AudioFlinger.cpp revision 9ee0540d3a61bff03d561ca431a371c3d9335d2b
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 <media/audiohal/DeviceHalInterface.h> 35#include <media/audiohal/DevicesFactoryHalInterface.h> 36#include <media/audiohal/EffectsFactoryHalInterface.h> 37#include <media/AudioParameter.h> 38#include <memunreachable/memunreachable.h> 39#include <utils/String16.h> 40#include <utils/threads.h> 41#include <utils/Atomic.h> 42 43#include <cutils/bitops.h> 44#include <cutils/properties.h> 45 46#include <system/audio.h> 47 48#include "AudioMixer.h" 49#include "AudioFlinger.h" 50#include "ServiceUtilities.h" 51 52#include <media/AudioResamplerPublic.h> 53 54#include <system/audio_effects/effect_visualizer.h> 55#include <system/audio_effects/effect_ns.h> 56#include <system/audio_effects/effect_aec.h> 57 58#include <audio_utils/primitives.h> 59 60#include <powermanager/PowerManager.h> 61 62#include <media/IMediaLogService.h> 63#include <media/MemoryLeakTrackUtil.h> 64#include <media/nbaio/Pipe.h> 65#include <media/nbaio/PipeReader.h> 66#include <media/AudioParameter.h> 67#include <mediautils/BatteryNotifier.h> 68#include <private/android_filesystem_config.h> 69 70// ---------------------------------------------------------------------------- 71 72// Note: the following macro is used for extremely verbose logging message. In 73// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 74// 0; but one side effect of this is to turn all LOGV's as well. Some messages 75// are so verbose that we want to suppress them even when we have ALOG_ASSERT 76// turned on. Do not uncomment the #def below unless you really know what you 77// are doing and want to see all of the extremely verbose messages. 78//#define VERY_VERY_VERBOSE_LOGGING 79#ifdef VERY_VERY_VERBOSE_LOGGING 80#define ALOGVV ALOGV 81#else 82#define ALOGVV(a...) do { } while(0) 83#endif 84 85namespace android { 86 87static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 88static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 89static const char kClientLockedString[] = "Client lock is taken\n"; 90static const char kNoEffectsFactory[] = "Effects Factory is absent\n"; 91 92 93nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 94 95uint32_t AudioFlinger::mScreenState; 96 97#ifdef TEE_SINK 98bool AudioFlinger::mTeeSinkInputEnabled = false; 99bool AudioFlinger::mTeeSinkOutputEnabled = false; 100bool AudioFlinger::mTeeSinkTrackEnabled = false; 101 102size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 103size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 104size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 105#endif 106 107// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 108// we define a minimum time during which a global effect is considered enabled. 109static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 110 111// ---------------------------------------------------------------------------- 112 113const char *formatToString(audio_format_t format) { 114 switch (audio_get_main_format(format)) { 115 case AUDIO_FORMAT_PCM: 116 switch (format) { 117 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 118 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 119 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 120 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 121 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 122 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 123 default: 124 break; 125 } 126 break; 127 case AUDIO_FORMAT_MP3: return "mp3"; 128 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 129 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 130 case AUDIO_FORMAT_AAC: return "aac"; 131 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 132 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 133 case AUDIO_FORMAT_VORBIS: return "vorbis"; 134 case AUDIO_FORMAT_OPUS: return "opus"; 135 case AUDIO_FORMAT_AC3: return "ac-3"; 136 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 137 case AUDIO_FORMAT_IEC61937: return "iec61937"; 138 case AUDIO_FORMAT_DTS: return "dts"; 139 case AUDIO_FORMAT_DTS_HD: return "dts-hd"; 140 case AUDIO_FORMAT_DOLBY_TRUEHD: return "dolby-truehd"; 141 default: 142 break; 143 } 144 return "unknown"; 145} 146 147// ---------------------------------------------------------------------------- 148 149AudioFlinger::AudioFlinger() 150 : BnAudioFlinger(), 151 mPrimaryHardwareDev(NULL), 152 mAudioHwDevs(NULL), 153 mHardwareStatus(AUDIO_HW_IDLE), 154 mMasterVolume(1.0f), 155 mMasterMute(false), 156 // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), 157 mMode(AUDIO_MODE_INVALID), 158 mBtNrecIsOff(false), 159 mIsLowRamDevice(true), 160 mIsDeviceTypeKnown(false), 161 mGlobalEffectEnableTime(0), 162 mSystemReady(false) 163{ 164 // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum 165 for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) { 166 // zero ID has a special meaning, so unavailable 167 mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX; 168 } 169 170 getpid_cached = getpid(); 171 const bool doLog = property_get_bool("ro.test_harness", false); 172 if (doLog) { 173 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 174 MemoryHeapBase::READ_ONLY); 175 } 176 177 // reset battery stats. 178 // if the audio service has crashed, battery stats could be left 179 // in bad state, reset the state upon service start. 180 BatteryNotifier::getInstance().noteResetAudio(); 181 182 mDevicesFactoryHal = DevicesFactoryHalInterface::create(); 183 mEffectsFactoryHal = EffectsFactoryHalInterface::create(); 184 185#ifdef TEE_SINK 186 char value[PROPERTY_VALUE_MAX]; 187 (void) property_get("ro.debuggable", value, "0"); 188 int debuggable = atoi(value); 189 int teeEnabled = 0; 190 if (debuggable) { 191 (void) property_get("af.tee", value, "0"); 192 teeEnabled = atoi(value); 193 } 194 // FIXME symbolic constants here 195 if (teeEnabled & 1) { 196 mTeeSinkInputEnabled = true; 197 } 198 if (teeEnabled & 2) { 199 mTeeSinkOutputEnabled = true; 200 } 201 if (teeEnabled & 4) { 202 mTeeSinkTrackEnabled = true; 203 } 204#endif 205} 206 207void AudioFlinger::onFirstRef() 208{ 209 Mutex::Autolock _l(mLock); 210 211 /* TODO: move all this work into an Init() function */ 212 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 213 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 214 uint32_t int_val; 215 if (1 == sscanf(val_str, "%u", &int_val)) { 216 mStandbyTimeInNsecs = milliseconds(int_val); 217 ALOGI("Using %u mSec as standby time.", int_val); 218 } else { 219 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 220 ALOGI("Using default %u mSec as standby time.", 221 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 222 } 223 } 224 225 mPatchPanel = new PatchPanel(this); 226 227 mMode = AUDIO_MODE_NORMAL; 228} 229 230AudioFlinger::~AudioFlinger() 231{ 232 while (!mRecordThreads.isEmpty()) { 233 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 234 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 235 } 236 while (!mPlaybackThreads.isEmpty()) { 237 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 238 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 239 } 240 241 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 242 // no mHardwareLock needed, as there are no other references to this 243 delete mAudioHwDevs.valueAt(i); 244 } 245 246 // Tell media.log service about any old writers that still need to be unregistered 247 if (mLogMemoryDealer != 0) { 248 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 249 if (binder != 0) { 250 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 251 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 252 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 253 mUnregisteredWriters.pop(); 254 mediaLogService->unregisterWriter(iMemory); 255 } 256 } 257 } 258} 259 260static const char * const audio_interfaces[] = { 261 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 262 AUDIO_HARDWARE_MODULE_ID_A2DP, 263 AUDIO_HARDWARE_MODULE_ID_USB, 264}; 265#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 266 267AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 268 audio_module_handle_t module, 269 audio_devices_t devices) 270{ 271 // if module is 0, the request comes from an old policy manager and we should load 272 // well known modules 273 if (module == 0) { 274 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 275 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 276 loadHwModule_l(audio_interfaces[i]); 277 } 278 // then try to find a module supporting the requested device. 279 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 280 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 281 sp<DeviceHalInterface> dev = audioHwDevice->hwDevice(); 282 uint32_t supportedDevices; 283 if (dev->getSupportedDevices(&supportedDevices) == OK && 284 (supportedDevices & devices) == devices) { 285 return audioHwDevice; 286 } 287 } 288 } else { 289 // check a match for the requested module handle 290 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 291 if (audioHwDevice != NULL) { 292 return audioHwDevice; 293 } 294 } 295 296 return NULL; 297} 298 299void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 300{ 301 const size_t SIZE = 256; 302 char buffer[SIZE]; 303 String8 result; 304 305 result.append("Clients:\n"); 306 for (size_t i = 0; i < mClients.size(); ++i) { 307 sp<Client> client = mClients.valueAt(i).promote(); 308 if (client != 0) { 309 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 310 result.append(buffer); 311 } 312 } 313 314 result.append("Notification Clients:\n"); 315 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 316 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 317 result.append(buffer); 318 } 319 320 result.append("Global session refs:\n"); 321 result.append(" session pid count\n"); 322 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 323 AudioSessionRef *r = mAudioSessionRefs[i]; 324 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 325 result.append(buffer); 326 } 327 write(fd, result.string(), result.size()); 328} 329 330 331void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 332{ 333 const size_t SIZE = 256; 334 char buffer[SIZE]; 335 String8 result; 336 hardware_call_state hardwareStatus = mHardwareStatus; 337 338 snprintf(buffer, SIZE, "Hardware status: %d\n" 339 "Standby Time mSec: %u\n", 340 hardwareStatus, 341 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 342 result.append(buffer); 343 write(fd, result.string(), result.size()); 344} 345 346void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 347{ 348 const size_t SIZE = 256; 349 char buffer[SIZE]; 350 String8 result; 351 snprintf(buffer, SIZE, "Permission Denial: " 352 "can't dump AudioFlinger from pid=%d, uid=%d\n", 353 IPCThreadState::self()->getCallingPid(), 354 IPCThreadState::self()->getCallingUid()); 355 result.append(buffer); 356 write(fd, result.string(), result.size()); 357} 358 359bool AudioFlinger::dumpTryLock(Mutex& mutex) 360{ 361 bool locked = false; 362 for (int i = 0; i < kDumpLockRetries; ++i) { 363 if (mutex.tryLock() == NO_ERROR) { 364 locked = true; 365 break; 366 } 367 usleep(kDumpLockSleepUs); 368 } 369 return locked; 370} 371 372status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 