AudioFlinger.cpp revision cc85abcf4ac398dca240db356b8b4db052b415a4
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 "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/AudioResamplerPublic.h> 49 50#include <media/EffectsFactoryApi.h> 51#include <audio_effects/effect_visualizer.h> 52#include <audio_effects/effect_ns.h> 53#include <audio_effects/effect_aec.h> 54 55#include <audio_utils/primitives.h> 56 57#include <powermanager/PowerManager.h> 58 59#include <common_time/cc_helper.h> 60 61#include <media/IMediaLogService.h> 62 63#include <media/nbaio/Pipe.h> 64#include <media/nbaio/PipeReader.h> 65#include <media/AudioParameter.h> 66#include <private/android_filesystem_config.h> 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92uint32_t AudioFlinger::mScreenState; 93 94#ifdef TEE_SINK 95bool AudioFlinger::mTeeSinkInputEnabled = false; 96bool AudioFlinger::mTeeSinkOutputEnabled = false; 97bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 102#endif 103 104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 105// we define a minimum time during which a global effect is considered enabled. 106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108// ---------------------------------------------------------------------------- 109 110const char *formatToString(audio_format_t format) { 111 switch (format & AUDIO_FORMAT_MAIN_MASK) { 112 case AUDIO_FORMAT_PCM: 113 switch (format) { 114 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 115 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 116 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 117 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 118 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 119 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 120 default: 121 break; 122 } 123 break; 124 case AUDIO_FORMAT_MP3: return "mp3"; 125 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 126 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 127 case AUDIO_FORMAT_AAC: return "aac"; 128 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 129 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 130 case AUDIO_FORMAT_VORBIS: return "vorbis"; 131 case AUDIO_FORMAT_OPUS: return "opus"; 132 case AUDIO_FORMAT_AC3: return "ac-3"; 133 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 134 default: 135 break; 136 } 137 return "unknown"; 138} 139 140static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 141{ 142 const hw_module_t *mod; 143 int rc; 144 145 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 146 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 147 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 148 if (rc) { 149 goto out; 150 } 151 rc = audio_hw_device_open(mod, dev); 152 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 153 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 154 if (rc) { 155 goto out; 156 } 157 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 158 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 159 rc = BAD_VALUE; 160 goto out; 161 } 162 return 0; 163 164out: 165 *dev = NULL; 166 return rc; 167} 168 169// ---------------------------------------------------------------------------- 170 171AudioFlinger::AudioFlinger() 172 : BnAudioFlinger(), 173 mPrimaryHardwareDev(NULL), 174 mAudioHwDevs(NULL), 175 mHardwareStatus(AUDIO_HW_IDLE), 176 mMasterVolume(1.0f), 177 mMasterMute(false), 178 mNextUniqueId(1), 179 mMode(AUDIO_MODE_INVALID), 180 mBtNrecIsOff(false), 181 mIsLowRamDevice(true), 182 mIsDeviceTypeKnown(false), 183 mGlobalEffectEnableTime(0), 184 mPrimaryOutputSampleRate(0) 185{ 186 getpid_cached = getpid(); 187 char value[PROPERTY_VALUE_MAX]; 188 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 189 if (doLog) { 190 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 191 MemoryHeapBase::READ_ONLY); 192 } 193 194#ifdef TEE_SINK 195 (void) property_get("ro.debuggable", value, "0"); 196 int debuggable = atoi(value); 197 int teeEnabled = 0; 198 if (debuggable) { 199 (void) property_get("af.tee", value, "0"); 200 teeEnabled = atoi(value); 201 } 202 // FIXME symbolic constants here 203 if (teeEnabled & 1) { 204 mTeeSinkInputEnabled = true; 205 } 206 if (teeEnabled & 2) { 207 mTeeSinkOutputEnabled = true; 208 } 209 if (teeEnabled & 4) { 210 mTeeSinkTrackEnabled = true; 211 } 212#endif 213} 214 215void AudioFlinger::onFirstRef() 216{ 217 int rc = 0; 218 219 Mutex::Autolock _l(mLock); 220 221 /* TODO: move all this work into an Init() function */ 222 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 223 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 224 uint32_t int_val; 225 if (1 == sscanf(val_str, "%u", &int_val)) { 226 mStandbyTimeInNsecs = milliseconds(int_val); 227 ALOGI("Using %u mSec as standby time.", int_val); 228 } else { 229 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 230 ALOGI("Using default %u mSec as standby time.", 231 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 232 } 233 } 234 235 mPatchPanel = new PatchPanel(this); 236 237 mMode = AUDIO_MODE_NORMAL; 238} 239 240AudioFlinger::~AudioFlinger() 241{ 242 while (!mRecordThreads.isEmpty()) { 243 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 244 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 245 } 246 while (!mPlaybackThreads.isEmpty()) { 247 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 248 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 249 } 250 251 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 252 // no mHardwareLock needed, as there are no other references to this 253 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 254 delete mAudioHwDevs.valueAt(i); 255 } 256 257 // Tell media.log service about any old writers that still need to be unregistered 258 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 259 if (binder != 0) { 260 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 261 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 262 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 263 mUnregisteredWriters.pop(); 264 mediaLogService->unregisterWriter(iMemory); 265 } 266 } 267 268} 269 270static const char * const audio_interfaces[] = { 271 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 272 AUDIO_HARDWARE_MODULE_ID_A2DP, 273 AUDIO_HARDWARE_MODULE_ID_USB, 274}; 275#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 276 277AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 278 audio_module_handle_t module, 279 audio_devices_t devices) 280{ 281 // if module is 0, the request comes from an old policy manager and we should load 282 // well known modules 283 if (module == 0) { 284 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 285 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 286 loadHwModule_l(audio_interfaces[i]); 287 } 288 // then try to find a module supporting the requested device. 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 291 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 292 if ((dev->get_supported_devices != NULL) && 293 (dev->get_supported_devices(dev) & devices) == devices) 294 return audioHwDevice; 295 } 296 } else { 297 // check a match for the requested module handle 298 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 299 if (audioHwDevice != NULL) { 300 return audioHwDevice; 301 } 302 } 303 304 return NULL; 305} 306 307void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Notification Clients:\n"); 323 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 324 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 325 result.append(buffer); 326 } 327 328 result.append("Global session refs:\n"); 329 result.append(" session pid count\n"); 330 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 331 AudioSessionRef *r = mAudioSessionRefs[i]; 332 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 333 result.append(buffer); 334 } 335 write(fd, result.string(), result.size()); 336} 337 338 339void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 340{ 341 const size_t SIZE = 256; 342 char buffer[SIZE]; 343 String8 result; 344 hardware_call_state hardwareStatus = mHardwareStatus; 345 346 snprintf(buffer, SIZE, "Hardware status: %d\n" 347 "Standby Time mSec: %u\n", 348 hardwareStatus, 349 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352} 353 354void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 355{ 356 const size_t SIZE = 256; 357 char buffer[SIZE]; 358 String8 result; 359 snprintf(buffer, SIZE, "Permission Denial: " 360 "can't dump AudioFlinger from pid=%d, uid=%d\n", 361 IPCThreadState::self()->getCallingPid(), 362 IPCThreadState::self()->getCallingUid()); 363 result.append(buffer); 364 write(fd, result.string(), result.size()); 365} 366 367bool AudioFlinger::dumpTryLock(Mutex& mutex) 368{ 369 bool locked = false; 370 for (int i = 0; i < kDumpLockRetries; ++i) { 371 if (mutex.tryLock() == NO_ERROR) { 372 locked = true; 373 break; 374 } 375 usleep(kDumpLockSleepUs); 376 } 377 return locked; 378} 379 380status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 381{ 382 if (!dumpAllowed()) { 383 dumpPermissionDenial(fd, args); 384 } else { 385 // get state of hardware lock 386 bool hardwareLocked = dumpTryLock(mHardwareLock); 387 if (!hardwareLocked) { 388 String8 result(kHardwareLockedString); 389 write(fd, result.string(), result.size()); 390 } else { 391 mHardwareLock.unlock(); 392 } 393 394 bool locked = dumpTryLock(mLock); 395 396 // failed to lock - AudioFlinger is probably deadlocked 397 if (!locked) { 398 String8 result(kDeadlockedString); 399 write(fd, result.string(), result.size()); 400 } 401 402 bool clientLocked = dumpTryLock(mClientLock); 403 if (!clientLocked) { 404 String8 result(kClientLockedString); 405 write(fd, result.string(), result.size()); 406 } 407 408 EffectDumpEffects(fd); 409 410 dumpClients(fd, args); 411 if (clientLocked) { 412 mClientLock.unlock(); 413 } 414 415 dumpInternals(fd, args); 416 417 // dump playback threads 418 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 419 mPlaybackThreads.valueAt(i)->dump(fd, args); 420 } 421 422 // dump record threads 423 for (size_t i = 0; i < mRecordThreads.size(); i++) { 424 mRecordThreads.valueAt(i)->dump(fd, args); 425 } 426 427 // dump orphan effect chains 428 if (mOrphanEffectChains.size() != 0) { 429 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 430 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 431 mOrphanEffectChains.valueAt(i)->dump(fd, args); 432 } 433 } 434 // dump all hardware devs 435 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 436 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 437 dev->dump(dev, fd); 438 } 439 440#ifdef TEE_SINK 441 // dump the serially shared record tee sink 442 if (mRecordTeeSource != 0) { 443 dumpTee(fd, mRecordTeeSource); 444 } 445#endif 446 447 if (locked) { 448 mLock.unlock(); 449 } 450 451 // append a copy of media.log here by forwarding fd to it, but don't attempt 452 // to lookup the service if it's not running, as it will block for a second 453 if (mLogMemoryDealer != 0) { 454 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 455 if (binder != 0) { 456 dprintf(fd, "\nmedia.log:\n"); 457 Vector<String16> args; 458 binder->dump(fd, args); 459 } 460 } 461 } 462 return NO_ERROR; 463} 464 465sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 466{ 467 Mutex::Autolock _cl(mClientLock); 468 // If pid is already in the mClients wp<> map, then use that entry 469 // (for which promote() is always != 0), otherwise create a new entry and Client. 