6707 Mutex::Autolock _l(mLock); 6708 6709 if (mState == DESTROYED || mEffectInterface == NULL || 6710 mConfig.inputCfg.buffer.raw == NULL || 6711 mConfig.outputCfg.buffer.raw == NULL) { 6712 return; 6713 } 6714 6715 if (isProcessEnabled()) { 6716 // do 32 bit to 16 bit conversion for auxiliary effect input buffer 6717 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6718 ditherAndClamp(mConfig.inputCfg.buffer.s32, 6719 mConfig.inputCfg.buffer.s32, 6720 mConfig.inputCfg.buffer.frameCount/2); 6721 } 6722 6723 // do the actual processing in the effect engine 6724 int ret = (*mEffectInterface)->process(mEffectInterface, 6725 &mConfig.inputCfg.buffer, 6726 &mConfig.outputCfg.buffer); 6727 6728 // force transition to IDLE state when engine is ready 6729 if (mState == STOPPED && ret == -ENODATA) { 6730 mDisableWaitCnt = 1; 6731 } 6732 6733 // clear auxiliary effect input buffer for next accumulation 6734 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6735 memset(mConfig.inputCfg.buffer.raw, 0, 6736 mConfig.inputCfg.buffer.frameCount*sizeof(int32_t)); 6737 } 6738 } else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT && 6739 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6740 // If an insert effect is idle and input buffer is different from output buffer, 6741 // accumulate input onto output 6742 sp<EffectChain> chain = mChain.promote(); 6743 if (chain != 0 && chain->activeTrackCnt() != 0) { 6744 size_t frameCnt = mConfig.inputCfg.buffer.frameCount * 2; //always stereo here 6745 int16_t *in = mConfig.inputCfg.buffer.s16; 6746 int16_t *out = mConfig.outputCfg.buffer.s16; 6747 for (size_t i = 0; i < frameCnt; i++) { 6748 out[i] = clamp16((int32_t)out[i] + (int32_t)in[i]); 6749 } 6750 } 6751 } 6752} 6753 6754void AudioFlinger::EffectModule::reset_l() 6755{ 6756 if (mEffectInterface == NULL) { 6757 return; 6758 } 6759 (*mEffectInterface)->command(mEffectInterface, EFFECT_CMD_RESET, 0, NULL, 0, NULL); 6760} 6761 6762status_t AudioFlinger::EffectModule::configure() 6763{ 6764 uint32_t channels; 6765 if (mEffectInterface == NULL) { 6766 return NO_INIT; 6767 } 6768 6769 sp<ThreadBase> thread = mThread.promote(); 6770 if (thread == 0) { 6771 return DEAD_OBJECT; 6772 } 6773 6774 // TODO: handle configuration of effects replacing track process 6775 if (thread->channelCount() == 1) { 6776 channels = AUDIO_CHANNEL_OUT_MONO; 6777 } else { 6778 channels = AUDIO_CHANNEL_OUT_STEREO; 6779 } 6780 6781 if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 6782 mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_MONO; 6783 } else { 6784 mConfig.inputCfg.channels = channels; 6785 } 6786 mConfig.outputCfg.channels = channels; 6787 mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6788 mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; 6789 mConfig.inputCfg.samplingRate = thread->sampleRate(); 6790 mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate; 6791 mConfig.inputCfg.bufferProvider.cookie = NULL; 6792 mConfig.inputCfg.bufferProvider.getBuffer = NULL; 6793 mConfig.inputCfg.bufferProvider.releaseBuffer = NULL; 6794 mConfig.outputCfg.bufferProvider.cookie = NULL; 6795 mConfig.outputCfg.bufferProvider.getBuffer = NULL; 6796 mConfig.outputCfg.bufferProvider.releaseBuffer = NULL; 6797 mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ; 6798 // Insert effect: 6799 // - in session AUDIO_SESSION_OUTPUT_MIX or AUDIO_SESSION_OUTPUT_STAGE, 6800 // always overwrites output buffer: input buffer == output buffer 6801 // - in other sessions: 6802 // last effect in the chain accumulates in output buffer: input buffer != output buffer 6803 // other effect: overwrites output buffer: input buffer == output buffer 6804 // Auxiliary effect: 6805 // accumulates in output buffer: input buffer != output buffer 6806 // Therefore: accumulate <=> input buffer != output buffer 6807 if (mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) { 6808 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE; 6809 } else { 6810 mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_WRITE; 6811 } 6812 mConfig.inputCfg.mask = EFFECT_CONFIG_ALL; 6813 mConfig.outputCfg.mask = EFFECT_CONFIG_ALL; 6814 mConfig.inputCfg.buffer.frameCount = thread->frameCount(); 6815 mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount; 6816 6817 ALOGV("configure() %p thread %p buffer %p framecount %d", 6818 this, thread.get(), mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount); 6819 6820 status_t cmdStatus; 6821 uint32_t size = sizeof(int); 6822 status_t status = (*mEffectInterface)->command(mEffectInterface, 6823 EFFECT_CMD_SET_CONFIG, 6824 sizeof(effect_config_t), 6825 &mConfig, 6826 &size, 6827 &cmdStatus); 6828 if (status == 0) { 6829 status = cmdStatus; 6830 } 6831 6832 mMaxDisableWaitCnt = (MAX_DISABLE_TIME_MS * mConfig.outputCfg.samplingRate) / 6833 (1000 * mConfig.outputCfg.buffer.frameCount); 6834 6835 return status; 6836} 6837 6838status_t AudioFlinger::EffectModule::init() 6839{ 6840 Mutex::Autolock _l(mLock); 6841 if (mEffectInterface == NULL) { 6842 return NO_INIT; 6843 } 6844 status_t cmdStatus; 6845 uint32_t size = sizeof(status_t); 6846 status_t status = (*mEffectInterface)->command(mEffectInterface, 6847 EFFECT_CMD_INIT, 6848 0, 6849 NULL, 6850 &size, 6851 &cmdStatus); 6852 if (status == 0) { 6853 status = cmdStatus; 6854 } 6855 return status; 