373{ 374 if (!dumpAllowed()) { 375 dumpPermissionDenial(fd, args); 376 } else { 377 // get state of hardware lock 378 bool hardwareLocked = dumpTryLock(mHardwareLock); 379 if (!hardwareLocked) { 380 String8 result(kHardwareLockedString); 381 write(fd, result.string(), result.size()); 382 } else { 383 mHardwareLock.unlock(); 384 } 385 386 bool locked = dumpTryLock(mLock); 387 388 // failed to lock - AudioFlinger is probably deadlocked 389 if (!locked) { 390 String8 result(kDeadlockedString); 391 write(fd, result.string(), result.size()); 392 } 393 394 bool clientLocked = dumpTryLock(mClientLock); 395 if (!clientLocked) { 396 String8 result(kClientLockedString); 397 write(fd, result.string(), result.size()); 398 } 399 400 if (mEffectsFactoryHal != 0) { 401 mEffectsFactoryHal->dumpEffects(fd); 402 } else { 403 String8 result(kNoEffectsFactory); 404 write(fd, result.string(), result.size()); 405 } 406 407 dumpClients(fd, args); 408 if (clientLocked) { 409 mClientLock.unlock(); 410 } 411 412 dumpInternals(fd, args); 413 414 // dump playback threads 415 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 416 mPlaybackThreads.valueAt(i)->dump(fd, args); 417 } 418 419 // dump record threads 420 for (size_t i = 0; i < mRecordThreads.size(); i++) { 421 mRecordThreads.valueAt(i)->dump(fd, args); 422 } 423 424 // dump orphan effect chains 425 if (mOrphanEffectChains.size() != 0) { 426 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 427 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 428 mOrphanEffectChains.valueAt(i)->dump(fd, args); 429 } 430 } 431 // dump all hardware devs 432 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 433 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 434 dev->dump(fd); 435 } 436 437#ifdef TEE_SINK 438 // dump the serially shared record tee sink 439 if (mRecordTeeSource != 0) { 440 dumpTee(fd, mRecordTeeSource); 441 } 442#endif 443 444 if (locked) { 445 mLock.unlock(); 446 } 447 448 // append a copy of media.log here by forwarding fd to it, but don't attempt 449 // to lookup the service if it's not running, as it will block for a second 450 if (mLogMemoryDealer != 0) { 451 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 452 if (binder != 0) { 453 dprintf(fd, "\nmedia.log:\n"); 454 Vector<String16> args; 455 binder->dump(fd, args); 456 } 457 } 458 459 // check for optional arguments 460 bool dumpMem = false; 461 bool unreachableMemory = false; 462 for (const auto &arg : args) { 463 if (arg == String16("-m")) { 464 dumpMem = true; 465 } else if (arg == String16("--unreachable")) { 466 unreachableMemory = true; 467 } 468 } 469 470 if (dumpMem) { 471 dprintf(fd, "\nDumping memory:\n"); 472 std::string s = dumpMemoryAddresses(100 /* limit */); 473 write(fd, s.c_str(), s.size()); 474 } 475 if (unreachableMemory) { 476 dprintf(fd, "\nDumping unreachable memory:\n"); 477 // TODO - should limit be an argument parameter? 478 std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */); 479 write(fd, s.c_str(), s.size()); 480 } 481 } 482 return NO_ERROR; 483} 484 485sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 486{ 487 Mutex::Autolock _cl(mClientLock); 488 // If pid is already in the mClients wp<> map, then use that entry 489 // (for which promote() is always != 0), otherwise create a new entry and Client. 490 sp<Client> client = mClients.valueFor(pid).promote(); 491 if (client == 0) { 492 client = new Client(this, pid); 493 mClients.add(pid, client); 494 } 495 496 return client; 497} 498 499sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 500{ 501 // If there is no memory allocated for logs, return a dummy writer that does nothing 502 if (mLogMemoryDealer == 0) { 503 return new NBLog::Writer(); 504 } 505 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 506 // Similarly if we can't contact the media.log service, also return a dummy writer 507 if (binder == 0) { 508 return new NBLog::Writer(); 509 } 510 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 511 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 512 // If allocation fails, consult the vector of previously unregistered writers 513 // and garbage-collect one or more them until an allocation succeeds 514 if (shared == 0) { 515 Mutex::Autolock _l(mUnregisteredWritersLock); 516 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 517 { 518 // Pick the oldest stale writer to garbage-collect 519 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 520 mUnregisteredWriters.removeAt(0); 521 mediaLogService->unregisterWriter(iMemory); 522 // Now the media.log remote reference to IMemory is gone. When our last local 523 // reference to IMemory also drops to zero at end of this block, 524 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 525 } 526 // Re-attempt the allocation 527 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 528 if (shared != 0) { 529 goto success; 530 } 531 } 532 // Even after garbage-collecting all old writers, there is still not enough memory, 533 // so return a dummy writer 534 return new NBLog::Writer(); 535 } 536success: 537 mediaLogService->registerWriter(shared, size, name); 538 return new NBLog::Writer(size, shared); 539} 540 541void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 542{ 543 if (writer == 0) { 544 return; 545 } 546 sp<IMemory> iMemory(writer->getIMemory()); 547 if (iMemory == 0) { 548 return; 549 } 550 // Rather than removing the writer immediately, append it to a queue of old writers to 551 // be garbage-collected later. This allows us to continue to view old logs for a while. 552 Mutex::Autolock _l(mUnregisteredWritersLock); 553 mUnregisteredWriters.push(writer); 554} 555 556// IAudioFlinger interface 557 558 559sp<IAudioTrack> AudioFlinger::createTrack( 560 audio_stream_type_t streamType, 561 uint32_t sampleRate, 562 audio_format_t format, 563 audio_channel_mask_t channelMask, 564 size_t *frameCount, 565 audio_output_flags_t *flags, 566 const sp<IMemory>& sharedBuffer, 567 audio_io_handle_t output, 568 pid_t pid, 569 pid_t tid, 570 audio_session_t *sessionId, 571 int clientUid, 572 status_t *status) 573{ 574 sp<PlaybackThread::Track> track; 575 sp<TrackHandle> trackHandle; 576 sp<Client> client; 577 status_t lStatus; 578 audio_session_t lSessionId; 579 580 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 581 if (pid == -1 || !isTrustedCallingUid(callingUid)) { 582 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 583 ALOGW_IF(pid != -1 && pid != callingPid, 584 "%s uid %d pid %d tried to pass itself off as pid %d", 585 __func__, callingUid, callingPid, pid); 586 pid = callingPid; 587 } 588 589 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 590 // but if someone uses binder directly they could bypass that and cause us to crash 591 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 592 ALOGE("createTrack() invalid stream type %d", streamType); 593 lStatus = BAD_VALUE; 594 goto Exit; 595 } 596 597 // further sample rate checks are performed by createTrack_l() depending on the thread type 598 if (sampleRate == 0) { 599 ALOGE("createTrack() invalid sample rate %u", sampleRate); 600 lStatus = BAD_VALUE; 601 goto Exit; 602 } 603 604 // further channel mask checks are performed by createTrack_l() depending on the thread type 605 if (!audio_is_output_channel(channelMask)) { 606 ALOGE("createTrack() invalid channel mask %#x", channelMask); 607 lStatus = BAD_VALUE; 608 goto Exit; 609 } 610 611 // further format checks are performed by createTrack_l() depending on the thread type 612 if (!audio_is_valid_format(format)) { 613 ALOGE("createTrack() invalid format %#x", format); 614 lStatus = BAD_VALUE; 615 goto Exit; 616 } 617 618 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 619 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 620 lStatus = BAD_VALUE; 621 goto Exit; 622 } 623 624 { 625 Mutex::Autolock _l(mLock); 626 PlaybackThread *thread = checkPlaybackThread_l(output); 627 if (thread == NULL) { 628 ALOGE("no playback thread found for output handle %d", output); 629 lStatus = BAD_VALUE; 630 goto Exit; 631 } 632 633 client = registerPid(pid); 634 635 PlaybackThread *effectThread = NULL; 636 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 637 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 638 ALOGE("createTrack() invalid session ID %d", *sessionId); 639 lStatus = BAD_VALUE; 640 goto Exit; 641 } 642 lSessionId = *sessionId; 643 // check if an effect chain with the same session ID is present on another 644 // output thread and move it here. 645 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 646 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 647 if (mPlaybackThreads.keyAt(i) != output) { 648 uint32_t sessions = t->hasAudioSession(lSessionId); 649 if (sessions & ThreadBase::EFFECT_SESSION) { 650 effectThread = t.get(); 651 break; 652 } 653 } 654 } 655 } else { 656 // if no audio session id is provided, create one here 657 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 658 if (sessionId != NULL) { 659 *sessionId = lSessionId; 660 } 661 } 662 ALOGV("createTrack() lSessionId: %d", lSessionId); 663 664 track = thread->createTrack_l(client, streamType, sampleRate, format, 665 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 666 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 667 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 668 669 // move effect chain to this output thread if an effect on same session was waiting 670 // for a track to be created 671 if (lStatus == NO_ERROR && effectThread != NULL) { 672 // no risk of deadlock because AudioFlinger::mLock is held 673 Mutex::Autolock _dl(thread->mLock); 674 Mutex::Autolock _sl(effectThread->mLock); 675 moveEffectChain_l(lSessionId, effectThread, thread, true); 676 } 677 678 // Look for sync events awaiting for a session to be used. 679 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 680 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 681 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 682 if (lStatus == NO_ERROR) { 683 (void) track->setSyncEvent(mPendingSyncEvents[i]); 684 } else { 685 mPendingSyncEvents[i]->cancel(); 686 } 687 mPendingSyncEvents.removeAt(i); 688 i--; 689 } 690 } 691 } 692 693 setAudioHwSyncForSession_l(thread, lSessionId); 694 } 695 696 if (lStatus != NO_ERROR) { 697 // remove local strong reference to Client before deleting the Track so that the 698 // Client destructor is called by the TrackBase destructor with mClientLock held 699 // Don't hold mClientLock when releasing the reference on the track as the 700 // destructor will acquire it. 701 { 702 Mutex::Autolock _cl(mClientLock); 703 client.clear(); 704 } 705 track.clear(); 706 goto Exit; 707 } 708 709 // return handle to client 710 trackHandle = new TrackHandle(track); 711 712Exit: 713 *status = lStatus; 714 return trackHandle; 715} 716 717uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 718{ 719 Mutex::Autolock _l(mLock); 720 ThreadBase *thread = checkThread_l(ioHandle); 721 if (thread == NULL) { 722 ALOGW("sampleRate() unknown thread %d", ioHandle); 723 return 0; 724 } 725 return thread->sampleRate(); 726} 727 728audio_format_t AudioFlinger::format(audio_io_handle_t output) const 729{ 730 Mutex::Autolock _l(mLock); 731 PlaybackThread *thread = checkPlaybackThread_l(output); 732 if (thread == NULL) { 733 ALOGW("format() unknown thread %d", output); 734 return AUDIO_FORMAT_INVALID; 735 } 736 return thread->format(); 737} 738 739size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 740{ 741 Mutex::Autolock _l(mLock); 742 ThreadBase *thread = checkThread_l(ioHandle); 743 if (thread == NULL) { 744 ALOGW("frameCount() unknown thread %d", ioHandle); 745 return 0; 746 } 747 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 748 // should examine all callers and fix them to handle smaller counts 749 return thread->frameCount(); 750} 751 752size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const 753{ 754 Mutex::Autolock _l(mLock); 755 ThreadBase *thread = checkThread_l(ioHandle); 756 if (thread == NULL) { 757 ALOGW("frameCountHAL() unknown thread %d", ioHandle); 758 return 0; 759 } 760 return thread->frameCountHAL(); 761} 762 763uint32_t AudioFlinger::latency(audio_io_handle_t output) const 764{ 765 Mutex::Autolock _l(mLock); 766 PlaybackThread *thread = checkPlaybackThread_l(output); 767 if (thread == NULL) { 768 ALOGW("latency(): no playback thread found for output handle %d", output); 769 return 0; 770 } 771 return thread->latency(); 772} 773 774status_t AudioFlinger::setMasterVolume(float value) 775{ 776 status_t ret = initCheck(); 777 if (ret != NO_ERROR) { 778 return ret; 779 } 780 781 // check calling permissions 782 if (!settingsAllowed()) { 783 return PERMISSION_DENIED; 784 } 785 786 Mutex::Autolock _l(mLock); 787 mMasterVolume = value; 788 789 // Set master volume in the HALs which support it. 790 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 791 AutoMutex lock(mHardwareLock); 792 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 793 794 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 795 if (dev->canSetMasterVolume()) { 796 dev->hwDevice()->setMasterVolume(value); 797 } 798 mHardwareStatus = AUDIO_HW_IDLE; 799 } 800 801 // Now set the master volume in each playback thread. Playback threads 802 // assigned to HALs which do not have master volume support will apply 803 // master volume during the mix operation. Threads with HALs which do 804 // support master volume will simply ignore the setting. 805 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 806 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 807 continue; 808 } 809 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 810 } 811 812 return NO_ERROR; 813} 814 815status_t AudioFlinger::setMode(audio_mode_t mode) 816{ 817 status_t ret = initCheck(); 818 if (ret != NO_ERROR) { 819 return ret; 820 } 821 822 // check calling permissions 823 if (!settingsAllowed()) { 824 return PERMISSION_DENIED; 825 } 826 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 827 ALOGW("Illegal value: setMode(%d)", mode); 828 return BAD_VALUE; 829 } 830 831 { // scope for the lock 832 AutoMutex lock(mHardwareLock); 833 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 834 mHardwareStatus = AUDIO_HW_SET_MODE; 835 ret = dev->setMode(mode); 836 mHardwareStatus = AUDIO_HW_IDLE; 837 } 838 839 if (NO_ERROR == ret) { 840 Mutex::Autolock _l(mLock); 841 mMode = mode; 842 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 843 mPlaybackThreads.valueAt(i)->setMode(mode); 844 } 845 846 return ret; 847} 848 849status_t AudioFlinger::setMicMute(bool state) 850{ 851 status_t ret = initCheck(); 852 if (ret != NO_ERROR) { 853 return ret; 854 } 855 856 // check calling permissions 857 if (!settingsAllowed()) { 858 return PERMISSION_DENIED; 859 } 860 861 AutoMutex lock(mHardwareLock); 862 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 863 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 864 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 865 status_t result = dev->setMicMute(state); 866 if (result != NO_ERROR) { 867 ret = result; 868 } 869 } 870 mHardwareStatus = AUDIO_HW_IDLE; 871 return ret; 872} 873 874bool AudioFlinger::getMicMute() const 875{ 876 status_t ret = initCheck(); 877 if (ret != NO_ERROR) { 878 return false; 879 } 880 bool mute = true; 881 bool state = AUDIO_MODE_INVALID; 882 AutoMutex lock(mHardwareLock); 883 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 884 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 885 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 886 status_t result = dev->getMicMute(&state); 887 if (result == NO_ERROR) { 888 mute = mute && state; 889 } 890 } 891 mHardwareStatus = AUDIO_HW_IDLE; 892 893 return mute; 894} 895 896status_t AudioFlinger::setMasterMute(bool muted) 897{ 898 status_t ret = initCheck(); 899 if (ret != NO_ERROR) { 900 return ret; 901 } 902 903 // check calling permissions 904 if (!settingsAllowed()) { 905 return PERMISSION_DENIED; 906 } 907 908 Mutex::Autolock _l(mLock); 909 mMasterMute = muted; 910 911 // Set master mute in the HALs which support it. 912 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 913 AutoMutex lock(mHardwareLock); 914 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 915 916 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 917 if (dev->canSetMasterMute()) { 918 dev->hwDevice()->setMasterMute(muted); 919 } 920 mHardwareStatus = AUDIO_HW_IDLE; 921 } 922 923 // Now set the master mute in each playback thread. Playback threads 924 // assigned to HALs which do not have master mute support will apply master 925 // mute during the mix operation. Threads with HALs which do support master 926 // mute will simply ignore the setting. 927 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 928 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 929 continue; 930 } 931 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 932 } 933 934 return NO_ERROR; 935} 936 937float AudioFlinger::masterVolume() const 938{ 939 Mutex::Autolock _l(mLock); 940 return masterVolume_l(); 941} 942 943bool AudioFlinger::masterMute() const 944{ 945 Mutex::Autolock _l(mLock); 946 return masterMute_l(); 947} 948 949float AudioFlinger::masterVolume_l() const 950{ 951 return mMasterVolume; 952} 953 954bool AudioFlinger::masterMute_l() const 955{ 956 return mMasterMute; 957} 958 959status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 960{ 961 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 962 ALOGW("setStreamVolume() invalid stream %d", stream); 963 return BAD_VALUE; 964 } 965 pid_t caller = IPCThreadState::self()->getCallingPid(); 966 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 967 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 968 return PERMISSION_DENIED; 969 } 970 971 return NO_ERROR; 972} 973 974status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 975 audio_io_handle_t output) 976{ 977 // check calling permissions 978 if (!settingsAllowed()) { 979 return PERMISSION_DENIED; 980 } 981 982 status_t status = checkStreamType(stream); 983 if (status != NO_ERROR) { 984 return status; 985 } 986 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 987 988 AutoMutex lock(mLock); 989 PlaybackThread *thread = NULL; 990 if (output != AUDIO_IO_HANDLE_NONE) { 991 thread = checkPlaybackThread_l(output); 992 if (thread == NULL) { 993 return BAD_VALUE; 994 } 995 } 996 997 mStreamTypes[stream].volume = value; 998 999 if (thread == NULL) { 1000 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1001 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 1002 } 1003 } else { 1004 thread->setStreamVolume(stream, value); 1005 } 1006 1007 return NO_ERROR; 1008} 1009 1010status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 1011{ 1012 // check calling permissions 1013 if (!settingsAllowed()) { 1014 return PERMISSION_DENIED; 1015 } 1016 1017 status_t status = checkStreamType(stream); 1018 if (status != NO_ERROR) { 1019 return status; 1020 } 1021 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 1022 1023 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 1024 ALOGE("setStreamMute() invalid stream %d", stream); 1025 return BAD_VALUE; 1026 } 1027 1028 AutoMutex lock(mLock); 1029 mStreamTypes[stream].mute = muted; 1030 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 1031 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 1032 1033 return NO_ERROR; 1034} 1035 1036float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1037{ 1038 status_t status = checkStreamType(stream); 1039 if (status != NO_ERROR) { 1040 return 0.0f; 1041 } 1042 1043 AutoMutex lock(mLock); 1044 float volume; 1045 if (output != AUDIO_IO_HANDLE_NONE) { 1046 PlaybackThread *thread = checkPlaybackThread_l(output); 1047 if (thread == NULL) { 1048 return 0.0f; 1049 } 1050 volume = thread->streamVolume(stream); 1051 } else { 1052 volume = streamVolume_l(stream); 1053 } 1054 1055 return volume; 1056} 1057 1058bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1059{ 1060 status_t status = checkStreamType(stream); 1061 if (status != NO_ERROR) { 1062 return true; 1063 } 1064 1065 AutoMutex lock(mLock); 1066 return streamMute_l(stream); 1067} 1068 1069 1070void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1071{ 1072 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1073 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1074 } 1075} 1076 1077status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1078{ 1079 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1080 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1081 1082 // check calling permissions 1083 if (!settingsAllowed()) { 1084 return PERMISSION_DENIED; 1085 } 1086 1087 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1088 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1089 Mutex::Autolock _l(mLock); 1090 // result will remain NO_INIT if no audio device is present 1091 status_t final_result = NO_INIT; 1092 { 1093 AutoMutex lock(mHardwareLock); 1094 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1095 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1096 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1097 status_t result = dev->setParameters(keyValuePairs); 1098 // return success if at least one audio device accepts the parameters as not all 1099 // HALs are requested to support all parameters. If no audio device supports the 1100 // requested parameters, the last error is reported. 