470 sp<Client> client = mClients.valueFor(pid).promote(); 471 if (client == 0) { 472 client = new Client(this, pid); 473 mClients.add(pid, client); 474 } 475 476 return client; 477} 478 479sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 480{ 481 // If there is no memory allocated for logs, return a dummy writer that does nothing 482 if (mLogMemoryDealer == 0) { 483 return new NBLog::Writer(); 484 } 485 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 486 // Similarly if we can't contact the media.log service, also return a dummy writer 487 if (binder == 0) { 488 return new NBLog::Writer(); 489 } 490 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 491 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 492 // If allocation fails, consult the vector of previously unregistered writers 493 // and garbage-collect one or more them until an allocation succeeds 494 if (shared == 0) { 495 Mutex::Autolock _l(mUnregisteredWritersLock); 496 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 497 { 498 // Pick the oldest stale writer to garbage-collect 499 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 500 mUnregisteredWriters.removeAt(0); 501 mediaLogService->unregisterWriter(iMemory); 502 // Now the media.log remote reference to IMemory is gone. When our last local 503 // reference to IMemory also drops to zero at end of this block, 504 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 505 } 506 // Re-attempt the allocation 507 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 508 if (shared != 0) { 509 goto success; 510 } 511 } 512 // Even after garbage-collecting all old writers, there is still not enough memory, 513 // so return a dummy writer 514 return new NBLog::Writer(); 515 } 516success: 517 mediaLogService->registerWriter(shared, size, name); 518 return new NBLog::Writer(size, shared); 519} 520 521void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 522{ 523 if (writer == 0) { 524 return; 525 } 526 sp<IMemory> iMemory(writer->getIMemory()); 527 if (iMemory == 0) { 528 return; 529 } 530 // Rather than removing the writer immediately, append it to a queue of old writers to 531 // be garbage-collected later. This allows us to continue to view old logs for a while. 532 Mutex::Autolock _l(mUnregisteredWritersLock); 533 mUnregisteredWriters.push(writer); 534} 535 536// IAudioFlinger interface 537 538 539sp<IAudioTrack> AudioFlinger::createTrack( 540 audio_stream_type_t streamType, 541 uint32_t sampleRate, 542 audio_format_t format, 543 audio_channel_mask_t channelMask, 544 size_t *frameCount, 545 IAudioFlinger::track_flags_t *flags, 546 const sp<IMemory>& sharedBuffer, 547 audio_io_handle_t output, 548 pid_t tid, 549 int *sessionId, 550 int clientUid, 551 status_t *status) 552{ 553 sp<PlaybackThread::Track> track; 554 sp<TrackHandle> trackHandle; 555 sp<Client> client; 556 status_t lStatus; 557 int lSessionId; 558 559 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 560 // but if someone uses binder directly they could bypass that and cause us to crash 561 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 562 ALOGE("createTrack() invalid stream type %d", streamType); 563 lStatus = BAD_VALUE; 564 goto Exit; 565 } 566 567 // further sample rate checks are performed by createTrack_l() depending on the thread type 568 if (sampleRate == 0) { 569 ALOGE("createTrack() invalid sample rate %u", sampleRate); 570 lStatus = BAD_VALUE; 571 goto Exit; 572 } 573 574 // further channel mask checks are performed by createTrack_l() depending on the thread type 575 if (!audio_is_output_channel(channelMask)) { 576 ALOGE("createTrack() invalid channel mask %#x", channelMask); 577 lStatus = BAD_VALUE; 578 goto Exit; 579 } 580 581 // further format checks are performed by createTrack_l() depending on the thread type 582 if (!audio_is_valid_format(format)) { 583 ALOGE("createTrack() invalid format %#x", format); 584 lStatus = BAD_VALUE; 585 goto Exit; 586 } 587 588 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 589 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 590 lStatus = BAD_VALUE; 591 goto Exit; 592 } 593 594 { 595 Mutex::Autolock _l(mLock); 596 PlaybackThread *thread = checkPlaybackThread_l(output); 597 if (thread == NULL) { 598 ALOGE("no playback thread found for output handle %d", output); 599 lStatus = BAD_VALUE; 600 goto Exit; 601 } 602 603 pid_t pid = IPCThreadState::self()->getCallingPid(); 604 client = registerPid(pid); 605 606 PlaybackThread *effectThread = NULL; 607 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 608 lSessionId = *sessionId; 609 // check if an effect chain with the same session ID is present on another 610 // output thread and move it here. 611 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 612 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 613 if (mPlaybackThreads.keyAt(i) != output) { 614 uint32_t sessions = t->hasAudioSession(lSessionId); 615 if (sessions & PlaybackThread::EFFECT_SESSION) { 616 effectThread = t.get(); 617 break; 618 } 619 } 620 } 621 } else { 622 // if no audio session id is provided, create one here 623 lSessionId = nextUniqueId(); 624 if (sessionId != NULL) { 625 *sessionId = lSessionId; 626 } 627 } 628 ALOGV("createTrack() lSessionId: %d", lSessionId); 629 630 track = thread->createTrack_l(client, streamType, sampleRate, format, 631 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 632 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 633 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 634 635 // move effect chain to this output thread if an effect on same session was waiting 636 // for a track to be created 637 if (lStatus == NO_ERROR && effectThread != NULL) { 638 // no risk of deadlock because AudioFlinger::mLock is held 639 Mutex::Autolock _dl(thread->mLock); 640 Mutex::Autolock _sl(effectThread->mLock); 641 moveEffectChain_l(lSessionId, effectThread, thread, true); 642 } 643 644 // Look for sync events awaiting for a session to be used. 645 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 646 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 647 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 648 if (lStatus == NO_ERROR) { 649 (void) track->setSyncEvent(mPendingSyncEvents[i]); 650 } else { 651 mPendingSyncEvents[i]->cancel(); 652 } 653 mPendingSyncEvents.removeAt(i); 654 i--; 655 } 656 } 657 } 658 659 setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); 660 } 661 662 if (lStatus != NO_ERROR) { 663 // remove local strong reference to Client before deleting the Track so that the 664 // Client destructor is called by the TrackBase destructor with mClientLock held 665 // Don't hold mClientLock when releasing the reference on the track as the 666 // destructor will acquire it. 667 { 668 Mutex::Autolock _cl(mClientLock); 669 client.clear(); 670 } 671 track.clear(); 672 goto Exit; 673 } 674 675 // return handle to client 676 trackHandle = new TrackHandle(track); 677 678Exit: 679 *status = lStatus; 680 return trackHandle; 681} 682 683uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 684{ 685 Mutex::Autolock _l(mLock); 686 PlaybackThread *thread = checkPlaybackThread_l(output); 687 if (thread == NULL) { 688 ALOGW("sampleRate() unknown thread %d", output); 689 return 0; 690 } 691 return thread->sampleRate(); 692} 693 694audio_format_t AudioFlinger::format(audio_io_handle_t output) const 695{ 696 Mutex::Autolock _l(mLock); 697 PlaybackThread *thread = checkPlaybackThread_l(output); 698 if (thread == NULL) { 699 ALOGW("format() unknown thread %d", output); 700 return AUDIO_FORMAT_INVALID; 701 } 702 return thread->format(); 703} 704 705size_t AudioFlinger::frameCount(audio_io_handle_t output) const 706{ 707 Mutex::Autolock _l(mLock); 708 PlaybackThread *thread = checkPlaybackThread_l(output); 709 if (thread == NULL) { 710 ALOGW("frameCount() unknown thread %d", output); 711 return 0; 712 } 713 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 714 // should examine all callers and fix them to handle smaller counts 715 return thread->frameCount(); 716} 717 718uint32_t AudioFlinger::latency(audio_io_handle_t output) const 719{ 720 Mutex::Autolock _l(mLock); 721 PlaybackThread *thread = checkPlaybackThread_l(output); 722 if (thread == NULL) { 723 ALOGW("latency(): no playback thread found for output handle %d", output); 724 return 0; 725 } 726 return thread->latency(); 727} 728 729status_t AudioFlinger::setMasterVolume(float value) 730{ 731 status_t ret = initCheck(); 732 if (ret != NO_ERROR) { 733 return ret; 734 } 735 736 // check calling permissions 737 if (!settingsAllowed()) { 738 return PERMISSION_DENIED; 739 } 740 741 Mutex::Autolock _l(mLock); 742 mMasterVolume = value; 743 744 // Set master volume in the HALs which support it. 745 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 746 AutoMutex lock(mHardwareLock); 747 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 748 749 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 750 if (dev->canSetMasterVolume()) { 751 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 752 } 753 mHardwareStatus = AUDIO_HW_IDLE; 754 } 755 756 // Now set the master volume in each playback thread. Playback threads 757 // assigned to HALs which do not have master volume support will apply 758 // master volume during the mix operation. Threads with HALs which do 759 // support master volume will simply ignore the setting. 