6856} 6857 6858status_t AudioFlinger::EffectModule::start() 6859{ 6860 Mutex::Autolock _l(mLock); 6861 return start_l(); 6862} 6863 6864status_t AudioFlinger::EffectModule::start_l() 6865{ 6866 if (mEffectInterface == NULL) { 6867 return NO_INIT; 6868 } 6869 status_t cmdStatus; 6870 uint32_t size = sizeof(status_t); 6871 status_t status = (*mEffectInterface)->command(mEffectInterface, 6872 EFFECT_CMD_ENABLE, 6873 0, 6874 NULL, 6875 &size, 6876 &cmdStatus); 6877 if (status == 0) { 6878 status = cmdStatus; 6879 } 6880 if (status == 0 && 6881 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6882 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6883 sp<ThreadBase> thread = mThread.promote(); 6884 if (thread != 0) { 6885 audio_stream_t *stream = thread->stream(); 6886 if (stream != NULL) { 6887 stream->add_audio_effect(stream, mEffectInterface); 6888 } 6889 } 6890 } 6891 return status; 6892} 6893 6894status_t AudioFlinger::EffectModule::stop() 6895{ 6896 Mutex::Autolock _l(mLock); 6897 return stop_l(); 6898} 6899 6900status_t AudioFlinger::EffectModule::stop_l() 6901{ 6902 if (mEffectInterface == NULL) { 6903 return NO_INIT; 6904 } 6905 status_t cmdStatus; 6906 uint32_t size = sizeof(status_t); 6907 status_t status = (*mEffectInterface)->command(mEffectInterface, 6908 EFFECT_CMD_DISABLE, 6909 0, 6910 NULL, 6911 &size, 6912 &cmdStatus); 6913 if (status == 0) { 6914 status = cmdStatus; 6915 } 6916 if (status == 0 && 6917 ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC || 6918 (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC)) { 6919 sp<ThreadBase> thread = mThread.promote(); 6920 if (thread != 0) { 6921 audio_stream_t *stream = thread->stream(); 6922 if (stream != NULL) { 6923 stream->remove_audio_effect(stream, mEffectInterface); 6924 } 6925 } 6926 } 6927 return status; 6928} 6929 6930status_t AudioFlinger::EffectModule::command(uint32_t cmdCode, 6931 uint32_t cmdSize, 6932 void *pCmdData, 6933 uint32_t *replySize, 6934 void *pReplyData) 6935{ 6936 Mutex::Autolock _l(mLock); 6937// ALOGV("command(), cmdCode: %d, mEffectInterface: %p", cmdCode, mEffectInterface); 6938 6939 if (mState == DESTROYED || mEffectInterface == NULL) { 6940 return NO_INIT; 6941 } 6942 status_t status = (*mEffectInterface)->command(mEffectInterface, 6943 cmdCode, 6944 cmdSize, 6945 pCmdData, 6946 replySize, 6947 pReplyData); 6948 if (cmdCode != EFFECT_CMD_GET_PARAM && status == NO_ERROR) { 6949 uint32_t size = (replySize == NULL) ? 0 : *replySize; 6950 for (size_t i = 1; i < mHandles.size(); i++) { 6951 sp<EffectHandle> h = mHandles[i].promote(); 6952 if (h != 0) { 6953 h->commandExecuted(cmdCode, cmdSize, pCmdData, size, pReplyData); 6954 } 6955 } 6956 } 6957 return status; 6958} 6959 6960status_t AudioFlinger::EffectModule::setEnabled(bool enabled) 6961{ 6962 6963 Mutex::Autolock _l(mLock); 6964 ALOGV("setEnabled %p enabled %d", this, enabled); 6965 6966 if (enabled != isEnabled()) { 6967 status_t status = AudioSystem::setEffectEnabled(mId, enabled); 6968 if (enabled && status != NO_ERROR) { 6969 return status; 6970 } 6971 6972 switch (mState) { 6973 // going from disabled to enabled 6974 case IDLE: 6975 mState = STARTING; 6976 break; 6977 case STOPPED: 6978 mState = RESTART; 6979 break; 6980 case STOPPING: 6981 mState = ACTIVE; 6982 break; 6983 6984 // going from enabled to disabled 6985 case RESTART: 6986 mState = STOPPED; 6987 break; 6988 case STARTING: 6989 mState = IDLE; 6990 break; 6991 case ACTIVE: 6992 mState = STOPPING; 6993 break; 6994 case DESTROYED: 6995 return NO_ERROR; // simply ignore as we are being destroyed 6996 } 6997 for (size_t i = 1; i < mHandles.size(); i++) { 6998 sp<EffectHandle> h = mHandles[i].promote(); 6999 if (h != 0) { 7000 h->setEnabled(enabled); 7001 } 7002 } 7003 } 7004 return NO_ERROR; 7005} 7006 7007bool AudioFlinger::EffectModule::isEnabled() const 7008{ 7009 switch (mState) { 7010 case RESTART: 7011 case STARTING: 7012 case ACTIVE: 7013 return true; 7014 case IDLE: 7015 case STOPPING: 7016 case STOPPED: 7017 case DESTROYED: 7018 default: 7019 return false; 7020 } 7021} 7022 7023bool AudioFlinger::EffectModule::isProcessEnabled() const 7024{ 7025 switch (mState) { 7026 case RESTART: 7027 case ACTIVE: 7028 case STOPPING: 7029 case STOPPED: 7030 return true; 7031 case IDLE: 7032 case STARTING: 7033 case DESTROYED: 7034 default: 7035 return false; 7036 } 7037} 7038 7039status_t AudioFlinger::EffectModule::setVolume(uint32_t *left, uint32_t *right, bool controller) 7040{ 7041 Mutex::Autolock _l(mLock); 7042 status_t status = NO_ERROR; 7043 7044 // Send volume indication if EFFECT_FLAG_VOLUME_IND is set and read back altered volume 7045 // if controller flag is set (Note that controller == TRUE => EFFECT_FLAG_VOLUME_CTRL set) 7046 if (isProcessEnabled() && 7047 ((mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL || 7048 (mDescriptor.flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_IND)) { 7049 status_t cmdStatus; 7050 uint32_t volume[2]; 7051 uint32_t *pVolume = NULL; 7052 uint32_t size = sizeof(volume); 7053 volume[0] = *left; 7054 volume[1] = *right; 7055 if (controller) { 7056 pVolume = volume; 7057 } 7058 status = (*mEffectInterface)->command(mEffectInterface, 7059 EFFECT_CMD_SET_VOLUME, 7060 size, 7061 volume, 7062 &size, 7063 pVolume); 7064 if (controller && status == NO_ERROR && size == sizeof(volume)) { 7065 *left = volume[0]; 7066 *right = volume[1]; 7067 } 7068 } 7069 return status; 7070} 7071 7072status_t AudioFlinger::EffectModule::setDevice(uint32_t device) 7073{ 7074 Mutex::Autolock _l(mLock); 7075 status_t status = NO_ERROR; 7076 if (device && (mDescriptor.flags & EFFECT_FLAG_DEVICE_MASK) == EFFECT_FLAG_DEVICE_IND) { 7077 // audio pre processing modules on RecordThread can receive both output and 7078 // input device indication in the same call 7079 uint32_t dev = device & AUDIO_DEVICE_OUT_ALL; 7080 if (dev) { 7081 status_t cmdStatus; 7082 uint32_t size = sizeof(status_t); 7083 7084 status = (*mEffectInterface)->command(mEffectInterface, 7085 EFFECT_CMD_SET_DEVICE, 7086 sizeof(uint32_t), 7087 &dev, 7088 &size, 7089 &cmdStatus); 7090 if (status == NO_ERROR) { 7091 status = cmdStatus; 7092 } 7093 } 7094 dev = device & AUDIO_DEVICE_IN_ALL; 7095 if (dev) { 7096 status_t cmdStatus; 7097 uint32_t size = sizeof(status_t); 7098 7099 status_t status2 = (*mEffectInterface)->command(mEffectInterface, 7100 EFFECT_CMD_SET_INPUT_DEVICE, 7101 sizeof(uint32_t), 7102 &dev, 7103 &size, 7104 &cmdStatus); 7105 if (status2 == NO_ERROR) { 7106 status2 = cmdStatus; 7107 } 7108 if (status == NO_ERROR) { 7109 status = status2; 7110 } 7111 } 7112 } 7113 return status; 7114} 7115 7116status_t AudioFlinger::EffectModule::setMode(audio_mode_t mode) 7117{ 7118 Mutex::Autolock _l(mLock); 7119 status_t status = NO_ERROR; 7120 if ((mDescriptor.flags & EFFECT_FLAG_AUDIO_MODE_MASK) == EFFECT_FLAG_AUDIO_MODE_IND) { 7121 status_t cmdStatus; 7122 uint32_t size = sizeof(status_t); 7123 status = (*mEffectInterface)->command(mEffectInterface, 7124 EFFECT_CMD_SET_AUDIO_MODE, 7125 sizeof(audio_mode_t), 7126 &mode, 7127 &size, 7128 &cmdStatus); 7129 if (status == NO_ERROR) { 7130 status = cmdStatus; 7131 } 7132 } 7133 return status; 7134} 7135 7136void AudioFlinger::EffectModule::setSuspended(bool suspended) 7137{ 7138 Mutex::Autolock _l(mLock); 7139 mSuspended = suspended; 7140} 7141 7142bool AudioFlinger::EffectModule::suspended() const 7143{ 7144 Mutex::Autolock _l(mLock); 7145 return mSuspended; 7146} 7147 7148status_t AudioFlinger::EffectModule::dump(int fd, const Vector<String16>& args) 7149{ 7150 const size_t SIZE = 256; 7151 char buffer[SIZE]; 7152 String8 result; 7153 7154 snprintf(buffer, SIZE, "\tEffect ID %d:\n", mId); 7155 result.append(buffer); 7156 7157 bool locked = tryLock(mLock); 7158 // failed to lock - AudioFlinger is probably deadlocked 7159 if (!locked) { 7160 result.append("\t\tCould not lock Fx mutex:\n"); 7161 } 7162 7163 result.append("\t\tSession Status State Engine:\n"); 7164 snprintf(buffer, SIZE, "\t\t%05d %03d %03d 0x%08x\n", 7165 mSessionId, mStatus, mState, (uint32_t)mEffectInterface); 7166 result.append(buffer); 7167 7168 result.append("\t\tDescriptor:\n"); 7169 snprintf(buffer, SIZE, "\t\t- UUID: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7170 mDescriptor.uuid.timeLow, mDescriptor.uuid.timeMid, mDescriptor.uuid.timeHiAndVersion, 7171 mDescriptor.uuid.clockSeq, mDescriptor.uuid.node[0], mDescriptor.uuid.node[1],mDescriptor.uuid.node[2], 7172 mDescriptor.uuid.node[3],mDescriptor.uuid.node[4],mDescriptor.uuid.node[5]); 7173 result.append(buffer); 7174 snprintf(buffer, SIZE, "\t\t- TYPE: %08X-%04X-%04X-%04X-%02X%02X%02X%02X%02X%02X\n", 7175 mDescriptor.type.timeLow, mDescriptor.type.timeMid, mDescriptor.type.timeHiAndVersion, 7176 mDescriptor.type.clockSeq, mDescriptor.type.node[0], mDescriptor.type.node[1],mDescriptor.type.node[2], 7177 mDescriptor.type.node[3],mDescriptor.type.node[4],mDescriptor.type.node[5]); 7178 result.append(buffer); 7179 snprintf(buffer, SIZE, "\t\t- apiVersion: %08X\n\t\t- flags: %08X\n", 7180 mDescriptor.apiVersion, 7181 mDescriptor.flags); 7182 result.append(buffer); 7183 snprintf(buffer, SIZE, "\t\t- name: %s\n", 7184 mDescriptor.name); 7185 result.append(buffer); 7186 snprintf(buffer, SIZE, "\t\t- implementor: %s\n", 7187 mDescriptor.implementor); 7188 result.append(buffer); 7189 7190 result.append("\t\t- Input configuration:\n"); 7191 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7192 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7193 (uint32_t)mConfig.inputCfg.buffer.raw, 7194 mConfig.inputCfg.buffer.frameCount, 7195 mConfig.inputCfg.samplingRate, 7196 mConfig.inputCfg.channels, 7197 mConfig.inputCfg.format); 7198 result.append(buffer); 7199 7200 result.append("\t\t- Output configuration:\n"); 7201 result.append("\t\t\tBuffer Frames Smp rate Channels Format\n"); 7202 snprintf(buffer, SIZE, "\t\t\t0x%08x %05d %05d %08x %d\n", 7203 (uint32_t)mConfig.outputCfg.buffer.raw, 7204 mConfig.outputCfg.buffer.frameCount, 7205 mConfig.outputCfg.samplingRate, 7206 mConfig.outputCfg.channels, 7207 mConfig.outputCfg.format); 7208 result.append(buffer); 7209 7210 snprintf(buffer, SIZE, "\t\t%d Clients:\n", mHandles.size()); 7211 result.append(buffer); 7212 result.append("\t\t\tPid Priority Ctrl Locked client server\n"); 7213 for (size_t i = 0; i < mHandles.size(); ++i) { 7214 sp<EffectHandle> handle = mHandles[i].promote(); 7215 if (handle != 0) { 7216 handle->dump(buffer, SIZE); 7217 result.append(buffer); 7218 } 7219 } 7220 7221 result.append("\n"); 7222 7223 write(fd, result.string(), result.length()); 7224 7225 if (locked) { 7226 mLock.unlock(); 7227 } 7228 7229 return NO_ERROR; 