1101 if (final_result != NO_ERROR) { 1102 final_result = result; 1103 } 1104 } 1105 mHardwareStatus = AUDIO_HW_IDLE; 1106 } 1107 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1108 AudioParameter param = AudioParameter(keyValuePairs); 1109 String8 value; 1110 if (param.get(String8(AudioParameter::keyBtNrec), value) == NO_ERROR) { 1111 bool btNrecIsOff = (value == AudioParameter::valueOff); 1112 if (mBtNrecIsOff != btNrecIsOff) { 1113 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1114 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1115 audio_devices_t device = thread->inDevice(); 1116 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1117 // collect all of the thread's session IDs 1118 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1119 // suspend effects associated with those session IDs 1120 for (size_t j = 0; j < ids.size(); ++j) { 1121 audio_session_t sessionId = ids.keyAt(j); 1122 thread->setEffectSuspended(FX_IID_AEC, 1123 suspend, 1124 sessionId); 1125 thread->setEffectSuspended(FX_IID_NS, 1126 suspend, 1127 sessionId); 1128 } 1129 } 1130 mBtNrecIsOff = btNrecIsOff; 1131 } 1132 } 1133 String8 screenState; 1134 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1135 bool isOff = (screenState == AudioParameter::valueOff); 1136 if (isOff != (AudioFlinger::mScreenState & 1)) { 1137 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1138 } 1139 } 1140 return final_result; 1141 } 1142 1143 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1144 // and the thread is exited once the lock is released 1145 sp<ThreadBase> thread; 1146 { 1147 Mutex::Autolock _l(mLock); 1148 thread = checkPlaybackThread_l(ioHandle); 1149 if (thread == 0) { 1150 thread = checkRecordThread_l(ioHandle); 1151 } else if (thread == primaryPlaybackThread_l()) { 1152 // indicate output device change to all input threads for pre processing 1153 AudioParameter param = AudioParameter(keyValuePairs); 1154 int value; 1155 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1156 (value != 0)) { 1157 broacastParametersToRecordThreads_l(keyValuePairs); 1158 } 1159 } 1160 } 1161 if (thread != 0) { 1162 return thread->setParameters(keyValuePairs); 1163 } 1164 return BAD_VALUE; 1165} 1166 1167String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1168{ 1169 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1170 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1171 1172 Mutex::Autolock _l(mLock); 1173 1174 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1175 String8 out_s8; 1176 1177 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1178 String8 s; 1179 status_t result; 1180 { 1181 AutoMutex lock(mHardwareLock); 1182 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1183 sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1184 result = dev->getParameters(keys, &s); 1185 mHardwareStatus = AUDIO_HW_IDLE; 1186 } 1187 if (result == OK) out_s8 += s; 1188 } 1189 return out_s8; 1190 } 1191 1192 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1193 if (playbackThread != NULL) { 1194 return playbackThread->getParameters(keys); 1195 } 1196 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1197 if (recordThread != NULL) { 1198 return recordThread->getParameters(keys); 1199 } 1200 return String8(""); 1201} 1202 1203size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1204 audio_channel_mask_t channelMask) const 1205{ 1206 status_t ret = initCheck(); 1207 if (ret != NO_ERROR) { 1208 return 0; 1209 } 1210 if ((sampleRate == 0) || 1211 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1212 !audio_is_input_channel(channelMask)) { 1213 return 0; 1214 } 1215 1216 AutoMutex lock(mHardwareLock); 1217 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1218 audio_config_t config, proposed; 1219 memset(&proposed, 0, sizeof(proposed)); 1220 proposed.sample_rate = sampleRate; 1221 proposed.channel_mask = channelMask; 1222 proposed.format = format; 1223 1224 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 1225 size_t frames; 1226 for (;;) { 1227 // Note: config is currently a const parameter for get_input_buffer_size() 1228 // but we use a copy from proposed in case config changes from the call. 1229 config = proposed; 1230 status_t result = dev->getInputBufferSize(&config, &frames); 1231 if (result == OK && frames != 0) { 1232 break; // hal success, config is the result 1233 } 1234 // change one parameter of the configuration each iteration to a more "common" value 1235 // to see if the device will support it. 1236 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1237 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1238 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1239 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1240 } else { 1241 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1242 "format %#x, channelMask 0x%X", 1243 sampleRate, format, channelMask); 1244 break; // retries failed, break out of loop with frames == 0. 1245 } 1246 } 1247 mHardwareStatus = AUDIO_HW_IDLE; 1248 if (frames > 0 && config.sample_rate != sampleRate) { 1249 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1250 } 1251 return frames; // may be converted to bytes at the Java level. 1252} 1253 1254uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1255{ 1256 Mutex::Autolock _l(mLock); 1257 1258 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1259 if (recordThread != NULL) { 1260 return recordThread->getInputFramesLost(); 1261 } 1262 return 0; 1263} 1264 1265status_t AudioFlinger::setVoiceVolume(float value) 1266{ 1267 status_t ret = initCheck(); 1268 if (ret != NO_ERROR) { 1269 return ret; 1270 } 1271 1272 // check calling permissions 1273 if (!settingsAllowed()) { 1274 return PERMISSION_DENIED; 1275 } 1276 1277 AutoMutex lock(mHardwareLock); 1278 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 1279 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1280 ret = dev->setVoiceVolume(value); 1281 mHardwareStatus = AUDIO_HW_IDLE; 1282 1283 return ret; 1284} 1285 1286status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1287 audio_io_handle_t output) const 1288{ 1289 Mutex::Autolock _l(mLock); 1290 1291 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1292 if (playbackThread != NULL) { 1293 return playbackThread->getRenderPosition(halFrames, dspFrames); 1294 } 1295 1296 return BAD_VALUE; 1297} 1298 1299void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1300{ 1301 Mutex::Autolock _l(mLock); 1302 if (client == 0) { 1303 return; 1304 } 1305 pid_t pid = IPCThreadState::self()->getCallingPid(); 1306 { 1307 Mutex::Autolock _cl(mClientLock); 1308 if (mNotificationClients.indexOfKey(pid) < 0) { 1309 sp<NotificationClient> notificationClient = new NotificationClient(this, 1310 client, 1311 pid); 1312 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1313 1314 mNotificationClients.add(pid, notificationClient); 1315 1316 sp<IBinder> binder = IInterface::asBinder(client); 1317 binder->linkToDeath(notificationClient); 1318 } 1319 } 1320 1321 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1322 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1323 // the config change is always sent from playback or record threads to avoid deadlock 1324 // with AudioSystem::gLock 1325 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1326 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1327 } 1328 1329 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1330 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1331 } 1332} 1333 1334void AudioFlinger::removeNotificationClient(pid_t pid) 1335{ 1336 Mutex::Autolock _l(mLock); 1337 { 1338 Mutex::Autolock _cl(mClientLock); 1339 mNotificationClients.removeItem(pid); 1340 } 1341 1342 ALOGV("%d died, releasing its sessions", pid); 1343 size_t num = mAudioSessionRefs.size(); 1344 bool removed = false; 1345 for (size_t i = 0; i< num; ) { 1346 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1347 ALOGV(" pid %d @ %zu", ref->mPid, i); 1348 if (ref->mPid == pid) { 1349 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1350 mAudioSessionRefs.removeAt(i); 1351 delete ref; 1352 removed = true; 1353 num--; 1354 } else { 1355 i++; 1356 } 1357 } 1358 if (removed) { 1359 purgeStaleEffects_l(); 1360 } 1361} 1362 1363void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1364 const sp<AudioIoDescriptor>& ioDesc, 1365 pid_t pid) 1366{ 1367 Mutex::Autolock _l(mClientLock); 1368 size_t size = mNotificationClients.size(); 1369 for (size_t i = 0; i < size; i++) { 1370 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1371 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1372 } 1373 } 1374} 1375 1376// removeClient_l() must be called with AudioFlinger::mClientLock held 1377void AudioFlinger::removeClient_l(pid_t pid) 1378{ 1379 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1380 IPCThreadState::self()->getCallingPid()); 1381 mClients.removeItem(pid); 1382} 1383 1384// getEffectThread_l() must be called with AudioFlinger::mLock held 1385sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1386 int EffectId) 1387{ 1388 sp<PlaybackThread> thread; 1389 1390 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1391 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1392 ALOG_ASSERT(thread == 0); 1393 thread = mPlaybackThreads.valueAt(i); 1394 } 1395 } 1396 1397 return thread; 1398} 1399 1400 1401 1402// ---------------------------------------------------------------------------- 1403 1404AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1405 : RefBase(), 1406 mAudioFlinger(audioFlinger), 1407 mPid(pid) 1408{ 1409 size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0); 1410 heapSize *= 1024; 1411 if (!heapSize) { 1412 heapSize = kClientSharedHeapSizeBytes; 1413 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1414 // invalidated tracks 1415 if (!audioFlinger->isLowRamDevice()) { 1416 heapSize *= kClientSharedHeapSizeMultiplier; 1417 } 1418 } 1419 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1420} 1421 1422// Client destructor must be called with AudioFlinger::mClientLock held 1423AudioFlinger::Client::~Client() 1424{ 1425 mAudioFlinger->removeClient_l(mPid); 1426} 1427 1428sp<MemoryDealer> AudioFlinger::Client::heap() const 1429{ 1430 return mMemoryDealer; 1431} 1432 1433// ---------------------------------------------------------------------------- 1434 1435AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1436 const sp<IAudioFlingerClient>& client, 1437 pid_t pid) 1438 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1439{ 1440} 1441 1442AudioFlinger::NotificationClient::~NotificationClient() 1443{ 1444} 1445 1446void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1447{ 1448 sp<NotificationClient> keep(this); 1449 mAudioFlinger->removeNotificationClient(mPid); 1450} 1451 1452 1453// ---------------------------------------------------------------------------- 1454 1455sp<IAudioRecord> AudioFlinger::openRecord( 1456 audio_io_handle_t input, 1457 uint32_t sampleRate, 1458 audio_format_t format, 1459 audio_channel_mask_t channelMask, 1460 const String16& opPackageName, 1461 size_t *frameCount, 1462 audio_input_flags_t *flags, 1463 pid_t pid, 1464 pid_t tid, 1465 int clientUid, 1466 audio_session_t *sessionId, 1467 size_t *notificationFrames, 1468 sp<IMemory>& cblk, 1469 sp<IMemory>& buffers, 1470 status_t *status) 1471{ 1472 sp<RecordThread::RecordTrack> recordTrack; 1473 sp<RecordHandle> recordHandle; 1474 sp<Client> client; 1475 status_t lStatus; 1476 audio_session_t lSessionId; 1477 1478 cblk.clear(); 1479 buffers.clear(); 1480 1481 bool updatePid = (pid == -1); 1482 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1483 if (!isTrustedCallingUid(callingUid)) { 1484 ALOGW_IF((uid_t)clientUid != callingUid, 1485 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1486 clientUid = callingUid; 1487 updatePid = true; 1488 } 1489 1490 if (updatePid) { 1491 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1492 ALOGW_IF(pid != -1 && pid != callingPid, 1493 "%s uid %d pid %d tried to pass itself off as pid %d", 1494 __func__, callingUid, callingPid, pid); 1495 pid = callingPid; 1496 } 1497 1498 // check calling permissions 1499 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1500 ALOGE("openRecord() permission denied: recording not allowed"); 1501 lStatus = PERMISSION_DENIED; 1502 goto Exit; 1503 } 1504 1505 // further sample rate checks are performed by createRecordTrack_l() 1506 if (sampleRate == 0) { 1507 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1508 lStatus = BAD_VALUE; 1509 goto Exit; 1510 } 1511 1512 // we don't yet support anything other than linear PCM 1513 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1514 ALOGE("openRecord() invalid format %#x", format); 1515 lStatus = BAD_VALUE; 1516 goto Exit; 1517 } 1518 1519 // further channel mask checks are performed by createRecordTrack_l() 1520 if (!audio_is_input_channel(channelMask)) { 1521 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1522 lStatus = BAD_VALUE; 1523 goto Exit; 1524 } 1525 1526 { 1527 Mutex::Autolock _l(mLock); 1528 RecordThread *thread = checkRecordThread_l(input); 1529 if (thread == NULL) { 1530 ALOGE("openRecord() checkRecordThread_l failed"); 1531 lStatus = BAD_VALUE; 1532 goto Exit; 1533 } 1534 1535 client = registerPid(pid); 1536 1537 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1538 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1539 lStatus = BAD_VALUE; 1540 goto Exit; 1541 } 1542 lSessionId = *sessionId; 1543 } else { 1544 // if no audio session id is provided, create one here 1545 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1546 if (sessionId != NULL) { 1547 *sessionId = lSessionId; 1548 } 1549 } 1550 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1551 1552 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1553 frameCount, lSessionId, notificationFrames, 1554 clientUid, flags, tid, &lStatus); 1555 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1556 1557 if (lStatus == NO_ERROR) { 1558 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1559 // session and move it to this thread. 