760 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 761 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 762 763 return NO_ERROR; 764} 765 766status_t AudioFlinger::setMode(audio_mode_t mode) 767{ 768 status_t ret = initCheck(); 769 if (ret != NO_ERROR) { 770 return ret; 771 } 772 773 // check calling permissions 774 if (!settingsAllowed()) { 775 return PERMISSION_DENIED; 776 } 777 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 778 ALOGW("Illegal value: setMode(%d)", mode); 779 return BAD_VALUE; 780 } 781 782 { // scope for the lock 783 AutoMutex lock(mHardwareLock); 784 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 785 mHardwareStatus = AUDIO_HW_SET_MODE; 786 ret = dev->set_mode(dev, mode); 787 mHardwareStatus = AUDIO_HW_IDLE; 788 } 789 790 if (NO_ERROR == ret) { 791 Mutex::Autolock _l(mLock); 792 mMode = mode; 793 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 794 mPlaybackThreads.valueAt(i)->setMode(mode); 795 } 796 797 return ret; 798} 799 800status_t AudioFlinger::setMicMute(bool state) 801{ 802 status_t ret = initCheck(); 803 if (ret != NO_ERROR) { 804 return ret; 805 } 806 807 // check calling permissions 808 if (!settingsAllowed()) { 809 return PERMISSION_DENIED; 810 } 811 812 AutoMutex lock(mHardwareLock); 813 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 814 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 815 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 816 status_t result = dev->set_mic_mute(dev, state); 817 if (result != NO_ERROR) { 818 ret = result; 819 } 820 } 821 mHardwareStatus = AUDIO_HW_IDLE; 822 return ret; 823} 824 825bool AudioFlinger::getMicMute() const 826{ 827 status_t ret = initCheck(); 828 if (ret != NO_ERROR) { 829 return false; 830 } 831 bool mute = true; 832 bool state = AUDIO_MODE_INVALID; 833 AutoMutex lock(mHardwareLock); 834 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 835 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 836 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 837 status_t result = dev->get_mic_mute(dev, &state); 838 if (result == NO_ERROR) { 839 mute = mute && state; 840 } 841 } 842 mHardwareStatus = AUDIO_HW_IDLE; 843 844 return mute; 845} 846 847status_t AudioFlinger::setMasterMute(bool muted) 848{ 849 status_t ret = initCheck(); 850 if (ret != NO_ERROR) { 851 return ret; 852 } 853 854 // check calling permissions 855 if (!settingsAllowed()) { 856 return PERMISSION_DENIED; 857 } 858 859 Mutex::Autolock _l(mLock); 860 mMasterMute = muted; 861 862 // Set master mute in the HALs which support it. 863 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 864 AutoMutex lock(mHardwareLock); 865 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 866 867 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 868 if (dev->canSetMasterMute()) { 869 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 870 } 871 mHardwareStatus = AUDIO_HW_IDLE; 872 } 873 874 // Now set the master mute in each playback thread. Playback threads 875 // assigned to HALs which do not have master mute support will apply master 876 // mute during the mix operation. Threads with HALs which do support master 877 // mute will simply ignore the setting. 878 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 879 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 880 881 return NO_ERROR; 882} 883 884float AudioFlinger::masterVolume() const 885{ 886 Mutex::Autolock _l(mLock); 887 return masterVolume_l(); 888} 889 890bool AudioFlinger::masterMute() const 891{ 892 Mutex::Autolock _l(mLock); 893 return masterMute_l(); 894} 895 896float AudioFlinger::masterVolume_l() const 897{ 898 return mMasterVolume; 899} 900 901bool AudioFlinger::masterMute_l() const 902{ 903 return mMasterMute; 904} 905 906status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 907{ 908 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 909 ALOGW("setStreamVolume() invalid stream %d", stream); 910 return BAD_VALUE; 911 } 912 pid_t caller = IPCThreadState::self()->getCallingPid(); 913 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 914 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 915 return PERMISSION_DENIED; 916 } 917 918 return NO_ERROR; 919} 920 921status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 922 audio_io_handle_t output) 923{ 924 // check calling permissions 925 if (!settingsAllowed()) { 926 return PERMISSION_DENIED; 927 } 928 929 status_t status = checkStreamType(stream); 930 if (status != NO_ERROR) { 931 return status; 932 } 933 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 934 935 AutoMutex lock(mLock); 936 PlaybackThread *thread = NULL; 937 if (output != AUDIO_IO_HANDLE_NONE) { 938 thread = checkPlaybackThread_l(output); 939 if (thread == NULL) { 940 return BAD_VALUE; 941 } 942 } 943 944 mStreamTypes[stream].volume = value; 945 946 if (thread == NULL) { 947 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 948 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 949 } 950 } else { 951 thread->setStreamVolume(stream, value); 952 } 953 954 return NO_ERROR; 955} 956 957status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 958{ 959 // check calling permissions 960 if (!settingsAllowed()) { 961 return PERMISSION_DENIED; 962 } 963 964 status_t status = checkStreamType(stream); 965 if (status != NO_ERROR) { 966 return status; 967 } 968 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 969 970 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 971 ALOGE("setStreamMute() invalid stream %d", stream); 972 return BAD_VALUE; 973 } 974 975 AutoMutex lock(mLock); 976 mStreamTypes[stream].mute = muted; 977 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 978 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 979 980 return NO_ERROR; 981} 982 983float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 984{ 985 status_t status = checkStreamType(stream); 986 if (status != NO_ERROR) { 987 return 0.0f; 988 } 989 990 AutoMutex lock(mLock); 991 float volume; 992 if (output != AUDIO_IO_HANDLE_NONE) { 993 PlaybackThread *thread = checkPlaybackThread_l(output); 994 if (thread == NULL) { 995 return 0.0f; 996 } 997 volume = thread->streamVolume(stream); 998 } else { 999 volume = streamVolume_l(stream); 1000 } 1001 1002 return volume; 1003} 1004 1005bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1006{ 1007 status_t status = checkStreamType(stream); 1008 if (status != NO_ERROR) { 1009 return true; 1010 } 1011 1012 AutoMutex lock(mLock); 1013 return streamMute_l(stream); 1014} 1015 1016 1017void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1018{ 1019 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1020 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1021 } 1022} 1023 1024status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1025{ 1026 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1027 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1028 1029 // check calling permissions 1030 if (!settingsAllowed()) { 1031 return PERMISSION_DENIED; 1032 } 1033 1034 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1035 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1036 Mutex::Autolock _l(mLock); 1037 status_t final_result = NO_ERROR; 1038 { 1039 AutoMutex lock(mHardwareLock); 1040 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1041 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1042 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1043 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1044 final_result = result ?: final_result; 1045 } 1046 mHardwareStatus = AUDIO_HW_IDLE; 1047 } 1048 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1049 AudioParameter param = AudioParameter(keyValuePairs); 1050 String8 value; 1051 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1052 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1053 if (mBtNrecIsOff != btNrecIsOff) { 1054 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1055 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1056 audio_devices_t device = thread->inDevice(); 1057 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1058 // collect all of the thread's session IDs 1059 KeyedVector<int, bool> ids = thread->sessionIds(); 1060 // suspend effects associated with those session IDs 1061 for (size_t j = 0; j < ids.size(); ++j) { 1062 int sessionId = ids.keyAt(j); 1063 thread->setEffectSuspended(FX_IID_AEC, 1064 suspend, 1065 sessionId); 1066 thread->setEffectSuspended(FX_IID_NS, 1067 suspend, 1068 sessionId); 1069 } 1070 } 1071 mBtNrecIsOff = btNrecIsOff; 1072 } 1073 } 1074 String8 screenState; 1075 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1076 bool isOff = screenState == "off"; 1077 if (isOff != (AudioFlinger::mScreenState & 1)) { 1078 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1079 } 1080 } 1081 return final_result; 1082 } 1083 1084 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1085 // and the thread is exited once the lock is released 1086 sp<ThreadBase> thread; 1087 { 1088 Mutex::Autolock _l(mLock); 1089 thread = checkPlaybackThread_l(ioHandle); 1090 if (thread == 0) { 1091 thread = checkRecordThread_l(ioHandle); 1092 } else if (thread == primaryPlaybackThread_l()) { 1093 // indicate output device change to all input threads for pre processing 1094 AudioParameter param = AudioParameter(keyValuePairs); 1095 int value; 1096 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1097 (value != 0)) { 1098 broacastParametersToRecordThreads_l(keyValuePairs); 1099 } 1100 } 1101 } 1102 if (thread != 0) { 1103 return thread->setParameters(keyValuePairs); 1104 } 1105 return BAD_VALUE; 1106} 1107 1108String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1109{ 1110 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1111 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1112 1113 Mutex::Autolock _l(mLock); 1114 1115 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1116 String8 out_s8; 1117 1118 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1119 char *s; 1120 { 1121 AutoMutex lock(mHardwareLock); 1122 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1123 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1124 s = dev->get_parameters(dev, keys.string()); 1125 mHardwareStatus = AUDIO_HW_IDLE; 1126 } 1127 out_s8 += String8(s ? s : ""); 1128 free(s); 1129 } 1130 return out_s8; 1131 } 1132 1133 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1134 if (playbackThread != NULL) { 1135 return playbackThread->getParameters(keys); 1136 } 1137 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1138 if (recordThread != NULL) { 1139 return recordThread->getParameters(keys); 1140 } 1141 return String8(""); 1142} 1143 1144size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1145 audio_channel_mask_t channelMask) const 1146{ 1147 status_t ret = initCheck(); 1148 if (ret != NO_ERROR) { 1149 return 0; 1150 } 1151 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1152 return 0; 1153 } 1154 1155 AutoMutex lock(mHardwareLock); 1156 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1157 audio_config_t config, proposed; 1158 memset(&proposed, 0, sizeof(proposed)); 1159 proposed.sample_rate = sampleRate; 1160 proposed.channel_mask = channelMask; 1161 proposed.format = format; 1162 1163 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1164 size_t frames; 1165 for (;;) { 1166 // Note: config is currently a const parameter for get_input_buffer_size() 1167 // but we use a copy from proposed in case config changes from the call. 1168 config = proposed; 1169 frames = dev->get_input_buffer_size(dev, &config); 1170 if (frames != 0) { 1171 break; // hal success, config is the result 1172 } 1173 // change one parameter of the configuration each iteration to a more "common" value 1174 // to see if the device will support it. 1175 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1176 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1177 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1178 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1179 } else { 1180 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1181 "format %#x, channelMask 0x%X", 1182 sampleRate, format, channelMask); 1183 break; // retries failed, break out of loop with frames == 0. 1184 } 1185 } 1186 mHardwareStatus = AUDIO_HW_IDLE; 1187 if (frames > 0 && config.sample_rate != sampleRate) { 1188 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1189 } 1190 return frames; // may be converted to bytes at the Java level. 