7230} 7231 7232// ---------------------------------------------------------------------------- 7233// EffectHandle implementation 7234// ---------------------------------------------------------------------------- 7235 7236#undef LOG_TAG 7237#define LOG_TAG "AudioFlinger::EffectHandle" 7238 7239AudioFlinger::EffectHandle::EffectHandle(const sp<EffectModule>& effect, 7240 const sp<AudioFlinger::Client>& client, 7241 const sp<IEffectClient>& effectClient, 7242 int32_t priority) 7243 : BnEffect(), 7244 mEffect(effect), mEffectClient(effectClient), mClient(client), mCblk(NULL), 7245 mPriority(priority), mHasControl(false), mEnabled(false) 7246{ 7247 ALOGV("constructor %p", this); 7248 7249 if (client == 0) { 7250 return; 7251 } 7252 int bufOffset = ((sizeof(effect_param_cblk_t) - 1) / sizeof(int) + 1) * sizeof(int); 7253 mCblkMemory = client->heap()->allocate(EFFECT_PARAM_BUFFER_SIZE + bufOffset); 7254 if (mCblkMemory != 0) { 7255 mCblk = static_cast<effect_param_cblk_t *>(mCblkMemory->pointer()); 7256 7257 if (mCblk != NULL) { 7258 new(mCblk) effect_param_cblk_t(); 7259 mBuffer = (uint8_t *)mCblk + bufOffset; 7260 } 7261 } else { 7262 ALOGE("not enough memory for Effect size=%u", EFFECT_PARAM_BUFFER_SIZE + sizeof(effect_param_cblk_t)); 7263 return; 7264 } 7265} 7266 7267AudioFlinger::EffectHandle::~EffectHandle() 7268{ 7269 ALOGV("Destructor %p", this); 7270 disconnect(false); 7271 ALOGV("Destructor DONE %p", this); 7272} 7273 7274status_t AudioFlinger::EffectHandle::enable() 7275{ 7276 ALOGV("enable %p", this); 7277 if (!mHasControl) return INVALID_OPERATION; 7278 if (mEffect == 0) return DEAD_OBJECT; 7279 7280 if (mEnabled) { 7281 return NO_ERROR; 7282 } 7283 7284 mEnabled = true; 7285 7286 sp<ThreadBase> thread = mEffect->thread().promote(); 7287 if (thread != 0) { 7288 thread->checkSuspendOnEffectEnabled(mEffect, true, mEffect->sessionId()); 7289 } 7290 7291 // checkSuspendOnEffectEnabled() can suspend this same effect when enabled 7292 if (mEffect->suspended()) { 7293 return NO_ERROR; 7294 } 7295 7296 status_t status = mEffect->setEnabled(true); 7297 if (status != NO_ERROR) { 7298 if (thread != 0) { 7299 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7300 } 7301 mEnabled = false; 7302 } 7303 return status; 7304} 7305 7306status_t AudioFlinger::EffectHandle::disable() 7307{ 7308 ALOGV("disable %p", this); 7309 if (!mHasControl) return INVALID_OPERATION; 7310 if (mEffect == 0) return DEAD_OBJECT; 7311 7312 if (!mEnabled) { 7313 return NO_ERROR; 7314 } 7315 mEnabled = false; 7316 7317 if (mEffect->suspended()) { 7318 return NO_ERROR; 7319 } 7320 7321 status_t status = mEffect->setEnabled(false); 7322 7323 sp<ThreadBase> thread = mEffect->thread().promote(); 7324 if (thread != 0) { 7325 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7326 } 7327 7328 return status; 7329} 7330 7331void AudioFlinger::EffectHandle::disconnect() 7332{ 7333 disconnect(true); 7334} 7335 7336void AudioFlinger::EffectHandle::disconnect(bool unpinIfLast) 7337{ 7338 ALOGV("disconnect(%s)", unpinIfLast ? "true" : "false"); 7339 if (mEffect == 0) { 7340 return; 7341 } 7342 mEffect->disconnect(this, unpinIfLast); 7343 7344 if (mHasControl && mEnabled) { 7345 sp<ThreadBase> thread = mEffect->thread().promote(); 7346 if (thread != 0) { 7347 thread->checkSuspendOnEffectEnabled(mEffect, false, mEffect->sessionId()); 7348 } 7349 } 7350 7351 // release sp on module => module destructor can be called now 7352 mEffect.clear(); 7353 if (mClient != 0) { 7354 if (mCblk != NULL) { 7355 // unlike ~TrackBase(), mCblk is never a local new, so don't delete 7356 mCblk->~effect_param_cblk_t(); // destroy our shared-structure. 7357 } 7358 mCblkMemory.clear(); // free the shared memory before releasing the heap it belongs to 7359 // Client destructor must run with AudioFlinger mutex locked 7360 Mutex::Autolock _l(mClient->audioFlinger()->mLock); 7361 mClient.clear(); 7362 } 7363} 7364 7365status_t AudioFlinger::EffectHandle::command(uint32_t cmdCode, 7366 uint32_t cmdSize, 7367 void *pCmdData, 7368 uint32_t *replySize, 7369 void *pReplyData) 7370{ 7371// ALOGV("command(), cmdCode: %d, mHasControl: %d, mEffect: %p", 7372// cmdCode, mHasControl, (mEffect == 0) ? 0 : mEffect.get()); 7373 7374 // only get parameter command is permitted for applications not controlling the effect 7375 if (!mHasControl && cmdCode != EFFECT_CMD_GET_PARAM) { 7376 return INVALID_OPERATION; 7377 } 7378 if (mEffect == 0) return DEAD_OBJECT; 7379 if (mClient == 0) return INVALID_OPERATION; 7380 7381 // handle commands that are not forwarded transparently to effect engine 7382 if (cmdCode == EFFECT_CMD_SET_PARAM_COMMIT) { 7383 // No need to trylock() here as this function is executed in the binder thread serving a particular client process: 7384 // no risk to block the whole media server process or mixer threads is we are stuck here 7385 Mutex::Autolock _l(mCblk->lock); 7386 if (mCblk->clientIndex > EFFECT_PARAM_BUFFER_SIZE || 7387 mCblk->serverIndex > EFFECT_PARAM_BUFFER_SIZE) { 7388 mCblk->serverIndex = 0; 7389 mCblk->clientIndex = 0; 7390 return BAD_VALUE; 7391 } 7392 status_t status = NO_ERROR; 7393 while (mCblk->serverIndex < mCblk->clientIndex) { 7394 int reply; 7395 uint32_t rsize = sizeof(int); 7396 int *p = (int *)(mBuffer + mCblk->serverIndex); 7397 int size = *p++; 7398 if (((uint8_t *)p + size) > mBuffer + mCblk->clientIndex) { 7399 ALOGW("command(): invalid parameter block size"); 7400 break; 7401 } 7402 effect_param_t *param = (effect_param_t *)p; 7403 if (param->psize == 0 || param->vsize == 0) { 7404 ALOGW("command(): null parameter or value size"); 7405 mCblk->serverIndex += size; 7406 continue; 7407 } 7408 uint32_t psize = sizeof(effect_param_t) + 7409 ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + 7410 param->vsize; 7411 status_t ret = mEffect->command(EFFECT_CMD_SET_PARAM, 7412 psize, 7413 p, 7414 &rsize, 7415 &reply); 7416 // stop at first error encountered 7417 if (ret != NO_ERROR) { 7418 status = ret; 7419 *(int *)pReplyData = reply; 7420 break; 7421 } else if (reply != NO_ERROR) { 7422 *(int *)pReplyData = reply; 7423 break; 7424 } 7425 mCblk->serverIndex += size; 7426 } 7427 mCblk->serverIndex = 0; 7428 mCblk->clientIndex = 0; 7429 return status; 7430 } else if (cmdCode == EFFECT_CMD_ENABLE) { 7431 *(int *)pReplyData = NO_ERROR; 7432 return enable(); 7433 } else if (cmdCode == EFFECT_CMD_DISABLE) { 7434 *(int *)pReplyData = NO_ERROR; 7435 return disable(); 7436 } 7437 7438 return mEffect->command(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7439} 7440 7441void AudioFlinger::EffectHandle::setControl(bool hasControl, bool signal, bool enabled) 7442{ 7443 ALOGV("setControl %p control %d", this, hasControl); 7444 7445 mHasControl = hasControl; 7446 mEnabled = enabled; 7447 7448 if (signal && mEffectClient != 0) { 7449 mEffectClient->controlStatusChanged(hasControl); 7450 } 7451} 7452 7453void AudioFlinger::EffectHandle::commandExecuted(uint32_t cmdCode, 7454 uint32_t cmdSize, 7455 void *pCmdData, 7456 uint32_t replySize, 7457 void *pReplyData) 7458{ 7459 if (mEffectClient != 0) { 7460 mEffectClient->commandExecuted(cmdCode, cmdSize, pCmdData, replySize, pReplyData); 7461 } 7462} 7463 7464 7465 7466void AudioFlinger::EffectHandle::setEnabled(bool enabled) 7467{ 7468 if (mEffectClient != 0) { 7469 mEffectClient->enableStatusChanged(enabled); 7470 } 7471} 7472 7473status_t AudioFlinger::EffectHandle::onTransact( 7474 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 7475{ 7476 return BnEffect::onTransact(code, data, reply, flags); 7477} 7478 7479 7480void AudioFlinger::EffectHandle::dump(char* buffer, size_t size) 7481{ 7482 bool locked = mCblk != NULL && tryLock(mCblk->lock); 7483 7484 snprintf(buffer, size, "\t\t\t%05d %05d %01u %01u %05u %05u\n", 7485 (mClient == 0) ? getpid_cached : mClient->pid(), 7486 mPriority, 7487 mHasControl, 7488 !locked, 7489 mCblk ? mCblk->clientIndex : 0, 7490 mCblk ? mCblk->serverIndex : 0 7491 ); 7492 7493 if (locked) { 7494 mCblk->lock.unlock(); 7495 } 7496} 7497 7498#undef LOG_TAG 7499#define LOG_TAG "AudioFlinger::EffectChain" 7500 7501AudioFlinger::EffectChain::EffectChain(ThreadBase *thread, 7502 int sessionId) 7503 : mThread(thread), mSessionId(sessionId), mActiveTrackCnt(0), mTrackCnt(0), mTailBufferCount(0), 7504 mOwnInBuffer(false), mVolumeCtrlIdx(-1), mLeftVolume(UINT_MAX), mRightVolume(UINT_MAX), 7505 mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX) 7506{ 7507 mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC); 7508 if (thread == NULL) { 7509 return; 7510 } 7511 mMaxTailBuffers = ((kProcessTailDurationMs * thread->sampleRate()) / 1000) / 7512 thread->frameCount(); 7513} 7514 7515AudioFlinger::EffectChain::~EffectChain() 7516{ 7517 if (mOwnInBuffer) { 7518 delete mInBuffer; 7519 } 7520 7521} 7522 7523// getEffectFromDesc_l() must be called with ThreadBase::mLock held 7524sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromDesc_l(effect_descriptor_t *descriptor) 7525{ 7526 size_t size = mEffects.size(); 7527 7528 for (size_t i = 0; i < size; i++) { 7529 if (memcmp(&mEffects[i]->desc().uuid, &descriptor->uuid, sizeof(effect_uuid_t)) == 0) { 7530 return mEffects[i]; 7531 } 7532 } 7533 return 0; 7534} 7535 7536// getEffectFromId_l() must be called with ThreadBase::mLock held 7537sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromId_l(int id) 7538{ 7539 size_t size = mEffects.size(); 7540 7541 for (size_t i = 0; i < size; i++) { 7542 // by convention, return first effect if id provided is 0 (0 is never a valid id) 7543 if (id == 0 || mEffects[i]->id() == id) { 7544 return mEffects[i]; 7545 } 7546 } 7547 return 0; 7548} 7549 7550// getEffectFromType_l() must be called with ThreadBase::mLock held 7551sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectFromType_l( 7552 const effect_uuid_t *type) 7553{ 7554 size_t size = mEffects.size(); 7555 7556 for (size_t i = 0; i < size; i++) { 7557 if (memcmp(&mEffects[i]->desc().type, type, sizeof(effect_uuid_t)) == 0) { 7558 return mEffects[i]; 7559 } 7560 } 7561 return 0; 7562} 7563 7564// Must be called with EffectChain::mLock locked 7565void AudioFlinger::EffectChain::process_l() 7566{ 7567 sp<ThreadBase> thread = mThread.promote(); 7568 if (thread == 0) { 7569 ALOGW("process_l(): cannot promote mixer thread"); 7570 return; 7571 } 7572 bool isGlobalSession = (mSessionId == AUDIO_SESSION_OUTPUT_MIX) || 7573 (mSessionId == AUDIO_SESSION_OUTPUT_STAGE); 7574 // always process effects unless no more tracks are on the session and the effect tail 7575 // has been rendered 7576 bool doProcess = true; 7577 if (!isGlobalSession) { 7578 bool tracksOnSession = (trackCnt() != 0); 7579 7580 if (!tracksOnSession && mTailBufferCount == 0) { 7581 doProcess = false; 7582 } 7583 7584 if (activeTrackCnt() == 0) { 7585 // if no track is active and the effect tail has not been rendered, 7586 // the input buffer must be cleared here as the mixer process will not do it 7587 if (tracksOnSession || mTailBufferCount > 0) { 7588 size_t numSamples = thread->frameCount() * thread->channelCount(); 7589 memset(mInBuffer, 0, numSamples * sizeof(int16_t)); 7590 if (mTailBufferCount > 0) { 7591 mTailBufferCount--; 7592 } 7593 } 7594 } 7595 } 7596 7597 size_t size = mEffects.size(); 7598 if (doProcess) { 7599 for (size_t i = 0; i < size; i++) { 7600 mEffects[i]->process(); 7601 } 7602 } 7603 for (size_t i = 0; i < size; i++) { 7604 mEffects[i]->updateState(); 7605 } 7606} 7607 7608// addEffect_l() must be called with PlaybackThread::mLock held 7609status_t AudioFlinger::EffectChain::addEffect_l(const sp<EffectModule>& effect) 7610{ 7611 effect_descriptor_t desc = effect->desc(); 7612 uint32_t insertPref = desc.flags & EFFECT_FLAG_INSERT_MASK; 7613 7614 Mutex::Autolock _l(mLock); 7615 effect->setChain(this); 7616 sp<ThreadBase> thread = mThread.promote(); 7617 if (thread == 0) { 7618 return NO_INIT; 7619 } 7620 effect->setThread(thread); 7621 7622 if ((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 7623 // Auxiliary effects are inserted at the beginning of mEffects vector as 7624 // they are processed first and accumulated in chain input buffer 7625 mEffects.insertAt(effect, 0); 7626 7627 // the input buffer for auxiliary effect contains mono samples in 7628 // 32 bit format. This is to avoid saturation in AudoMixer 7629 // accumulation stage. Saturation is done in EffectModule::process() before 7630 // calling the process in effect engine 7631 size_t numSamples = thread->frameCount(); 7632 int32_t *buffer = new int32_t[numSamples]; 7633 memset(buffer, 0, numSamples * sizeof(int32_t)); 7634 effect->setInBuffer((int16_t *)buffer); 7635 // auxiliary effects output samples to chain input buffer for further processing 7636 // by insert effects 7637 effect->setOutBuffer(mInBuffer); 7638 } else { 7639 // Insert effects are inserted at the end of mEffects vector as they are processed 7640 // after track and auxiliary effects. 7641 // Insert effect order as a function of indicated preference: 7642 // if EFFECT_FLAG_INSERT_EXCLUSIVE, insert in first position or reject if 7643 // another effect is present 7644 // else if EFFECT_FLAG_INSERT_FIRST, insert in first position or after the 7645 // last effect claiming first position 7646 // else if EFFECT_FLAG_INSERT_LAST, insert in last position or before the 7647 // first effect claiming last position 7648 // else if EFFECT_FLAG_INSERT_ANY insert after first or before last 7649 // Reject insertion if an effect with EFFECT_FLAG_INSERT_EXCLUSIVE is 7650 // already present 7651 7652 size_t size = mEffects.size(); 7653 size_t idx_insert = size; 7654 ssize_t idx_insert_first = -1; 7655 ssize_t idx_insert_last = -1; 7656 7657 for (size_t i = 0; i < size; i++) { 7658 effect_descriptor_t d = mEffects[i]->desc(); 7659 uint32_t iMode = d.flags & EFFECT_FLAG_TYPE_MASK; 7660 uint32_t iPref = d.flags & EFFECT_FLAG_INSERT_MASK; 7661 if (iMode == EFFECT_FLAG_TYPE_INSERT) { 7662 // check invalid effect chaining combinations 7663 if (insertPref == EFFECT_FLAG_INSERT_EXCLUSIVE || 7664 iPref == EFFECT_FLAG_INSERT_EXCLUSIVE) { 7665 ALOGW("addEffect_l() could not insert effect %s: exclusive conflict with %s", desc.name, d.name); 7666 return INVALID_OPERATION; 7667 } 7668 // remember position of first insert effect and by default 7669 // select this as insert position for new effect 7670 if (idx_insert == size) { 7671 idx_insert = i; 7672 } 7673 // remember position of last insert effect claiming 7674 // first position 7675 if (iPref == EFFECT_FLAG_INSERT_FIRST) { 7676 idx_insert_first = i; 7677 } 7678 // remember position of first insert effect claiming 7679 // last position 7680 if (iPref == EFFECT_FLAG_INSERT_LAST && 7681 idx_insert_last == -1) { 7682 idx_insert_last = i; 7683 } 7684 } 7685 } 7686 7687 // modify idx_insert from first position if needed 7688 if (insertPref == EFFECT_FLAG_INSERT_LAST) { 7689 if (idx_insert_last != -1) { 7690 idx_insert = idx_insert_last; 7691 } else { 7692 idx_insert = size; 7693 } 7694 } else { 7695 if (idx_insert_first != -1) { 7696 idx_insert = idx_insert_first + 1; 7697 } 7698 } 7699 7700 // always read samples from chain input buffer 7701 effect->setInBuffer(mInBuffer); 7702 7703 // if last effect in the chain, output samples to chain 7704 // output buffer, otherwise to chain input buffer 7705 if (idx_insert == size) { 7706 if (idx_insert != 0) { 7707 mEffects[idx_insert-1]->setOutBuffer(mInBuffer); 7708 mEffects[idx_insert-1]->configure(); 7709 } 7710 effect->setOutBuffer(mOutBuffer); 7711 } else { 7712 effect->setOutBuffer(mInBuffer); 7713 } 7714 mEffects.insertAt(effect, idx_insert); 7715 7716 ALOGV("addEffect_l() effect %p, added in chain %p at rank %d", effect.get(), this, idx_insert); 7717 } 7718 effect->configure(); 7719 return NO_ERROR; 7720} 7721 7722// removeEffect_l() must be called with PlaybackThread::mLock held 7723size_t AudioFlinger::EffectChain::removeEffect_l(const sp<EffectModule>& effect) 7724{ 7725 Mutex::Autolock _l(mLock); 7726 size_t size = mEffects.size(); 