1560 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1561 if (chain != 0) { 1562 Mutex::Autolock _l(thread->mLock); 1563 thread->addEffectChain_l(chain); 1564 } 1565 } 1566 } 1567 1568 if (lStatus != NO_ERROR) { 1569 // remove local strong reference to Client before deleting the RecordTrack so that the 1570 // Client destructor is called by the TrackBase destructor with mClientLock held 1571 // Don't hold mClientLock when releasing the reference on the track as the 1572 // destructor will acquire it. 1573 { 1574 Mutex::Autolock _cl(mClientLock); 1575 client.clear(); 1576 } 1577 recordTrack.clear(); 1578 goto Exit; 1579 } 1580 1581 cblk = recordTrack->getCblk(); 1582 buffers = recordTrack->getBuffers(); 1583 1584 // return handle to client 1585 recordHandle = new RecordHandle(recordTrack); 1586 1587Exit: 1588 *status = lStatus; 1589 return recordHandle; 1590} 1591 1592 1593 1594// ---------------------------------------------------------------------------- 1595 1596audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1597{ 1598 if (name == NULL) { 1599 return AUDIO_MODULE_HANDLE_NONE; 1600 } 1601 if (!settingsAllowed()) { 1602 return AUDIO_MODULE_HANDLE_NONE; 1603 } 1604 Mutex::Autolock _l(mLock); 1605 return loadHwModule_l(name); 1606} 1607 1608// loadHwModule_l() must be called with AudioFlinger::mLock held 1609audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1610{ 1611 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1612 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1613 ALOGW("loadHwModule() module %s already loaded", name); 1614 return mAudioHwDevs.keyAt(i); 1615 } 1616 } 1617 1618 sp<DeviceHalInterface> dev; 1619 1620 int rc = mDevicesFactoryHal->openDevice(name, &dev); 1621 if (rc) { 1622 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1623 return AUDIO_MODULE_HANDLE_NONE; 1624 } 1625 1626 mHardwareStatus = AUDIO_HW_INIT; 1627 rc = dev->initCheck(); 1628 mHardwareStatus = AUDIO_HW_IDLE; 1629 if (rc) { 1630 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1631 return AUDIO_MODULE_HANDLE_NONE; 1632 } 1633 1634 // Check and cache this HAL's level of support for master mute and master 1635 // volume. If this is the first HAL opened, and it supports the get 1636 // methods, use the initial values provided by the HAL as the current 1637 // master mute and volume settings. 1638 1639 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1640 { // scope for auto-lock pattern 1641 AutoMutex lock(mHardwareLock); 1642 1643 if (0 == mAudioHwDevs.size()) { 1644 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1645 float mv; 1646 if (OK == dev->getMasterVolume(&mv)) { 1647 mMasterVolume = mv; 1648 } 1649 1650 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1651 bool mm; 1652 if (OK == dev->getMasterMute(&mm)) { 1653 mMasterMute = mm; 1654 } 1655 } 1656 1657 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1658 if (OK == dev->setMasterVolume(mMasterVolume)) { 1659 flags = static_cast<AudioHwDevice::Flags>(flags | 1660 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1661 } 1662 1663 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1664 if (OK == dev->setMasterMute(mMasterMute)) { 1665 flags = static_cast<AudioHwDevice::Flags>(flags | 1666 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1667 } 1668 1669 mHardwareStatus = AUDIO_HW_IDLE; 1670 } 1671 1672 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1673 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1674 1675 ALOGI("loadHwModule() Loaded %s audio interface, handle %d", name, handle); 1676 1677 return handle; 1678 1679} 1680 1681// ---------------------------------------------------------------------------- 1682 1683uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1684{ 1685 Mutex::Autolock _l(mLock); 1686 PlaybackThread *thread = fastPlaybackThread_l(); 1687 return thread != NULL ? thread->sampleRate() : 0; 1688} 1689 1690size_t AudioFlinger::getPrimaryOutputFrameCount() 1691{ 1692 Mutex::Autolock _l(mLock); 1693 PlaybackThread *thread = fastPlaybackThread_l(); 1694 return thread != NULL ? thread->frameCountHAL() : 0; 1695} 1696 1697// ---------------------------------------------------------------------------- 1698 1699status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1700{ 1701 uid_t uid = IPCThreadState::self()->getCallingUid(); 1702 if (uid != AID_SYSTEM) { 1703 return PERMISSION_DENIED; 1704 } 1705 Mutex::Autolock _l(mLock); 1706 if (mIsDeviceTypeKnown) { 1707 return INVALID_OPERATION; 1708 } 1709 mIsLowRamDevice = isLowRamDevice; 1710 mIsDeviceTypeKnown = true; 1711 return NO_ERROR; 1712} 1713 1714audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1715{ 1716 Mutex::Autolock _l(mLock); 1717 1718 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1719 if (index >= 0) { 1720 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1721 mHwAvSyncIds.valueAt(index), sessionId); 1722 return mHwAvSyncIds.valueAt(index); 1723 } 1724 1725 sp<DeviceHalInterface> dev = mPrimaryHardwareDev->hwDevice(); 1726 if (dev == NULL) { 1727 return AUDIO_HW_SYNC_INVALID; 1728 } 1729 String8 reply; 1730 AudioParameter param; 1731 if (dev->getParameters(String8(AudioParameter::keyHwAvSync), &reply) == OK) { 1732 param = AudioParameter(reply); 1733 } 1734 1735 int value; 1736 if (param.getInt(String8(AudioParameter::keyHwAvSync), value) != NO_ERROR) { 1737 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1738 return AUDIO_HW_SYNC_INVALID; 1739 } 1740 1741 // allow only one session for a given HW A/V sync ID. 1742 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1743 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1744 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1745 value, mHwAvSyncIds.keyAt(i)); 1746 mHwAvSyncIds.removeItemsAt(i); 1747 break; 1748 } 1749 } 1750 1751 mHwAvSyncIds.add(sessionId, value); 1752 1753 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1754 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1755 uint32_t sessions = thread->hasAudioSession(sessionId); 1756 if (sessions & ThreadBase::TRACK_SESSION) { 1757 AudioParameter param = AudioParameter(); 1758 param.addInt(String8(AudioParameter::keyStreamHwAvSync), value); 1759 thread->setParameters(param.toString()); 1760 break; 1761 } 1762 } 1763 1764 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1765 return (audio_hw_sync_t)value; 1766} 1767 1768status_t AudioFlinger::systemReady() 1769{ 1770 Mutex::Autolock _l(mLock); 1771 ALOGI("%s", __FUNCTION__); 1772 if (mSystemReady) { 1773 ALOGW("%s called twice", __FUNCTION__); 1774 return NO_ERROR; 1775 } 1776 mSystemReady = true; 1777 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1778 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1779 thread->systemReady(); 1780 } 1781 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1782 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1783 thread->systemReady(); 1784 } 1785 return NO_ERROR; 1786} 1787 1788// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1789void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1790{ 1791 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1792 if (index >= 0) { 1793 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1794 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1795 AudioParameter param = AudioParameter(); 1796 param.addInt(String8(AudioParameter::keyStreamHwAvSync), syncId); 1797 thread->setParameters(param.toString()); 1798 } 1799} 1800 1801 1802// ---------------------------------------------------------------------------- 1803 1804 1805sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1806 audio_io_handle_t *output, 1807 audio_config_t *config, 1808 audio_devices_t devices, 1809 const String8& address, 1810 audio_output_flags_t flags) 1811{ 1812 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1813 if (outHwDev == NULL) { 1814 return 0; 1815 } 1816 1817 if (*output == AUDIO_IO_HANDLE_NONE) { 1818 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1819 } else { 1820 // Audio Policy does not currently request a specific output handle. 