1191} 1192 1193uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1194{ 1195 Mutex::Autolock _l(mLock); 1196 1197 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1198 if (recordThread != NULL) { 1199 return recordThread->getInputFramesLost(); 1200 } 1201 return 0; 1202} 1203 1204status_t AudioFlinger::setVoiceVolume(float value) 1205{ 1206 status_t ret = initCheck(); 1207 if (ret != NO_ERROR) { 1208 return ret; 1209 } 1210 1211 // check calling permissions 1212 if (!settingsAllowed()) { 1213 return PERMISSION_DENIED; 1214 } 1215 1216 AutoMutex lock(mHardwareLock); 1217 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1218 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1219 ret = dev->set_voice_volume(dev, value); 1220 mHardwareStatus = AUDIO_HW_IDLE; 1221 1222 return ret; 1223} 1224 1225status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1226 audio_io_handle_t output) const 1227{ 1228 status_t status; 1229 1230 Mutex::Autolock _l(mLock); 1231 1232 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1233 if (playbackThread != NULL) { 1234 return playbackThread->getRenderPosition(halFrames, dspFrames); 1235 } 1236 1237 return BAD_VALUE; 1238} 1239 1240void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1241{ 1242 Mutex::Autolock _l(mLock); 1243 if (client == 0) { 1244 return; 1245 } 1246 bool clientAdded = false; 1247 { 1248 Mutex::Autolock _cl(mClientLock); 1249 1250 pid_t pid = IPCThreadState::self()->getCallingPid(); 1251 if (mNotificationClients.indexOfKey(pid) < 0) { 1252 sp<NotificationClient> notificationClient = new NotificationClient(this, 1253 client, 1254 pid); 1255 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1256 1257 mNotificationClients.add(pid, notificationClient); 1258 1259 sp<IBinder> binder = IInterface::asBinder(client); 1260 binder->linkToDeath(notificationClient); 1261 clientAdded = true; 1262 } 1263 } 1264 1265 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1266 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1267 if (clientAdded) { 1268 // the config change is always sent from playback or record threads to avoid deadlock 1269 // with AudioSystem::gLock 1270 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1271 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED); 1272 } 1273 1274 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1275 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED); 1276 } 1277 } 1278} 1279 1280void AudioFlinger::removeNotificationClient(pid_t pid) 1281{ 1282 Mutex::Autolock _l(mLock); 1283 { 1284 Mutex::Autolock _cl(mClientLock); 1285 mNotificationClients.removeItem(pid); 1286 } 1287 1288 ALOGV("%d died, releasing its sessions", pid); 1289 size_t num = mAudioSessionRefs.size(); 1290 bool removed = false; 1291 for (size_t i = 0; i< num; ) { 1292 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1293 ALOGV(" pid %d @ %d", ref->mPid, i); 1294 if (ref->mPid == pid) { 1295 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1296 mAudioSessionRefs.removeAt(i); 1297 delete ref; 1298 removed = true; 1299 num--; 1300 } else { 1301 i++; 1302 } 1303 } 1304 if (removed) { 1305 purgeStaleEffects_l(); 1306 } 1307} 1308 1309void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1310 const sp<AudioIoDescriptor>& ioDesc) 1311{ 1312 Mutex::Autolock _l(mClientLock); 1313 size_t size = mNotificationClients.size(); 1314 for (size_t i = 0; i < size; i++) { 1315 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1316 } 1317} 1318 1319// removeClient_l() must be called with AudioFlinger::mClientLock held 1320void AudioFlinger::removeClient_l(pid_t pid) 1321{ 1322 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1323 IPCThreadState::self()->getCallingPid()); 1324 mClients.removeItem(pid); 1325} 1326 1327// getEffectThread_l() must be called with AudioFlinger::mLock held 1328sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1329{ 1330 sp<PlaybackThread> thread; 1331 1332 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1333 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1334 ALOG_ASSERT(thread == 0); 1335 thread = mPlaybackThreads.valueAt(i); 1336 } 1337 } 1338 1339 return thread; 1340} 1341 1342 1343 1344// ---------------------------------------------------------------------------- 1345 1346AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1347 : RefBase(), 1348 mAudioFlinger(audioFlinger), 1349 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1350 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1351 mPid(pid), 1352 mTimedTrackCount(0) 1353{ 1354 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1355} 1356 1357// Client destructor must be called with AudioFlinger::mClientLock held 1358AudioFlinger::Client::~Client() 1359{ 1360 mAudioFlinger->removeClient_l(mPid); 1361} 1362 1363sp<MemoryDealer> AudioFlinger::Client::heap() const 1364{ 1365 return mMemoryDealer; 1366} 1367 1368// Reserve one of the limited slots for a timed audio track associated 1369// with this client 1370bool AudioFlinger::Client::reserveTimedTrack() 1371{ 1372 const int kMaxTimedTracksPerClient = 4; 1373 1374 Mutex::Autolock _l(mTimedTrackLock); 1375 1376 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1377 ALOGW("can not create timed track - pid %d has exceeded the limit", 1378 mPid); 1379 return false; 1380 } 1381 1382 mTimedTrackCount++; 1383 return true; 1384} 1385 1386// Release a slot for a timed audio track 1387void AudioFlinger::Client::releaseTimedTrack() 1388{ 1389 Mutex::Autolock _l(mTimedTrackLock); 1390 mTimedTrackCount--; 1391} 1392 1393// ---------------------------------------------------------------------------- 1394 1395AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1396 const sp<IAudioFlingerClient>& client, 1397 pid_t pid) 1398 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1399{ 1400} 1401 1402AudioFlinger::NotificationClient::~NotificationClient() 1403{ 1404} 1405 1406void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1407{ 1408 sp<NotificationClient> keep(this); 1409 mAudioFlinger->removeNotificationClient(mPid); 1410} 1411 1412 1413// ---------------------------------------------------------------------------- 1414 1415static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1416 return audio_is_remote_submix_device(inDevice); 1417} 1418 1419sp<IAudioRecord> AudioFlinger::openRecord( 1420 audio_io_handle_t input, 1421 uint32_t sampleRate, 1422 audio_format_t format, 1423 audio_channel_mask_t channelMask, 1424 const String16& opPackageName, 1425 size_t *frameCount, 1426 IAudioFlinger::track_flags_t *flags, 1427 pid_t tid, 1428 int *sessionId, 1429 size_t *notificationFrames, 1430 sp<IMemory>& cblk, 1431 sp<IMemory>& buffers, 1432 status_t *status) 1433{ 1434 sp<RecordThread::RecordTrack> recordTrack; 1435 sp<RecordHandle> recordHandle; 1436 sp<Client> client; 1437 status_t lStatus; 1438 int lSessionId; 1439 1440 cblk.clear(); 1441 buffers.clear(); 1442 1443 // check calling permissions 1444 if (!recordingAllowed(opPackageName)) { 1445 ALOGE("openRecord() permission denied: recording not allowed"); 1446 lStatus = PERMISSION_DENIED; 1447 goto Exit; 1448 } 1449 1450 // further sample rate checks are performed by createRecordTrack_l() 1451 if (sampleRate == 0) { 1452 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1453 lStatus = BAD_VALUE; 1454 goto Exit; 1455 } 1456 1457 // we don't yet support anything other than linear PCM 1458 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1459 ALOGE("openRecord() invalid format %#x", format); 1460 lStatus = BAD_VALUE; 1461 goto Exit; 1462 } 1463 1464 // further channel mask checks are performed by createRecordTrack_l() 1465 if (!audio_is_input_channel(channelMask)) { 1466 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1467 lStatus = BAD_VALUE; 1468 goto Exit; 1469 } 1470 1471 { 1472 Mutex::Autolock _l(mLock); 1473 RecordThread *thread = checkRecordThread_l(input); 1474 if (thread == NULL) { 1475 ALOGE("openRecord() checkRecordThread_l failed"); 1476 lStatus = BAD_VALUE; 1477 goto Exit; 1478 } 1479 1480 pid_t pid = IPCThreadState::self()->getCallingPid(); 1481 client = registerPid(pid); 1482 1483 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1484 lSessionId = *sessionId; 1485 } else { 1486 // if no audio session id is provided, create one here 1487 lSessionId = nextUniqueId(); 1488 if (sessionId != NULL) { 1489 *sessionId = lSessionId; 1490 } 1491 } 1492 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1493 1494 // TODO: the uid should be passed in as a parameter to openRecord 1495 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1496 frameCount, lSessionId, notificationFrames, 1497 IPCThreadState::self()->getCallingUid(), 1498 flags, tid, &lStatus); 1499 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1500 1501 if (lStatus == NO_ERROR) { 1502 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1503 // session and move it to this thread. 1504 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId); 1505 if (chain != 0) { 1506 Mutex::Autolock _l(thread->mLock); 1507 thread->addEffectChain_l(chain); 1508 } 1509 } 1510 } 1511 1512 if (lStatus != NO_ERROR) { 1513 // remove local strong reference to Client before deleting the RecordTrack so that the 1514 // Client destructor is called by the TrackBase destructor with mClientLock held 1515 // Don't hold mClientLock when releasing the reference on the track as the 1516 // destructor will acquire it. 1517 { 1518 Mutex::Autolock _cl(mClientLock); 1519 client.clear(); 1520 } 1521 recordTrack.clear(); 1522 goto Exit; 1523 } 1524 1525 cblk = recordTrack->getCblk(); 1526 buffers = recordTrack->getBuffers(); 1527 1528 // return handle to client 1529 recordHandle = new RecordHandle(recordTrack); 1530 1531Exit: 1532 *status = lStatus; 1533 return recordHandle; 1534} 1535 1536 1537 1538// ---------------------------------------------------------------------------- 1539 1540audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1541{ 1542 if (name == NULL) { 1543 return 0; 1544 } 1545 if (!settingsAllowed()) { 1546 return 0; 1547 } 1548 Mutex::Autolock _l(mLock); 1549 return loadHwModule_l(name); 1550} 1551 1552// loadHwModule_l() must be called with AudioFlinger::mLock held 1553audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1554{ 1555 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1556 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1557 ALOGW("loadHwModule() module %s already loaded", name); 1558 return mAudioHwDevs.keyAt(i); 1559 } 1560 } 1561 1562 audio_hw_device_t *dev; 1563 1564 int rc = load_audio_interface(name, &dev); 1565 if (rc) { 1566 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1567 return 0; 1568 } 1569 1570 mHardwareStatus = AUDIO_HW_INIT; 1571 rc = dev->init_check(dev); 1572 mHardwareStatus = AUDIO_HW_IDLE; 1573 if (rc) { 1574 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1575 return 0; 1576 } 1577 1578 // Check and cache this HAL's level of support for master mute and master 1579 // volume. If this is the first HAL opened, and it supports the get 1580 // methods, use the initial values provided by the HAL as the current 1581 // master mute and volume settings. 