7727 uint32_t type = effect->desc().flags & EFFECT_FLAG_TYPE_MASK; 7728 7729 for (size_t i = 0; i < size; i++) { 7730 if (effect == mEffects[i]) { 7731 // calling stop here will remove pre-processing effect from the audio HAL. 7732 // This is safe as we hold the EffectChain mutex which guarantees that we are not in 7733 // the middle of a read from audio HAL 7734 if (mEffects[i]->state() == EffectModule::ACTIVE || 7735 mEffects[i]->state() == EffectModule::STOPPING) { 7736 mEffects[i]->stop(); 7737 } 7738 if (type == EFFECT_FLAG_TYPE_AUXILIARY) { 7739 delete[] effect->inBuffer(); 7740 } else { 7741 if (i == size - 1 && i != 0) { 7742 mEffects[i - 1]->setOutBuffer(mOutBuffer); 7743 mEffects[i - 1]->configure(); 7744 } 7745 } 7746 mEffects.removeAt(i); 7747 ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %d", effect.get(), this, i); 7748 break; 7749 } 7750 } 7751 7752 return mEffects.size(); 7753} 7754 7755// setDevice_l() must be called with PlaybackThread::mLock held 7756void AudioFlinger::EffectChain::setDevice_l(uint32_t device) 7757{ 7758 size_t size = mEffects.size(); 7759 for (size_t i = 0; i < size; i++) { 7760 mEffects[i]->setDevice(device); 7761 } 7762} 7763 7764// setMode_l() must be called with PlaybackThread::mLock held 7765void AudioFlinger::EffectChain::setMode_l(audio_mode_t mode) 7766{ 7767 size_t size = mEffects.size(); 7768 for (size_t i = 0; i < size; i++) { 7769 mEffects[i]->setMode(mode); 7770 } 7771} 7772 7773// setVolume_l() must be called with PlaybackThread::mLock held 7774bool AudioFlinger::EffectChain::setVolume_l(uint32_t *left, uint32_t *right) 7775{ 7776 uint32_t newLeft = *left; 7777 uint32_t newRight = *right; 7778 bool hasControl = false; 7779 int ctrlIdx = -1; 7780 size_t size = mEffects.size(); 7781 7782 // first update volume controller 7783 for (size_t i = size; i > 0; i--) { 7784 if (mEffects[i - 1]->isProcessEnabled() && 7785 (mEffects[i - 1]->desc().flags & EFFECT_FLAG_VOLUME_MASK) == EFFECT_FLAG_VOLUME_CTRL) { 7786 ctrlIdx = i - 1; 7787 hasControl = true; 7788 break; 7789 } 7790 } 7791 7792 if (ctrlIdx == mVolumeCtrlIdx && *left == mLeftVolume && *right == mRightVolume) { 7793 if (hasControl) { 7794 *left = mNewLeftVolume; 7795 *right = mNewRightVolume; 7796 } 7797 return hasControl; 7798 } 7799 7800 mVolumeCtrlIdx = ctrlIdx; 7801 mLeftVolume = newLeft; 7802 mRightVolume = newRight; 7803 7804 // second get volume update from volume controller 7805 if (ctrlIdx >= 0) { 7806 mEffects[ctrlIdx]->setVolume(&newLeft, &newRight, true); 7807 mNewLeftVolume = newLeft; 7808 mNewRightVolume = newRight; 7809 } 7810 // then indicate volume to all other effects in chain. 7811 // Pass altered volume to effects before volume controller 7812 // and requested volume to effects after controller 7813 uint32_t lVol = newLeft; 7814 uint32_t rVol = newRight; 7815 7816 for (size_t i = 0; i < size; i++) { 7817 if ((int)i == ctrlIdx) continue; 7818 // this also works for ctrlIdx == -1 when there is no volume controller 7819 if ((int)i > ctrlIdx) { 7820 lVol = *left; 7821 rVol = *right; 7822 } 7823 mEffects[i]->setVolume(&lVol, &rVol, false); 7824 } 7825 *left = newLeft; 7826 *right = newRight; 7827 7828 return hasControl; 7829} 7830 7831status_t AudioFlinger::EffectChain::dump(int fd, const Vector<String16>& args) 7832{ 7833 const size_t SIZE = 256; 7834 char buffer[SIZE]; 7835 String8 result; 7836 7837 snprintf(buffer, SIZE, "Effects for session %d:\n", mSessionId); 7838 result.append(buffer); 7839 7840 bool locked = tryLock(mLock); 7841 // failed to lock - AudioFlinger is probably deadlocked 7842 if (!locked) { 7843 result.append("\tCould not lock mutex:\n"); 7844 } 7845 7846 result.append("\tNum fx In buffer Out buffer Active tracks:\n"); 7847 snprintf(buffer, SIZE, "\t%02d 0x%08x 0x%08x %d\n", 7848 mEffects.size(), 7849 (uint32_t)mInBuffer, 7850 (uint32_t)mOutBuffer, 7851 mActiveTrackCnt); 7852 result.append(buffer); 7853 write(fd, result.string(), result.size()); 7854 7855 for (size_t i = 0; i < mEffects.size(); ++i) { 7856 sp<EffectModule> effect = mEffects[i]; 7857 if (effect != 0) { 7858 effect->dump(fd, args); 7859 } 7860 } 7861 7862 if (locked) { 7863 mLock.unlock(); 7864 } 7865 7866 return NO_ERROR; 7867} 7868 7869// must be called with ThreadBase::mLock held 7870void AudioFlinger::EffectChain::setEffectSuspended_l( 7871 const effect_uuid_t *type, bool suspend) 7872{ 7873 sp<SuspendedEffectDesc> desc; 7874 // use effect type UUID timelow as key as there is no real risk of identical 7875 // timeLow fields among effect type UUIDs. 