1821 // If this is ever needed, see openInput_l() for example code. 1822 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1823 return 0; 1824 } 1825 1826 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1827 1828 // FOR TESTING ONLY: 1829 // This if statement allows overriding the audio policy settings 1830 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1831 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1832 // Check only for Normal Mixing mode 1833 if (kEnableExtendedPrecision) { 1834 // Specify format (uncomment one below to choose) 1835 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1836 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1837 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1838 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1839 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1840 } 1841 if (kEnableExtendedChannels) { 1842 // Specify channel mask (uncomment one below to choose) 1843 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1844 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1845 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1846 } 1847 } 1848 1849 AudioStreamOut *outputStream = NULL; 1850 status_t status = outHwDev->openOutputStream( 1851 &outputStream, 1852 *output, 1853 devices, 1854 flags, 1855 config, 1856 address.string()); 1857 1858 mHardwareStatus = AUDIO_HW_IDLE; 1859 1860 if (status == NO_ERROR) { 1861 1862 PlaybackThread *thread; 1863 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1864 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1865 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1866 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1867 || !isValidPcmSinkFormat(config->format) 1868 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1869 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1870 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1871 } else { 1872 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1873 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1874 } 1875 mPlaybackThreads.add(*output, thread); 1876 return thread; 1877 } 1878 1879 return 0; 1880} 1881 1882status_t AudioFlinger::openOutput(audio_module_handle_t module, 1883 audio_io_handle_t *output, 1884 audio_config_t *config, 1885 audio_devices_t *devices, 1886 const String8& address, 1887 uint32_t *latencyMs, 1888 audio_output_flags_t flags) 1889{ 1890 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1891 module, 1892 (devices != NULL) ? *devices : 0, 1893 config->sample_rate, 1894 config->format, 1895 config->channel_mask, 1896 flags); 1897 1898 if (*devices == AUDIO_DEVICE_NONE) { 1899 return BAD_VALUE; 1900 } 1901 1902 Mutex::Autolock _l(mLock); 1903 1904 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1905 if (thread != 0) { 1906 *latencyMs = thread->latency(); 1907 1908 // notify client processes of the new output creation 1909 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1910 1911 // the first primary output opened designates the primary hw device 1912 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1913 ALOGI("Using module %d has the primary audio interface", module); 1914 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1915 1916 AutoMutex lock(mHardwareLock); 1917 mHardwareStatus = AUDIO_HW_SET_MODE; 1918 mPrimaryHardwareDev->hwDevice()->setMode(mMode); 1919 mHardwareStatus = AUDIO_HW_IDLE; 1920 } 1921 return NO_ERROR; 1922 } 1923 1924 return NO_INIT; 1925} 1926 1927audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1928 audio_io_handle_t output2) 1929{ 1930 Mutex::Autolock _l(mLock); 1931 MixerThread *thread1 = checkMixerThread_l(output1); 1932 MixerThread *thread2 = checkMixerThread_l(output2); 1933 1934 if (thread1 == NULL || thread2 == NULL) { 1935 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1936 output2); 1937 return AUDIO_IO_HANDLE_NONE; 1938 } 1939 1940 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1941 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1942 thread->addOutputTrack(thread2); 1943 mPlaybackThreads.add(id, thread); 1944 // notify client processes of the new output creation 1945 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1946 return id; 1947} 1948 1949status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1950{ 1951 return closeOutput_nonvirtual(output); 1952} 1953 1954status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1955{ 1956 // keep strong reference on the playback thread so that 1957 // it is not destroyed while exit() is executed 1958 sp<PlaybackThread> thread; 1959 { 1960 Mutex::Autolock _l(mLock); 1961 thread = checkPlaybackThread_l(output); 1962 if (thread == NULL) { 1963 return BAD_VALUE; 1964 } 1965 1966 ALOGV("closeOutput() %d", output); 1967 1968 if (thread->type() == ThreadBase::MIXER) { 1969 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1970 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1971 DuplicatingThread *dupThread = 1972 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1973 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1974 } 1975 } 1976 } 1977 1978 1979 mPlaybackThreads.removeItem(output); 1980 // save all effects to the default thread 1981 if (mPlaybackThreads.size()) { 1982 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1983 if (dstThread != NULL) { 1984 // audioflinger lock is held here so the acquisition order of thread locks does not 1985 // matter 1986 Mutex::Autolock _dl(dstThread->mLock); 1987 Mutex::Autolock _sl(thread->mLock); 1988 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1989 for (size_t i = 0; i < effectChains.size(); i ++) { 1990 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1991 } 1992 } 1993 } 1994 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 1995 ioDesc->mIoHandle = output; 1996 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 1997 } 1998 thread->exit(); 1999 // The thread entity (active unit of execution) is no longer running here, 2000 // but the ThreadBase container still exists. 2001 2002 if (!thread->isDuplicating()) { 2003 closeOutputFinish(thread); 2004 } 2005 2006 return NO_ERROR; 2007} 2008 2009void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread) 2010{ 2011 AudioStreamOut *out = thread->clearOutput(); 2012 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 2013 // from now on thread->mOutput is NULL 2014 delete out; 2015} 2016 2017void AudioFlinger::closeOutputInternal_l(const sp<PlaybackThread>& thread) 2018{ 2019 mPlaybackThreads.removeItem(thread->mId); 2020 thread->exit(); 2021 closeOutputFinish(thread); 2022} 2023 2024status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 2025{ 2026 Mutex::Autolock _l(mLock); 2027 PlaybackThread *thread = checkPlaybackThread_l(output); 2028 2029 if (thread == NULL) { 2030 return BAD_VALUE; 2031 } 2032 2033 ALOGV("suspendOutput() %d", output); 2034 thread->suspend(); 2035 2036 return NO_ERROR; 2037} 2038 2039status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2040{ 2041 Mutex::Autolock _l(mLock); 2042 PlaybackThread *thread = checkPlaybackThread_l(output); 2043 2044 if (thread == NULL) { 2045 return BAD_VALUE; 2046 } 2047 2048 ALOGV("restoreOutput() %d", output); 2049 2050 thread->restore(); 2051 2052 return NO_ERROR; 2053} 2054 2055status_t AudioFlinger::openInput(audio_module_handle_t module, 2056 audio_io_handle_t *input, 2057 audio_config_t *config, 2058 audio_devices_t *devices, 2059 const String8& address, 2060 audio_source_t source, 2061 audio_input_flags_t flags) 2062{ 2063 Mutex::Autolock _l(mLock); 2064 2065 if (*devices == AUDIO_DEVICE_NONE) { 2066 return BAD_VALUE; 2067 } 2068 2069 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2070 2071 if (thread != 0) { 2072 // notify client processes of the new input creation 2073 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2074 return NO_ERROR; 2075 } 2076 return NO_INIT; 2077} 2078 2079sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2080 audio_io_handle_t *input, 2081 audio_config_t *config, 2082 audio_devices_t devices, 2083 const String8& address, 2084 audio_source_t source, 2085 audio_input_flags_t flags) 2086{ 2087 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2088 if (inHwDev == NULL) { 2089 *input = AUDIO_IO_HANDLE_NONE; 2090 return 0; 2091 } 2092 2093 // Audio Policy can request a specific handle for hardware hotword. 2094 // The goal here is not to re-open an already opened input. 2095 // It is to use a pre-assigned I/O handle. 2096 if (*input == AUDIO_IO_HANDLE_NONE) { 2097 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2098 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2099 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2100 return 0; 2101 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2102 // This should not happen in a transient state with current design. 2103 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2104 return 0; 2105 } 2106 2107 audio_config_t halconfig = *config; 2108 sp<DeviceHalInterface> inHwHal = inHwDev->hwDevice(); 2109 sp<StreamInHalInterface> inStream; 2110 status_t status = inHwHal->openInputStream( 2111 *input, devices, &halconfig, flags, address.string(), source, &inStream); 2112 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2113 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2114 inStream.get(), 2115 halconfig.sample_rate, 2116 halconfig.format, 2117 halconfig.channel_mask, 2118 flags, 2119 status, address.string()); 2120 2121 // If the input could not be opened with the requested parameters and we can handle the 2122 // conversion internally, try to open again with the proposed parameters. 2123 if (status == BAD_VALUE && 2124 audio_is_linear_pcm(config->format) && 2125 audio_is_linear_pcm(halconfig.format) && 2126 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2127 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) && 2128 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) { 2129 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2130 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2131 inStream.clear(); 2132 status = inHwHal->openInputStream( 2133 *input, devices, &halconfig, flags, address.string(), source, &inStream); 2134 // FIXME log this new status; HAL should not propose any further changes 2135 } 2136 2137 if (status == NO_ERROR && inStream != 0) { 2138 2139#ifdef TEE_SINK 2140 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2141 // or (re-)create if current Pipe is idle and does not match the new format 2142 sp<NBAIO_Sink> teeSink; 2143 enum { 2144 TEE_SINK_NO, // don't copy input 2145 TEE_SINK_NEW, // copy input using a new pipe 2146 TEE_SINK_OLD, // copy input using an existing pipe 2147 } kind; 2148 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2149 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2150 if (!mTeeSinkInputEnabled) { 2151 kind = TEE_SINK_NO; 2152 } else if (!Format_isValid(format)) { 2153 kind = TEE_SINK_NO; 2154 } else if (mRecordTeeSink == 0) { 2155 kind = TEE_SINK_NEW; 2156 } else if (mRecordTeeSink->getStrongCount() != 1) { 2157 kind = TEE_SINK_NO; 2158 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2159 kind = TEE_SINK_OLD; 2160 } else { 2161 kind = TEE_SINK_NEW; 2162 } 2163 switch (kind) { 2164 case TEE_SINK_NEW: { 2165 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2166 size_t numCounterOffers = 0; 2167 const NBAIO_Format offers[1] = {format}; 2168 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2169 ALOG_ASSERT(index == 0); 2170 PipeReader *pipeReader = new PipeReader(*pipe); 2171 numCounterOffers = 0; 2172 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2173 ALOG_ASSERT(index == 0); 2174 mRecordTeeSink = pipe; 2175 mRecordTeeSource = pipeReader; 2176 teeSink = pipe; 2177 } 2178 break; 2179 case TEE_SINK_OLD: 2180 teeSink = mRecordTeeSink; 2181 break; 2182 case TEE_SINK_NO: 2183 default: 2184 break; 2185 } 2186#endif 2187 2188 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags); 2189 2190 // Start record thread 2191 // RecordThread requires both input and output device indication to forward to audio 2192 // pre processing modules 2193 sp<RecordThread> thread = new RecordThread(this, 2194 inputStream, 2195 *input, 2196 primaryOutputDevice_l(), 2197 devices, 2198 mSystemReady 2199#ifdef TEE_SINK 2200 , teeSink 2201#endif 2202 ); 2203 mRecordThreads.add(*input, thread); 2204 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2205 return thread; 2206 } 2207 2208 *input = AUDIO_IO_HANDLE_NONE; 2209 return 0; 2210} 2211 2212status_t AudioFlinger::closeInput(audio_io_handle_t input) 2213{ 2214 return closeInput_nonvirtual(input); 2215} 2216 2217status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2218{ 2219 // keep strong reference on the record thread so that 2220 // it is not destroyed while exit() is executed 2221 sp<RecordThread> thread; 2222 { 2223 Mutex::Autolock _l(mLock); 2224 thread = checkRecordThread_l(input); 2225 if (thread == 0) { 2226 return BAD_VALUE; 2227 } 2228 2229 ALOGV("closeInput() %d", input); 2230 2231 // If we still have effect chains, it means that a client still holds a handle 2232 // on at least one effect. We must either move the chain to an existing thread with the 2233 // same session ID or put it aside in case a new record thread is opened for a 2234 // new capture on the same session 2235 sp<EffectChain> chain; 2236 { 2237 Mutex::Autolock _sl(thread->mLock); 2238 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2239 // Note: maximum one chain per record thread 2240 if (effectChains.size() != 0) { 2241 chain = effectChains[0]; 2242 } 2243 } 2244 if (chain != 0) { 2245 // first check if a record thread is already opened with a client on the same session. 