1582 1583 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1584 { // scope for auto-lock pattern 1585 AutoMutex lock(mHardwareLock); 1586 1587 if (0 == mAudioHwDevs.size()) { 1588 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1589 if (NULL != dev->get_master_volume) { 1590 float mv; 1591 if (OK == dev->get_master_volume(dev, &mv)) { 1592 mMasterVolume = mv; 1593 } 1594 } 1595 1596 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1597 if (NULL != dev->get_master_mute) { 1598 bool mm; 1599 if (OK == dev->get_master_mute(dev, &mm)) { 1600 mMasterMute = mm; 1601 } 1602 } 1603 } 1604 1605 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1606 if ((NULL != dev->set_master_volume) && 1607 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1608 flags = static_cast<AudioHwDevice::Flags>(flags | 1609 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1610 } 1611 1612 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1613 if ((NULL != dev->set_master_mute) && 1614 (OK == dev->set_master_mute(dev, mMasterMute))) { 1615 flags = static_cast<AudioHwDevice::Flags>(flags | 1616 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1617 } 1618 1619 mHardwareStatus = AUDIO_HW_IDLE; 1620 } 1621 1622 audio_module_handle_t handle = nextUniqueId(); 1623 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1624 1625 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1626 name, dev->common.module->name, dev->common.module->id, handle); 1627 1628 return handle; 1629 1630} 1631 1632// ---------------------------------------------------------------------------- 1633 1634uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1635{ 1636 Mutex::Autolock _l(mLock); 1637 PlaybackThread *thread = primaryPlaybackThread_l(); 1638 return thread != NULL ? thread->sampleRate() : 0; 1639} 1640 1641size_t AudioFlinger::getPrimaryOutputFrameCount() 1642{ 1643 Mutex::Autolock _l(mLock); 1644 PlaybackThread *thread = primaryPlaybackThread_l(); 1645 return thread != NULL ? thread->frameCountHAL() : 0; 1646} 1647 1648// ---------------------------------------------------------------------------- 1649 1650status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1651{ 1652 uid_t uid = IPCThreadState::self()->getCallingUid(); 1653 if (uid != AID_SYSTEM) { 1654 return PERMISSION_DENIED; 1655 } 1656 Mutex::Autolock _l(mLock); 1657 if (mIsDeviceTypeKnown) { 1658 return INVALID_OPERATION; 1659 } 1660 mIsLowRamDevice = isLowRamDevice; 1661 mIsDeviceTypeKnown = true; 1662 return NO_ERROR; 1663} 1664 1665audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1666{ 1667 Mutex::Autolock _l(mLock); 1668 1669 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1670 if (index >= 0) { 1671 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1672 mHwAvSyncIds.valueAt(index), sessionId); 1673 return mHwAvSyncIds.valueAt(index); 1674 } 1675 1676 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1677 if (dev == NULL) { 1678 return AUDIO_HW_SYNC_INVALID; 1679 } 1680 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1681 AudioParameter param = AudioParameter(String8(reply)); 1682 free(reply); 1683 1684 int value; 1685 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1686 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1687 return AUDIO_HW_SYNC_INVALID; 1688 } 1689 1690 // allow only one session for a given HW A/V sync ID. 1691 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1692 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1693 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1694 value, mHwAvSyncIds.keyAt(i)); 1695 mHwAvSyncIds.removeItemsAt(i); 1696 break; 1697 } 1698 } 1699 1700 mHwAvSyncIds.add(sessionId, value); 1701 1702 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1703 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1704 uint32_t sessions = thread->hasAudioSession(sessionId); 1705 if (sessions & PlaybackThread::TRACK_SESSION) { 1706 AudioParameter param = AudioParameter(); 1707 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1708 thread->setParameters(param.toString()); 1709 break; 1710 } 1711 } 1712 1713 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1714 return (audio_hw_sync_t)value; 1715} 1716 1717// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1718void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1719{ 1720 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1721 if (index >= 0) { 1722 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1723 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1724 AudioParameter param = AudioParameter(); 1725 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1726 thread->setParameters(param.toString()); 1727 } 1728} 1729 1730 1731// ---------------------------------------------------------------------------- 1732 1733 1734sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1735 audio_io_handle_t *output, 1736 audio_config_t *config, 1737 audio_devices_t devices, 1738 const String8& address, 1739 audio_output_flags_t flags) 1740{ 1741 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1742 if (outHwDev == NULL) { 1743 return 0; 1744 } 1745 1746 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1747 if (*output == AUDIO_IO_HANDLE_NONE) { 1748 *output = nextUniqueId(); 1749 } 1750 1751 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1752 1753 // FOR TESTING ONLY: 1754 // This if statement allows overriding the audio policy settings 1755 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1756 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1757 // Check only for Normal Mixing mode 1758 if (kEnableExtendedPrecision) { 1759 // Specify format (uncomment one below to choose) 1760 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1761 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1762 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1763 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1764 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1765 } 1766 if (kEnableExtendedChannels) { 1767 // Specify channel mask (uncomment one below to choose) 1768 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1769 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1770 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1771 } 1772 } 1773 1774 AudioStreamOut *outputStream = NULL; 1775 status_t status = outHwDev->openOutputStream( 1776 &outputStream, 1777 *output, 1778 devices, 1779 flags, 1780 config, 1781 address.string()); 1782 1783 mHardwareStatus = AUDIO_HW_IDLE; 1784 1785 if (status == NO_ERROR) { 1786 1787 PlaybackThread *thread; 1788 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1789 thread = new OffloadThread(this, outputStream, *output, devices); 1790 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1791 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1792 || !isValidPcmSinkFormat(config->format) 1793 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1794 thread = new DirectOutputThread(this, outputStream, *output, devices); 1795 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1796 } else { 1797 thread = new MixerThread(this, outputStream, *output, devices); 1798 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1799 } 1800 mPlaybackThreads.add(*output, thread); 1801 return thread; 1802 } 1803 1804 return 0; 1805} 1806 1807status_t AudioFlinger::openOutput(audio_module_handle_t module, 1808 audio_io_handle_t *output, 1809 audio_config_t *config, 1810 audio_devices_t *devices, 1811 const String8& address, 1812 uint32_t *latencyMs, 1813 audio_output_flags_t flags) 1814{ 1815 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1816 module, 1817 (devices != NULL) ? *devices : 0, 1818 config->sample_rate, 1819 config->format, 1820 config->channel_mask, 1821 flags); 1822 1823 if (*devices == AUDIO_DEVICE_NONE) { 1824 return BAD_VALUE; 1825 } 1826 1827 Mutex::Autolock _l(mLock); 1828 1829 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1830 if (thread != 0) { 1831 *latencyMs = thread->latency(); 1832 1833 // notify client processes of the new output creation 1834 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1835 1836 // the first primary output opened designates the primary hw device 1837 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1838 ALOGI("Using module %d has the primary audio interface", module); 1839 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1840 1841 AutoMutex lock(mHardwareLock); 1842 mHardwareStatus = AUDIO_HW_SET_MODE; 1843 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1844 mHardwareStatus = AUDIO_HW_IDLE; 1845 1846 mPrimaryOutputSampleRate = config->sample_rate; 1847 } 1848 return NO_ERROR; 1849 } 1850 1851 return NO_INIT; 1852} 1853 1854audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1855 audio_io_handle_t output2) 1856{ 1857 Mutex::Autolock _l(mLock); 1858 MixerThread *thread1 = checkMixerThread_l(output1); 1859 MixerThread *thread2 = checkMixerThread_l(output2); 1860 1861 if (thread1 == NULL || thread2 == NULL) { 1862 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1863 output2); 1864 return AUDIO_IO_HANDLE_NONE; 1865 } 1866 1867 audio_io_handle_t id = nextUniqueId(); 1868 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1869 thread->addOutputTrack(thread2); 1870 mPlaybackThreads.add(id, thread); 1871 // notify client processes of the new output creation 1872 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1873 return id; 1874} 1875 1876status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1877{ 1878 return closeOutput_nonvirtual(output); 1879} 1880 1881status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1882{ 1883 // keep strong reference on the playback thread so that 1884 // it is not destroyed while exit() is executed 1885 sp<PlaybackThread> thread; 1886 { 1887 Mutex::Autolock _l(mLock); 1888 thread = checkPlaybackThread_l(output); 1889 if (thread == NULL) { 1890 return BAD_VALUE; 1891 } 1892 1893 ALOGV("closeOutput() %d", output); 1894 1895 if (thread->type() == ThreadBase::MIXER) { 1896 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1897 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1898 DuplicatingThread *dupThread = 1899 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1900 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1901 1902 } 1903 } 1904 } 1905 1906 1907 mPlaybackThreads.removeItem(output); 1908 // save all effects to the default thread 1909 if (mPlaybackThreads.size()) { 1910 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1911 if (dstThread != NULL) { 1912 // audioflinger lock is held here so the acquisition order of thread locks does not 1913 // matter 1914 Mutex::Autolock _dl(dstThread->mLock); 1915 Mutex::Autolock _sl(thread->mLock); 1916 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1917 for (size_t i = 0; i < effectChains.size(); i ++) { 1918 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1919 } 1920 } 1921 } 1922 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1923 ioDesc->mIoHandle = output; 1924 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1925 } 1926 thread->exit(); 1927 // The thread entity (active unit of execution) is no longer running here, 1928 // but the ThreadBase container still exists. 