7876 ssize_t index = mSuspendedEffects.indexOfKey(type->timeLow); 7877 if (suspend) { 7878 if (index >= 0) { 7879 desc = mSuspendedEffects.valueAt(index); 7880 } else { 7881 desc = new SuspendedEffectDesc(); 7882 memcpy(&desc->mType, type, sizeof(effect_uuid_t)); 7883 mSuspendedEffects.add(type->timeLow, desc); 7884 ALOGV("setEffectSuspended_l() add entry for %08x", type->timeLow); 7885 } 7886 if (desc->mRefCount++ == 0) { 7887 sp<EffectModule> effect = getEffectIfEnabled(type); 7888 if (effect != 0) { 7889 desc->mEffect = effect; 7890 effect->setSuspended(true); 7891 effect->setEnabled(false); 7892 } 7893 } 7894 } else { 7895 if (index < 0) { 7896 return; 7897 } 7898 desc = mSuspendedEffects.valueAt(index); 7899 if (desc->mRefCount <= 0) { 7900 ALOGW("setEffectSuspended_l() restore refcount should not be 0 %d", desc->mRefCount); 7901 desc->mRefCount = 1; 7902 } 7903 if (--desc->mRefCount == 0) { 7904 ALOGV("setEffectSuspended_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7905 if (desc->mEffect != 0) { 7906 sp<EffectModule> effect = desc->mEffect.promote(); 7907 if (effect != 0) { 7908 effect->setSuspended(false); 7909 sp<EffectHandle> handle = effect->controlHandle(); 7910 if (handle != 0) { 7911 effect->setEnabled(handle->enabled()); 7912 } 7913 } 7914 desc->mEffect.clear(); 7915 } 7916 mSuspendedEffects.removeItemsAt(index); 7917 } 7918 } 7919} 7920 7921// must be called with ThreadBase::mLock held 7922void AudioFlinger::EffectChain::setEffectSuspendedAll_l(bool suspend) 7923{ 7924 sp<SuspendedEffectDesc> desc; 7925 7926 ssize_t index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 7927 if (suspend) { 7928 if (index >= 0) { 7929 desc = mSuspendedEffects.valueAt(index); 7930 } else { 7931 desc = new SuspendedEffectDesc(); 7932 mSuspendedEffects.add((int)kKeyForSuspendAll, desc); 7933 ALOGV("setEffectSuspendedAll_l() add entry for 0"); 7934 } 7935 if (desc->mRefCount++ == 0) { 7936 Vector< sp<EffectModule> > effects; 7937 getSuspendEligibleEffects(effects); 7938 for (size_t i = 0; i < effects.size(); i++) { 7939 setEffectSuspended_l(&effects[i]->desc().type, true); 7940 } 7941 } 7942 } else { 7943 if (index < 0) { 7944 return; 7945 } 7946 desc = mSuspendedEffects.valueAt(index); 7947 if (desc->mRefCount <= 0) { 7948 ALOGW("setEffectSuspendedAll_l() restore refcount should not be 0 %d", desc->mRefCount); 7949 desc->mRefCount = 1; 7950 } 7951 if (--desc->mRefCount == 0) { 7952 Vector<const effect_uuid_t *> types; 7953 for (size_t i = 0; i < mSuspendedEffects.size(); i++) { 7954 if (mSuspendedEffects.keyAt(i) == (int)kKeyForSuspendAll) { 7955 continue; 7956 } 7957 types.add(&mSuspendedEffects.valueAt(i)->mType); 7958 } 7959 for (size_t i = 0; i < types.size(); i++) { 7960 setEffectSuspended_l(types[i], false); 7961 } 7962 ALOGV("setEffectSuspendedAll_l() remove entry for %08x", mSuspendedEffects.keyAt(index)); 7963 mSuspendedEffects.removeItem((int)kKeyForSuspendAll); 7964 } 7965 } 7966} 7967 7968 7969// The volume effect is used for automated tests only 7970#ifndef OPENSL_ES_H_ 7971static const effect_uuid_t SL_IID_VOLUME_ = { 0x09e8ede0, 0xddde, 0x11db, 0xb4f6, 7972 { 0x00, 0x02, 0xa5, 0xd5, 0xc5, 0x1b } }; 7973const effect_uuid_t * const SL_IID_VOLUME = &SL_IID_VOLUME_; 7974#endif //OPENSL_ES_H_ 7975 7976bool AudioFlinger::EffectChain::isEffectEligibleForSuspend(const effect_descriptor_t& desc) 7977{ 7978 // auxiliary effects and visualizer are never suspended on output mix 7979 if ((mSessionId == AUDIO_SESSION_OUTPUT_MIX) && 7980 (((desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) || 7981 (memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) || 7982 (memcmp(&desc.type, SL_IID_VOLUME, sizeof(effect_uuid_t)) == 0))) { 7983 return false; 7984 } 7985 return true; 7986} 7987 7988void AudioFlinger::EffectChain::getSuspendEligibleEffects(Vector< sp<AudioFlinger::EffectModule> > &effects) 7989{ 7990 effects.clear(); 7991 for (size_t i = 0; i < mEffects.size(); i++) { 7992 if (isEffectEligibleForSuspend(mEffects[i]->desc())) { 7993 effects.add(mEffects[i]); 7994 } 7995 } 7996} 7997 7998sp<AudioFlinger::EffectModule> AudioFlinger::EffectChain::getEffectIfEnabled( 7999 const effect_uuid_t *type) 8000{ 8001 sp<EffectModule> effect = getEffectFromType_l(type); 8002 return effect != 0 && effect->isEnabled() ? effect : 0; 8003} 8004 8005void AudioFlinger::EffectChain::checkSuspendOnEffectEnabled(const sp<EffectModule>& effect, 8006 bool enabled) 8007{ 8008 ssize_t index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8009 if (enabled) { 8010 if (index < 0) { 8011 // if the effect is not suspend check if all effects are suspended 8012 index = mSuspendedEffects.indexOfKey((int)kKeyForSuspendAll); 8013 if (index < 0) { 8014 return; 8015 } 8016 if (!isEffectEligibleForSuspend(effect->desc())) { 8017 return; 8018 } 8019 setEffectSuspended_l(&effect->desc().type, enabled); 8020 index = mSuspendedEffects.indexOfKey(effect->desc().type.timeLow); 8021 if (index < 0) { 8022 ALOGW("checkSuspendOnEffectEnabled() Fx should be suspended here!"); 8023 return; 8024 } 8025 } 8026 ALOGV("checkSuspendOnEffectEnabled() enable suspending fx %08x", 8027 effect->desc().type.timeLow); 8028 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8029 // if effect is requested to suspended but was not yet enabled, supend it now. 8030 if (desc->mEffect == 0) { 8031 desc->mEffect = effect; 8032 effect->setEnabled(false); 8033 effect->setSuspended(true); 8034 } 8035 } else { 8036 if (index < 0) { 8037 return; 8038 } 8039 ALOGV("checkSuspendOnEffectEnabled() disable restoring fx %08x", 8040 effect->desc().type.timeLow); 8041 sp<SuspendedEffectDesc> desc = mSuspendedEffects.valueAt(index); 8042 desc->mEffect.clear(); 8043 effect->setSuspended(false); 8044 } 8045} 8046 8047#undef LOG_TAG 8048#define LOG_TAG "AudioFlinger" 8049 8050// ---------------------------------------------------------------------------- 8051 8052status_t AudioFlinger::onTransact( 8053 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 8054{ 8055 return BnAudioFlinger::onTransact(code, data, reply, flags); 8056} 8057 8058}; // namespace android 8059