2246 // This should only happen in case of overlap between one thread tear down and the 2247 // creation of its replacement 2248 size_t i; 2249 for (i = 0; i < mRecordThreads.size(); i++) { 2250 sp<RecordThread> t = mRecordThreads.valueAt(i); 2251 if (t == thread) { 2252 continue; 2253 } 2254 if (t->hasAudioSession(chain->sessionId()) != 0) { 2255 Mutex::Autolock _l(t->mLock); 2256 ALOGV("closeInput() found thread %d for effect session %d", 2257 t->id(), chain->sessionId()); 2258 t->addEffectChain_l(chain); 2259 break; 2260 } 2261 } 2262 // put the chain aside if we could not find a record thread with the same session id. 2263 if (i == mRecordThreads.size()) { 2264 putOrphanEffectChain_l(chain); 2265 } 2266 } 2267 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2268 ioDesc->mIoHandle = input; 2269 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2270 mRecordThreads.removeItem(input); 2271 } 2272 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2273 // we have a different lock for notification client 2274 closeInputFinish(thread); 2275 return NO_ERROR; 2276} 2277 2278void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread) 2279{ 2280 thread->exit(); 2281 AudioStreamIn *in = thread->clearInput(); 2282 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2283 // from now on thread->mInput is NULL 2284 delete in; 2285} 2286 2287void AudioFlinger::closeInputInternal_l(const sp<RecordThread>& thread) 2288{ 2289 mRecordThreads.removeItem(thread->mId); 2290 closeInputFinish(thread); 2291} 2292 2293status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2294{ 2295 Mutex::Autolock _l(mLock); 2296 ALOGV("invalidateStream() stream %d", stream); 2297 2298 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2299 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2300 thread->invalidateTracks(stream); 2301 } 2302 2303 return NO_ERROR; 2304} 2305 2306 2307audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2308{ 2309 // This is a binder API, so a malicious client could pass in a bad parameter. 2310 // Check for that before calling the internal API nextUniqueId(). 2311 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2312 ALOGE("newAudioUniqueId invalid use %d", use); 2313 return AUDIO_UNIQUE_ID_ALLOCATE; 2314 } 2315 return nextUniqueId(use); 2316} 2317 2318void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2319{ 2320 Mutex::Autolock _l(mLock); 2321 pid_t caller = IPCThreadState::self()->getCallingPid(); 2322 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2323 if (pid != -1 && (caller == getpid_cached)) { 2324 caller = pid; 2325 } 2326 2327 { 2328 Mutex::Autolock _cl(mClientLock); 2329 // Ignore requests received from processes not known as notification client. The request 2330 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2331 // called from a different pid leaving a stale session reference. Also we don't know how 2332 // to clear this reference if the client process dies. 2333 if (mNotificationClients.indexOfKey(caller) < 0) { 2334 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2335 return; 2336 } 2337 } 2338 2339 size_t num = mAudioSessionRefs.size(); 2340 for (size_t i = 0; i< num; i++) { 2341 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2342 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2343 ref->mCnt++; 2344 ALOGV(" incremented refcount to %d", ref->mCnt); 2345 return; 2346 } 2347 } 2348 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2349 ALOGV(" added new entry for %d", audioSession); 2350} 2351 2352void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2353{ 2354 Mutex::Autolock _l(mLock); 2355 pid_t caller = IPCThreadState::self()->getCallingPid(); 2356 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2357 if (pid != -1 && (caller == getpid_cached)) { 2358 caller = pid; 2359 } 2360 size_t num = mAudioSessionRefs.size(); 2361 for (size_t i = 0; i< num; i++) { 2362 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2363 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2364 ref->mCnt--; 2365 ALOGV(" decremented refcount to %d", ref->mCnt); 2366 if (ref->mCnt == 0) { 2367 mAudioSessionRefs.removeAt(i); 2368 delete ref; 2369 purgeStaleEffects_l(); 2370 } 2371 return; 2372 } 2373 } 2374 // If the caller is mediaserver it is likely that the session being released was acquired 2375 // on behalf of a process not in notification clients and we ignore the warning. 2376 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2377} 2378 2379void AudioFlinger::purgeStaleEffects_l() { 2380 2381 ALOGV("purging stale effects"); 2382 2383 Vector< sp<EffectChain> > chains; 2384 2385 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2386 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2387 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2388 sp<EffectChain> ec = t->mEffectChains[j]; 2389 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2390 chains.push(ec); 2391 } 2392 } 2393 } 2394 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2395 sp<RecordThread> t = mRecordThreads.valueAt(i); 2396 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2397 sp<EffectChain> ec = t->mEffectChains[j]; 2398 chains.push(ec); 2399 } 2400 } 2401 2402 for (size_t i = 0; i < chains.size(); i++) { 2403 sp<EffectChain> ec = chains[i]; 2404 int sessionid = ec->sessionId(); 2405 sp<ThreadBase> t = ec->mThread.promote(); 2406 if (t == 0) { 2407 continue; 2408 } 2409 size_t numsessionrefs = mAudioSessionRefs.size(); 2410 bool found = false; 2411 for (size_t k = 0; k < numsessionrefs; k++) { 2412 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2413 if (ref->mSessionid == sessionid) { 2414 ALOGV(" session %d still exists for %d with %d refs", 2415 sessionid, ref->mPid, ref->mCnt); 2416 found = true; 2417 break; 2418 } 2419 } 2420 if (!found) { 2421 Mutex::Autolock _l(t->mLock); 2422 // remove all effects from the chain 2423 while (ec->mEffects.size()) { 2424 sp<EffectModule> effect = ec->mEffects[0]; 2425 effect->unPin(); 2426 t->removeEffect_l(effect); 2427 if (effect->purgeHandles()) { 2428 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2429 } 2430 AudioSystem::unregisterEffect(effect->id()); 2431 } 2432 } 2433 } 2434 return; 2435} 2436 2437// checkThread_l() must be called with AudioFlinger::mLock held 2438AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2439{ 2440 ThreadBase *thread = NULL; 2441 switch (audio_unique_id_get_use(ioHandle)) { 2442 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2443 thread = checkPlaybackThread_l(ioHandle); 2444 break; 2445 case AUDIO_UNIQUE_ID_USE_INPUT: 2446 thread = checkRecordThread_l(ioHandle); 2447 break; 2448 default: 2449 break; 2450 } 2451 return thread; 2452} 2453 2454// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2455AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2456{ 2457 return mPlaybackThreads.valueFor(output).get(); 2458} 2459 2460// checkMixerThread_l() must be called with AudioFlinger::mLock held 2461AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2462{ 2463 PlaybackThread *thread = checkPlaybackThread_l(output); 2464 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2465} 2466 2467// checkRecordThread_l() must be called with AudioFlinger::mLock held 2468AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2469{ 2470 return mRecordThreads.valueFor(input).get(); 2471} 2472 2473audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2474{ 2475 // This is the internal API, so it is OK to assert on bad parameter. 2476 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2477 const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1; 2478 for (int retry = 0; retry < maxRetries; retry++) { 2479 // The cast allows wraparound from max positive to min negative instead of abort 2480 uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use], 2481 (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel); 2482 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2483 // allow wrap by skipping 0 and -1 for session ids 2484 if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) { 2485 ALOGW_IF(retry != 0, "unique ID overflow for use %d", use); 2486 return (audio_unique_id_t) (base | use); 2487 } 2488 } 2489 // We have no way of recovering from wraparound 2490 LOG_ALWAYS_FATAL("unique ID overflow for use %d", use); 2491 // TODO Use a floor after wraparound. This may need a mutex. 2492} 2493 2494AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2495{ 2496 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2497 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2498 if(thread->isDuplicating()) { 2499 continue; 2500 } 2501 AudioStreamOut *output = thread->getOutput(); 2502 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2503 return thread; 2504 } 2505 } 2506 return NULL; 2507} 2508 2509audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2510{ 2511 PlaybackThread *thread = primaryPlaybackThread_l(); 2512 2513 if (thread == NULL) { 2514 return 0; 2515 } 2516 2517 return thread->outDevice(); 2518} 2519 2520AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const 2521{ 2522 size_t minFrameCount = 0; 2523 PlaybackThread *minThread = NULL; 2524 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2525 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2526 if (!thread->isDuplicating()) { 2527 size_t frameCount = thread->frameCountHAL(); 2528 if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount || 2529 (frameCount == minFrameCount && thread->hasFastMixer() && 2530 /*minThread != NULL &&*/ !minThread->hasFastMixer()))) { 2531 minFrameCount = frameCount; 2532 minThread = thread; 2533 } 2534 } 2535 } 2536 return minThread; 2537} 2538 2539sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2540 audio_session_t triggerSession, 2541 audio_session_t listenerSession, 2542 sync_event_callback_t callBack, 2543 const wp<RefBase>& cookie) 2544{ 2545 Mutex::Autolock _l(mLock); 2546 2547 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2548 status_t playStatus = NAME_NOT_FOUND; 2549 status_t recStatus = NAME_NOT_FOUND; 2550 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2551 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2552 if (playStatus == NO_ERROR) { 2553 return event; 2554 } 2555 } 2556 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2557 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2558 if (recStatus == NO_ERROR) { 2559 return