1929 1930 if (thread->type() != ThreadBase::DUPLICATING) { 1931 closeOutputFinish(thread); 1932 } 1933 1934 return NO_ERROR; 1935} 1936 1937void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1938{ 1939 AudioStreamOut *out = thread->clearOutput(); 1940 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1941 // from now on thread->mOutput is NULL 1942 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1943 delete out; 1944} 1945 1946void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1947{ 1948 mPlaybackThreads.removeItem(thread->mId); 1949 thread->exit(); 1950 closeOutputFinish(thread); 1951} 1952 1953status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1954{ 1955 Mutex::Autolock _l(mLock); 1956 PlaybackThread *thread = checkPlaybackThread_l(output); 1957 1958 if (thread == NULL) { 1959 return BAD_VALUE; 1960 } 1961 1962 ALOGV("suspendOutput() %d", output); 1963 thread->suspend(); 1964 1965 return NO_ERROR; 1966} 1967 1968status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1969{ 1970 Mutex::Autolock _l(mLock); 1971 PlaybackThread *thread = checkPlaybackThread_l(output); 1972 1973 if (thread == NULL) { 1974 return BAD_VALUE; 1975 } 1976 1977 ALOGV("restoreOutput() %d", output); 1978 1979 thread->restore(); 1980 1981 return NO_ERROR; 1982} 1983 1984status_t AudioFlinger::openInput(audio_module_handle_t module, 1985 audio_io_handle_t *input, 1986 audio_config_t *config, 1987 audio_devices_t *devices, 1988 const String8& address, 1989 audio_source_t source, 1990 audio_input_flags_t flags) 1991{ 1992 Mutex::Autolock _l(mLock); 1993 1994 if (*devices == AUDIO_DEVICE_NONE) { 1995 return BAD_VALUE; 1996 } 1997 1998 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 1999 2000 if (thread != 0) { 2001 // notify client processes of the new input creation 2002 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2003 return NO_ERROR; 2004 } 2005 return NO_INIT; 2006} 2007 2008sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2009 audio_io_handle_t *input, 2010 audio_config_t *config, 2011 audio_devices_t devices, 2012 const String8& address, 2013 audio_source_t source, 2014 audio_input_flags_t flags) 2015{ 2016 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2017 if (inHwDev == NULL) { 2018 *input = AUDIO_IO_HANDLE_NONE; 2019 return 0; 2020 } 2021 2022 if (*input == AUDIO_IO_HANDLE_NONE) { 2023 *input = nextUniqueId(); 2024 } 2025 2026 audio_config_t halconfig = *config; 2027 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2028 audio_stream_in_t *inStream = NULL; 2029 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2030 &inStream, flags, address.string(), source); 2031 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2032 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2033 inStream, 2034 halconfig.sample_rate, 2035 halconfig.format, 2036 halconfig.channel_mask, 2037 flags, 2038 status, address.string()); 2039 2040 // If the input could not be opened with the requested parameters and we can handle the 2041 // conversion internally, try to open again with the proposed parameters. 2042 if (status == BAD_VALUE && 2043 audio_is_linear_pcm(config->format) && 2044 audio_is_linear_pcm(halconfig.format) && 2045 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2046 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2047 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2048 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2049 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2050 inStream = NULL; 2051 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2052 &inStream, flags, address.string(), source); 2053 // FIXME log this new status; HAL should not propose any further changes 2054 } 2055 2056 if (status == NO_ERROR && inStream != NULL) { 2057 2058#ifdef TEE_SINK 2059 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2060 // or (re-)create if current Pipe is idle and does not match the new format 2061 sp<NBAIO_Sink> teeSink; 2062 enum { 2063 TEE_SINK_NO, // don't copy input 2064 TEE_SINK_NEW, // copy input using a new pipe 2065 TEE_SINK_OLD, // copy input using an existing pipe 2066 } kind; 2067 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2068 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2069 if (!mTeeSinkInputEnabled) { 2070 kind = TEE_SINK_NO; 2071 } else if (!Format_isValid(format)) { 2072 kind = TEE_SINK_NO; 2073 } else if (mRecordTeeSink == 0) { 2074 kind = TEE_SINK_NEW; 2075 } else if (mRecordTeeSink->getStrongCount() != 1) { 2076 kind = TEE_SINK_NO; 2077 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2078 kind = TEE_SINK_OLD; 2079 } else { 2080 kind = TEE_SINK_NEW; 2081 } 2082 switch (kind) { 2083 case TEE_SINK_NEW: { 2084 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2085 size_t numCounterOffers = 0; 2086 const NBAIO_Format offers[1] = {format}; 2087 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2088 ALOG_ASSERT(index == 0); 2089 PipeReader *pipeReader = new PipeReader(*pipe); 2090 numCounterOffers = 0; 2091 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2092 ALOG_ASSERT(index == 0); 2093 mRecordTeeSink = pipe; 2094 mRecordTeeSource = pipeReader; 2095 teeSink = pipe; 2096 } 2097 break; 2098 case TEE_SINK_OLD: 2099 teeSink = mRecordTeeSink; 2100 break; 2101 case TEE_SINK_NO: 2102 default: 2103 break; 2104 } 2105#endif 2106 2107 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2108 2109 // Start record thread 2110 // RecordThread requires both input and output device indication to forward to audio 2111 // pre processing modules 2112 sp<RecordThread> thread = new RecordThread(this, 2113 inputStream, 2114 *input, 2115 primaryOutputDevice_l(), 2116 devices 2117#ifdef TEE_SINK 2118 , teeSink 2119#endif 2120 ); 2121 mRecordThreads.add(*input, thread); 2122 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2123 return thread; 2124 } 2125 2126 *input = AUDIO_IO_HANDLE_NONE; 2127 return 0; 2128} 2129 2130status_t AudioFlinger::closeInput(audio_io_handle_t input) 2131{ 2132 return closeInput_nonvirtual(input); 2133} 2134 2135status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2136{ 2137 // keep strong reference on the record thread so that 2138 // it is not destroyed while exit() is executed 2139 sp<RecordThread> thread; 2140 { 2141 Mutex::Autolock _l(mLock); 2142 thread = checkRecordThread_l(input); 2143 if (thread == 0) { 2144 return BAD_VALUE; 2145 } 2146 2147 ALOGV("closeInput() %d", input); 2148 2149 // If we still have effect chains, it means that a client still holds a handle 2150 // on at least one effect. We must either move the chain to an existing thread with the 2151 // same session ID or put it aside in case a new record thread is opened for a 2152 // new capture on the same session 2153 sp<EffectChain> chain; 2154 { 2155 Mutex::Autolock _sl(thread->mLock); 2156 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2157 // Note: maximum one chain per record thread 2158 if (effectChains.size() != 0) { 2159 chain = effectChains[0]; 2160 } 2161 } 2162 if (chain != 0) { 2163 // first check if a record thread is already opened with a client on the same session. 2164 // This should only happen in case of overlap between one thread tear down and the 2165 // creation of its replacement 2166 size_t i; 2167 for (i = 0; i < mRecordThreads.size(); i++) { 2168 sp<RecordThread> t = mRecordThreads.valueAt(i); 2169 if (t == thread) { 2170 continue; 2171 } 2172 if (t->hasAudioSession(chain->sessionId()) != 0) { 2173 Mutex::Autolock _l(t->mLock); 2174 ALOGV("closeInput() found thread %d for effect session %d", 2175 t->id(), chain->sessionId()); 2176 t->addEffectChain_l(chain); 2177 break; 2178 } 2179 } 2180 // put the chain aside if we could not find a record thread with the same session id. 2181 if (i == mRecordThreads.size()) { 2182 putOrphanEffectChain_l(chain); 2183 } 2184 } 2185 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2186 ioDesc->mIoHandle = input; 2187 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2188 mRecordThreads.removeItem(input); 2189 } 2190 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2191 // we have a different lock for notification client 2192 closeInputFinish(thread); 2193 return NO_ERROR; 2194} 2195 2196void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2197{ 2198 thread->exit(); 2199 AudioStreamIn *in = thread->clearInput(); 2200 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2201 // from now on thread->mInput is NULL 2202 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2203 delete in; 2204} 2205 2206void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2207{ 2208 mRecordThreads.removeItem(thread->mId); 2209 closeInputFinish(thread); 2210} 2211 2212status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2213{ 2214 Mutex::Autolock _l(mLock); 2215 ALOGV("invalidateStream() stream %d", stream); 2216 2217 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2218 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2219 thread->invalidateTracks(stream); 2220 } 2221 2222 return NO_ERROR; 2223} 2224 2225 2226audio_unique_id_t AudioFlinger::newAudioUniqueId() 2227{ 2228 return nextUniqueId(); 2229} 2230 2231void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2232{ 2233 Mutex::Autolock _l(mLock); 2234 pid_t caller = IPCThreadState::self()->getCallingPid(); 2235 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2236 if (pid != -1 && (caller == getpid_cached)) { 2237 caller = pid; 2238 } 2239 2240 { 2241 Mutex::Autolock _cl(mClientLock); 2242 // Ignore requests received from processes not known as notification client. The request 2243 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2244 // called from a different pid leaving a stale session reference. Also we don't know how 2245 // to clear this reference if the client process dies. 2246 if (mNotificationClients.indexOfKey(caller) < 0) { 2247 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2248 return; 2249 } 2250 } 2251 2252 size_t num = mAudioSessionRefs.size(); 2253 for (size_t i = 0; i< num; i++) { 2254 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2255 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2256 ref->mCnt++; 2257 ALOGV(" incremented refcount to %d", ref->mCnt); 2258 return; 2259 } 2260 } 2261 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2262 ALOGV(" added new entry for %d", audioSession); 2263} 2264 2265void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2266{ 2267 Mutex::Autolock _l(mLock); 2268 pid_t caller = IPCThreadState::self()->getCallingPid(); 2269 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2270 if (pid != -1 && (caller == getpid_cached)) { 2271 caller = pid; 2272 } 2273 size_t num = mAudioSessionRefs.size(); 2274 for (size_t i = 0; i< num; i++) { 2275 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2276 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2277 ref->mCnt--; 2278 ALOGV(" decremented refcount to %d", ref->mCnt); 2279 if (ref->mCnt == 0) { 2280 mAudioSessionRefs.removeAt(i); 2281 delete ref; 2282 purgeStaleEffects_l(); 2283 } 2284 return; 2285 } 2286 } 2287 // If the caller is mediaserver it is likely that the session being released was acquired 2288 // on behalf of a process not in notification clients and we ignore the warning. 2289 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2290} 2291 2292void AudioFlinger::purgeStaleEffects_l() { 2293 2294 ALOGV("purging stale effects"); 2295 2296 Vector< sp<EffectChain> > chains; 2297 2298 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2299 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2300 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2301 sp<EffectChain> ec = t->mEffectChains[j]; 2302 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2303 chains.push(ec); 2304 } 2305 } 2306 } 2307 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2308 sp<RecordThread> t = mRecordThreads.valueAt(i); 2309 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2310 sp<EffectChain> ec = t->mEffectChains[j]; 2311 chains.push(ec); 2312 } 2313 } 2314 2315 for (size_t i = 0; i < chains.size(); i++) { 2316 sp<EffectChain> ec = chains[i]; 2317 int sessionid = ec->sessionId(); 2318 sp<ThreadBase> t = ec->mThread.promote(); 2319 if (t == 0) { 2320 continue; 2321 } 2322 size_t numsessionrefs = mAudioSessionRefs.size(); 2323 bool found = false; 2324 for (size_t k = 0; k < numsessionrefs; k++) { 2325 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2326 if (ref->mSessionid == sessionid) { 2327 ALOGV(" session %d still exists for %d with %d refs", 2328 sessionid, ref->mPid, ref->mCnt); 2329 found = true; 2330 break; 2331 } 2332 } 2333 if (!found) { 2334 Mutex::Autolock _l(t->mLock); 2335 // remove all effects from the chain 2336 while (ec->mEffects.size()) { 2337 sp<EffectModule> effect = ec->mEffects[0]; 2338 effect->unPin(); 2339 t->removeEffect_l(effect); 2340 if (effect->purgeHandles()) { 2341 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2342 } 2343 AudioSystem::unregisterEffect(effect->id()); 2344 } 2345 } 2346 } 2347 return; 2348} 2349 2350// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2351AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2352{ 2353 return mPlaybackThreads.valueFor(output).get(); 2354} 2355 2356// checkMixerThread_l() must be called with AudioFlinger::mLock held 2357AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2358{ 2359 PlaybackThread *thread = checkPlaybackThread_l(output); 2360 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2361} 2362 2363// checkRecordThread_l() must be called with AudioFlinger::mLock held 2364AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2365{ 2366 return mRecordThreads.valueFor(input).get(); 2367} 2368 2369uint32_t AudioFlinger::nextUniqueId() 2370{ 2371 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2372} 2373 2374AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2375{ 2376 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2377 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2378 AudioStreamOut *output = thread->getOutput(); 2379 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2380 return thread; 2381 } 2382 } 2383 return NULL; 2384} 2385 2386audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2387{ 2388 PlaybackThread *thread = primaryPlaybackThread_l(); 2389 2390 if (thread == NULL) { 2391 return 0; 2392 } 2393 2394 return thread->outDevice(); 2395} 2396 2397sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2398 int triggerSession, 2399 int listenerSession, 2400 sync_event_callback_t callBack, 2401 wp<RefBase> cookie) 2402{ 2403 Mutex::Autolock _l(mLock); 2404 2405 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2406 status_t playStatus = NAME_NOT_FOUND; 2407 status_t recStatus = NAME_NOT_FOUND; 2408 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2409 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2410 if (playStatus == NO_ERROR) { 2411 return event; 2412 } 2413 } 2414 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2415 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2416 if (recStatus == NO_ERROR) { 2417 return event; 2418 } 2419 } 2420 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2421 mPendingSyncEvents.add(event); 2422 } else { 2423 ALOGV("createSyncEvent() invalid event %d", event->type()); 2424 event.clear(); 2425 } 2426 return event; 2427} 2428 2429// ---------------------------------------------------------------------------- 2430// Effect management 2431// ---------------------------------------------------------------------------- 2432 2433 2434status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2435{ 2436 Mutex::Autolock _l(mLock); 2437 return EffectQueryNumberEffects(numEffects); 2438} 2439 2440status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2441{ 2442 Mutex::Autolock _l(mLock); 2443 return EffectQueryEffect(index, descriptor); 2444} 2445 2446status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2447 effect_descriptor_t *descriptor) const 2448{ 2449 Mutex::Autolock _l(mLock); 2450 return EffectGetDescriptor(pUuid, descriptor); 2451} 2452 2453 2454sp<IEffect> AudioFlinger::createEffect( 2455 effect_descriptor_t *pDesc, 2456 const sp<IEffectClient>& effectClient, 2457 int32_t priority, 2458 audio_io_handle_t io, 2459 int sessionId, 2460 const String16& opPackageName, 2461 status_t *status, 2462 int *id, 2463 int *enabled) 2464{ 2465 status_t lStatus = NO_ERROR; 2466 sp<EffectHandle> handle; 2467 effect_descriptor_t desc; 2468 2469 pid_t pid = IPCThreadState::self()->getCallingPid(); 2470 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2471 pid, effectClient.get(), priority, sessionId, io); 2472 2473 if (pDesc == NULL) { 2474 lStatus = BAD_VALUE; 2475 goto Exit; 2476 } 2477 2478 // check audio settings permission for global effects 2479 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2480 lStatus = PERMISSION_DENIED; 2481 goto Exit; 2482 } 2483 2484 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2485 // that can only be created by audio policy manager (running in same process) 2486 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2487 lStatus = PERMISSION_DENIED; 2488 goto Exit; 2489 } 2490 2491 { 2492 if (!EffectIsNullUuid(&pDesc->uuid)) { 2493 // if uuid is specified, request effect descriptor 2494 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2495 if (lStatus < 0) { 2496 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2497 goto Exit; 2498 } 2499 } else { 2500 // if uuid is not specified, look for an available implementation 2501 // of the required type in effect factory 2502 if (EffectIsNullUuid(&pDesc->type)) { 2503 ALOGW("createEffect() no effect type"); 2504 lStatus = BAD_VALUE; 2505 goto Exit; 2506 } 2507 uint32_t numEffects = 0; 2508 effect_descriptor_t d; 2509 d.flags = 0; // prevent compiler warning 2510 bool found = false; 2511 2512 lStatus = EffectQueryNumberEffects(&numEffects); 2513 if (lStatus < 0) { 2514 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2515 goto Exit; 2516 } 2517 for (uint32_t i = 0; i < numEffects; i++) { 2518 lStatus = EffectQueryEffect(i, &desc); 2519 if (lStatus < 0) { 2520 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2521 continue; 2522 } 2523 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2524 // If matching type found save effect descriptor. If the session is 2525 // 0 and the effect is not auxiliary, continue enumeration in case 2526 // an auxiliary version of this effect type is available 2527 found = true; 2528 d = desc; 2529 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2530 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2531 break; 2532 } 2533 } 2534 } 2535 if (!found) { 2536 lStatus = BAD_VALUE; 2537 ALOGW("createEffect() effect not found"); 2538 goto Exit; 2539 } 2540 // For same effect type, chose auxiliary version over insert version if 2541 // connect to output mix (Compliance to OpenSL ES) 2542 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2543 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2544 desc = d; 2545 } 2546 } 2547 2548 // Do not allow auxiliary effects on a session different from 0 (output mix) 2549 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2550 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2551 lStatus = INVALID_OPERATION; 2552 goto Exit; 2553 } 2554 2555 // check recording permission for visualizer 2556 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2557 !recordingAllowed(opPackageName)) { 2558 lStatus = PERMISSION_DENIED; 2559 goto Exit; 2560 } 2561 2562 // return effect descriptor 2563 *pDesc = desc; 2564 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2565 // if the output returned by getOutputForEffect() is removed before we lock the 2566 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2567 // and we will exit safely 2568 io = AudioSystem::getOutputForEffect(&desc); 2569 ALOGV("createEffect got output %d", io); 2570 } 2571 2572 Mutex::Autolock _l(mLock); 2573 2574 // If output is not specified try to find a matching audio session ID in one of the 2575 // output threads. 2576 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2577 // because of code checking output when entering the function. 2578 // Note: io is never 0 when creating an effect on an input 2579 if (io == AUDIO_IO_HANDLE_NONE) { 2580 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2581 // output must be specified by AudioPolicyManager when using session 2582 // AUDIO_SESSION_OUTPUT_STAGE 2583 lStatus = BAD_VALUE; 2584 goto Exit; 2585 } 2586 // look for the thread where the specified audio session is present 2587 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2588 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2589 io = mPlaybackThreads.keyAt(i); 2590 break; 2591 } 2592 } 2593 if (io == 0) { 2594 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2595 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2596 io = mRecordThreads.keyAt(i); 2597 break; 2598 } 2599 } 2600 } 2601 // If no output thread contains the requested session ID, default to 2602 // first output. The effect chain will be moved to the correct output 2603 // thread when a track with the same session ID is created 2604 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2605 io = mPlaybackThreads.keyAt(0); 2606 } 2607 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2608 } 2609 ThreadBase *thread = checkRecordThread_l(io); 2610 if (thread == NULL) { 2611 thread = checkPlaybackThread_l(io); 2612 if (thread == NULL) { 2613 ALOGE("createEffect() unknown output thread"); 2614 lStatus = BAD_VALUE; 2615 goto Exit; 2616 } 2617 } else { 2618 // Check if one effect chain was awaiting for an effect to be created on this 2619 // session and used it instead of creating a new one. 2620 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2621 if (chain != 0) { 2622 Mutex::Autolock _l(thread->mLock); 2623 thread->addEffectChain_l(chain); 2624 } 2625 } 2626 2627 sp<Client> client = registerPid(pid); 2628 2629 // create effect on selected output thread 2630 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2631 &desc, enabled, &lStatus); 2632 if (handle != 0 && id != NULL) { 2633 *id = handle->id(); 2634 } 2635 if (handle == 0) { 2636 // remove local strong reference to Client with mClientLock held 2637 Mutex::Autolock _cl(mClientLock); 2638 client.clear(); 2639 } 2640 } 2641 2642Exit: 2643 *status = lStatus; 2644 return handle; 2645} 2646 2647status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2648 audio_io_handle_t dstOutput) 2649{ 2650 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2651 sessionId, srcOutput, dstOutput); 2652 Mutex::Autolock _l(mLock); 2653 if (srcOutput == dstOutput) { 2654 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2655 return NO_ERROR; 2656 } 2657 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2658 if (srcThread == NULL) { 2659 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2660 return BAD_VALUE; 2661 } 2662 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2663 if (dstThread == NULL) { 2664 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2665 return BAD_VALUE; 2666 } 2667 2668 Mutex::Autolock _dl(dstThread->mLock); 2669 Mutex::Autolock _sl(srcThread->mLock); 2670 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2671} 2672 2673// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2674status_t AudioFlinger::moveEffectChain_l(int sessionId, 2675 AudioFlinger::PlaybackThread *srcThread, 2676 AudioFlinger::PlaybackThread *dstThread, 2677 bool reRegister) 2678{ 2679 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2680 sessionId, srcThread, dstThread); 2681 2682 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2683 if (chain == 0) { 2684 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2685 sessionId, srcThread); 2686 return INVALID_OPERATION; 2687 } 2688 2689 // Check whether the destination thread has a channel count of FCC_2, which is 2690 // currently required for (most) effects. Prevent moving the effect chain here rather 2691 // than disabling the addEffect_l() call in dstThread below. 2692 if ((dstThread->type() == ThreadBase::MIXER || dstThread->type() == ThreadBase::DUPLICATING) && 2693 dstThread->mChannelCount != FCC_2) { 2694 ALOGW("moveEffectChain_l() effect chain failed because" 2695 " destination thread %p channel count(%u) != %u", 2696 dstThread, dstThread->mChannelCount, FCC_2); 2697 return INVALID_OPERATION; 2698 } 2699 2700 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2701 // so that a new chain is created with correct parameters when first effect is added. This is 2702 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2703 // removed. 2704 srcThread->removeEffectChain_l(chain); 2705 2706 // transfer all effects one by one so that new effect chain is created on new thread with 2707 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2708 sp<EffectChain> dstChain; 2709 uint32_t strategy = 0; // prevent compiler warning 2710 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2711 Vector< sp<EffectModule> > removed; 2712 status_t status = NO_ERROR; 2713 while (effect != 0) { 2714 srcThread->removeEffect_l(effect); 2715 removed.add(effect); 2716 status = dstThread->addEffect_l(effect); 2717 if (status != NO_ERROR) { 2718 break; 2719 } 2720 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2721 if (effect->state() == EffectModule::ACTIVE || 2722 effect->state() == EffectModule::STOPPING) { 2723 effect->start(); 2724 } 2725 // if the move request is not received from audio policy manager, the effect must be 2726 // re-registered with the new strategy and output 2727 if (dstChain == 0) { 2728 dstChain = effect->chain().promote(); 2729 if (dstChain == 0) { 2730 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2731 status = NO_INIT; 2732 break; 2733 } 2734 strategy = dstChain->strategy(); 2735 } 2736 if (reRegister) { 2737 AudioSystem::unregisterEffect(effect->id()); 2738 AudioSystem::registerEffect(&effect->desc(), 2739 dstThread->id(), 2740 strategy, 2741 sessionId, 2742 effect->id()); 2743 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2744 } 2745 effect = chain->getEffectFromId_l(0); 2746 } 2747 2748 if (status != NO_ERROR) { 2749 for (size_t i = 0; i < removed.size(); i++) { 2750 srcThread->addEffect_l(removed[i]); 2751 if (dstChain != 0 && reRegister) { 2752 AudioSystem::unregisterEffect(removed[i]->id()); 2753 AudioSystem::registerEffect(&removed[i]->desc(), 2754 srcThread->id(), 2755 strategy, 2756 sessionId, 2757 removed[i]->id()); 2758 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2759 } 2760 } 2761 } 2762 2763 return status; 2764} 2765 2766bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2767{ 2768 if (mGlobalEffectEnableTime != 0 && 2769 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2770 return true; 2771 } 2772 2773 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2774 sp<EffectChain> ec = 2775 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2776 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2777 return true; 2778 } 2779 } 2780 return false; 2781} 2782 2783void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2784{ 2785 Mutex::Autolock _l(mLock); 2786 2787 mGlobalEffectEnableTime = systemTime(); 2788 2789 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2790 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2791 if (t->mType == ThreadBase::OFFLOAD) { 2792 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2793 } 2794 } 2795 2796} 2797 2798status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2799{ 2800 audio_session_t session = (audio_session_t)chain->sessionId(); 2801 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2802 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2803 if (index >= 0) { 2804 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2805 return ALREADY_EXISTS; 2806 } 2807 mOrphanEffectChains.add(session, chain); 2808 return NO_ERROR; 2809} 2810 2811sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2812{ 2813 sp<EffectChain> chain; 2814 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2815 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2816 if (index >= 0) { 2817 chain = mOrphanEffectChains.valueAt(index); 2818 mOrphanEffectChains.removeItemsAt(index); 2819 } 2820 return chain; 2821} 2822 2823bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2824{ 2825 Mutex::Autolock _l(mLock); 2826 audio_session_t session = (audio_session_t)effect->sessionId(); 2827 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2828 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2829 if (index >= 0) { 2830 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2831 if (chain->removeEffect_l(effect) == 0) { 2832 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2833 mOrphanEffectChains.removeItemsAt(index); 2834 } 2835 return true; 2836 } 2837 return false; 2838} 2839 2840 2841struct Entry { 2842#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2843 char mFileName[TEE_MAX_FILENAME]; 2844}; 2845 2846int comparEntry(const void *p1, const void *p2) 2847{ 2848 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2849} 2850 2851#ifdef TEE_SINK 2852void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2853{ 2854 NBAIO_Source *teeSource = source.get(); 2855 if (teeSource != NULL) { 2856 // .wav rotation 2857 // There is a benign race condition if 2 threads call this simultaneously. 2858 // They would both traverse the directory, but the result would simply be 2859 // failures at unlink() which are ignored. It's also unlikely since 2860 // normally dumpsys is only done by bugreport or from the command line. 2861 char teePath[32+256]; 2862 strcpy(teePath, "/data/misc/media"); 2863 size_t teePathLen = strlen(teePath); 2864 DIR *dir = opendir(teePath); 2865 teePath[teePathLen++] = '/'; 2866 if (dir != NULL) { 2867#define TEE_MAX_SORT 20 // number of entries to sort 2868#define TEE_MAX_KEEP 10 // number of entries to keep 2869 struct Entry entries[TEE_MAX_SORT]; 2870 size_t entryCount = 0; 2871 while (entryCount < TEE_MAX_SORT) { 2872 struct dirent de; 2873 struct dirent *result = NULL; 2874 int rc = readdir_r(dir, &de, &result); 2875 if (rc != 0) { 2876 ALOGW("readdir_r failed %d", rc); 2877 break; 2878 } 2879 if (result == NULL) { 2880 break; 2881 } 2882 if (result != &de) { 2883 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2884 break; 2885 } 2886 // ignore non .wav file entries 2887 size_t nameLen = strlen(de.d_name); 2888 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2889 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2890 continue; 2891 } 2892 strcpy(entries[entryCount++].mFileName, de.d_name); 2893 } 2894 (void) closedir(dir); 2895 if (entryCount > TEE_MAX_KEEP) { 2896 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2897 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 2898 strcpy(&teePath[teePathLen], entries[i].mFileName); 2899 (void) unlink(teePath); 2900 } 2901 } 2902 } else { 2903 if (fd >= 0) { 2904 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2905 } 2906 } 2907 char teeTime[16]; 2908 struct timeval tv; 2909 gettimeofday(&tv, NULL); 2910 struct tm tm; 2911 localtime_r(&tv.tv_sec, &tm); 2912 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2913 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2914 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2915 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2916 if (teeFd >= 0) { 2917 // FIXME use libsndfile 2918 char wavHeader[44]; 2919 memcpy(wavHeader, 2920 "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0", 2921 sizeof(wavHeader)); 2922 NBAIO_Format format = teeSource->format(); 2923 unsigned channelCount = Format_channelCount(format); 2924 uint32_t sampleRate = Format_sampleRate(format); 2925 size_t frameSize = Format_frameSize(format); 2926 wavHeader[22] = channelCount; // number of channels 2927 wavHeader[24] = sampleRate; // sample rate 2928 wavHeader[25] = sampleRate >> 8; 2929 wavHeader[32] = frameSize; // block alignment 2930 wavHeader[33] = frameSize >> 8; 2931 write(teeFd, wavHeader, sizeof(wavHeader)); 2932 size_t total = 0; 2933 bool firstRead = true; 2934#define TEE_SINK_READ 1024 // frames per I/O operation 2935 void *buffer = malloc(TEE_SINK_READ * frameSize); 2936 for (;;) { 2937 size_t count = TEE_SINK_READ; 2938 ssize_t actual = teeSource->read(buffer, count, 2939 AudioBufferProvider::kInvalidPTS); 2940 bool wasFirstRead = firstRead; 2941 firstRead = false; 2942 if (actual <= 0) { 2943 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2944 continue; 2945 } 2946 break; 2947 } 2948 ALOG_ASSERT(actual <= (ssize_t)count); 2949 write(teeFd, buffer, actual * frameSize); 2950 total += actual; 2951 } 2952 free(buffer); 2953 lseek(teeFd, (off_t) 4, SEEK_SET); 2954 uint32_t temp = 44 + total * frameSize - 8; 2955 // FIXME not big-endian safe 2956 write(teeFd, &temp, sizeof(temp)); 2957 lseek(teeFd, (off_t) 40, SEEK_SET); 2958 temp = total * frameSize; 2959 // FIXME not big-endian safe 2960 write(teeFd, &temp, sizeof(temp)); 2961 close(teeFd); 2962 if (fd >= 0) { 2963 dprintf(fd, "tee copied to %s\n", teePath); 2964 } 2965 } else { 2966 if (fd >= 0) { 2967 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2968 } 2969 } 2970 } 2971} 2972#endif 2973 2974// ---------------------------------------------------------------------------- 2975 2976status_t AudioFlinger::onTransact( 2977 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2978{ 2979 return BnAudioFlinger::onTransact(code, data, reply, flags); 2980} 2981 2982} // namespace android 2983