event; 2560 } 2561 } 2562 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2563 mPendingSyncEvents.add(event); 2564 } else { 2565 ALOGV("createSyncEvent() invalid event %d", event->type()); 2566 event.clear(); 2567 } 2568 return event; 2569} 2570 2571// ---------------------------------------------------------------------------- 2572// Effect management 2573// ---------------------------------------------------------------------------- 2574 2575sp<EffectsFactoryHalInterface> AudioFlinger::getEffectsFactory() { 2576 return mEffectsFactoryHal; 2577} 2578 2579status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2580{ 2581 Mutex::Autolock _l(mLock); 2582 if (mEffectsFactoryHal.get()) { 2583 return mEffectsFactoryHal->queryNumberEffects(numEffects); 2584 } else { 2585 return -ENODEV; 2586 } 2587} 2588 2589status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2590{ 2591 Mutex::Autolock _l(mLock); 2592 if (mEffectsFactoryHal.get()) { 2593 return mEffectsFactoryHal->getDescriptor(index, descriptor); 2594 } else { 2595 return -ENODEV; 2596 } 2597} 2598 2599status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2600 effect_descriptor_t *descriptor) const 2601{ 2602 Mutex::Autolock _l(mLock); 2603 if (mEffectsFactoryHal.get()) { 2604 return mEffectsFactoryHal->getDescriptor(pUuid, descriptor); 2605 } else { 2606 return -ENODEV; 2607 } 2608} 2609 2610 2611sp<IEffect> AudioFlinger::createEffect( 2612 effect_descriptor_t *pDesc, 2613 const sp<IEffectClient>& effectClient, 2614 int32_t priority, 2615 audio_io_handle_t io, 2616 audio_session_t sessionId, 2617 const String16& opPackageName, 2618 status_t *status, 2619 int *id, 2620 int *enabled) 2621{ 2622 status_t lStatus = NO_ERROR; 2623 sp<EffectHandle> handle; 2624 effect_descriptor_t desc; 2625 2626 pid_t pid = IPCThreadState::self()->getCallingPid(); 2627 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d, factory %p", 2628 pid, effectClient.get(), priority, sessionId, io, mEffectsFactoryHal.get()); 2629 2630 if (pDesc == NULL) { 2631 lStatus = BAD_VALUE; 2632 goto Exit; 2633 } 2634 2635 // check audio settings permission for global effects 2636 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2637 lStatus = PERMISSION_DENIED; 2638 goto Exit; 2639 } 2640 2641 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2642 // that can only be created by audio policy manager (running in same process) 2643 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2644 lStatus = PERMISSION_DENIED; 2645 goto Exit; 2646 } 2647 2648 if (mEffectsFactoryHal == 0) { 2649 lStatus = NO_INIT; 2650 goto Exit; 2651 } 2652 2653 { 2654 if (!EffectsFactoryHalInterface::isNullUuid(&pDesc->uuid)) { 2655 // if uuid is specified, request effect descriptor 2656 lStatus = mEffectsFactoryHal->getDescriptor(&pDesc->uuid, &desc); 2657 if (lStatus < 0) { 2658 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2659 goto Exit; 2660 } 2661 } else { 2662 // if uuid is not specified, look for an available implementation 2663 // of the required type in effect factory 2664 if (EffectsFactoryHalInterface::isNullUuid(&pDesc->type)) { 2665 ALOGW("createEffect() no effect type"); 2666 lStatus = BAD_VALUE; 2667 goto Exit; 2668 } 2669 uint32_t numEffects = 0; 2670 effect_descriptor_t d; 2671 d.flags = 0; // prevent compiler warning 2672 bool found = false; 2673 2674 lStatus = mEffectsFactoryHal->queryNumberEffects(&numEffects); 2675 if (lStatus < 0) { 2676 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2677 goto Exit; 2678 } 2679 for (uint32_t i = 0; i < numEffects; i++) { 2680 lStatus = mEffectsFactoryHal->getDescriptor(i, &desc); 2681 if (lStatus < 0) { 2682 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2683 continue; 2684 } 2685 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2686 // If matching type found save effect descriptor. If the session is 2687 // 0 and the effect is not auxiliary, continue enumeration in case 2688 // an auxiliary version of this effect type is available 2689 found = true; 2690 d = desc; 2691 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2692 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2693 break; 2694 } 2695 } 2696 } 2697 if (!found) { 2698 lStatus = BAD_VALUE; 2699 ALOGW("createEffect() effect not found"); 2700 goto Exit; 2701 } 2702 // For same effect type, chose auxiliary version over insert version if 2703 // connect to output mix (Compliance to OpenSL ES) 2704 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2705 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2706 desc = d; 2707 } 2708 } 2709 2710 // Do not allow auxiliary effects on a session different from 0 (output mix) 2711 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2712 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2713 lStatus = INVALID_OPERATION; 2714 goto Exit; 2715 } 2716 2717 // check recording permission for visualizer 2718 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2719 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2720 lStatus = PERMISSION_DENIED; 2721 goto Exit; 2722 } 2723 2724 // return effect descriptor 2725 *pDesc = desc; 2726 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2727 // if the output returned by getOutputForEffect() is removed before we lock the 2728 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2729 // and we will exit safely 2730 io = AudioSystem::getOutputForEffect(&desc); 2731 ALOGV("createEffect got output %d", io); 2732 } 2733 2734 Mutex::Autolock _l(mLock); 2735 2736 // If output is not specified try to find a matching audio session ID in one of the 2737 // output threads. 2738 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2739 // because of code checking output when entering the function. 2740 // Note: io is never 0 when creating an effect on an input 2741 if (io == AUDIO_IO_HANDLE_NONE) { 2742 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2743 // output must be specified by AudioPolicyManager when using session 2744 // AUDIO_SESSION_OUTPUT_STAGE 2745 lStatus = BAD_VALUE; 2746 goto Exit; 2747 } 2748 // look for the thread where the specified audio session is present 2749 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2750 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2751 io = mPlaybackThreads.keyAt(i); 2752 break; 2753 } 2754 } 2755 if (io == 0) { 2756 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2757 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2758 io = mRecordThreads.keyAt(i); 2759 break; 2760 } 2761 } 2762 } 2763 // If no output thread contains the requested session ID, default to 2764 // first output. The effect chain will be moved to the correct output 2765 // thread when a track with the same session ID is created 2766 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2767 io = mPlaybackThreads.keyAt(0); 2768 } 2769 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2770 } 2771 ThreadBase *thread = checkRecordThread_l(io); 2772 if (thread == NULL) { 2773 thread = checkPlaybackThread_l(io); 2774 if (thread == NULL) { 2775 ALOGE("createEffect() unknown output thread"); 2776 lStatus = BAD_VALUE; 2777 goto Exit; 2778 } 2779 } else { 2780 // Check if one effect chain was awaiting for an effect to be created on this 2781 // session and used it instead of creating a new one. 2782 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2783 if (chain != 0) { 2784 Mutex::Autolock _l(thread->mLock); 2785 thread->addEffectChain_l(chain); 2786 } 2787 } 2788 2789 sp<Client> client = registerPid(pid); 2790 2791 // create effect on selected output thread 2792 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2793 &desc, enabled, &lStatus); 2794 if (handle != 0 && id != NULL) { 2795 *id = handle->id(); 2796 } 2797 if (handle == 0) { 2798 // remove local strong reference to Client with mClientLock held 2799 Mutex::Autolock _cl(mClientLock); 2800 client.clear(); 2801 } 2802 } 2803 2804Exit: 2805 *status = lStatus; 2806 return handle; 2807} 2808 2809status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2810 audio_io_handle_t dstOutput) 2811{ 2812 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2813 sessionId, srcOutput, dstOutput); 2814 Mutex::Autolock _l(mLock); 2815 if (srcOutput == dstOutput) { 2816 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2817 return NO_ERROR; 2818 } 2819 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2820 if (srcThread == NULL) { 2821 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2822 return BAD_VALUE; 2823 } 2824 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2825 if (dstThread == NULL) { 2826 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2827 return BAD_VALUE; 2828 } 2829 2830 Mutex::Autolock _dl(dstThread->mLock); 2831 Mutex::Autolock _sl(srcThread->mLock); 2832 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2833} 2834 2835// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2836status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2837 AudioFlinger::PlaybackThread *srcThread, 2838 AudioFlinger::PlaybackThread *dstThread, 2839 bool reRegister) 2840{ 2841 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2842 sessionId, srcThread, dstThread); 2843 2844 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2845 if (chain == 0) { 2846 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2847 sessionId, srcThread); 2848 return INVALID_OPERATION; 2849 } 2850 2851 // Check whether the destination thread and all effects in the chain are compatible 2852 if (!chain->isCompatibleWithThread_l(dstThread)) { 2853 ALOGW("moveEffectChain_l() effect chain failed because" 2854 " destination thread %p is not compatible with effects in the chain", 2855 dstThread); 2856 return INVALID_OPERATION; 2857 } 2858 2859 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2860 // so that a new chain is created with correct parameters when first effect is added. This is 2861 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2862 // removed. 2863 srcThread->removeEffectChain_l(chain); 2864 2865 // transfer all effects one by one so that new effect chain is created on new thread with 2866 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2867 sp<EffectChain> dstChain; 2868 uint32_t strategy = 0; // prevent compiler warning 2869 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2870 Vector< sp<EffectModule> > removed; 2871 status_t status = NO_ERROR; 2872 while (effect != 0) { 2873 srcThread->removeEffect_l(effect); 2874 removed.add(effect); 2875 status = dstThread->addEffect_l(effect); 2876 if (status != NO_ERROR) { 2877 break; 2878 } 2879 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2880 if (effect->state() == EffectModule::ACTIVE || 2881 effect->state() == EffectModule::STOPPING) { 2882 effect->start(); 2883 } 2884 // if the move request is not received from audio policy manager, the effect must be 2885 // re-registered with the new strategy and output 2886 if (dstChain == 0) { 2887 dstChain = effect->chain().promote(); 2888 if (dstChain == 0) { 2889 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2890 status = NO_INIT; 2891 break; 2892 } 2893 strategy = dstChain->strategy(); 2894 } 2895 if (reRegister) { 2896 AudioSystem::unregisterEffect(effect->id()); 2897 AudioSystem::registerEffect(&effect->desc(), 2898 dstThread->id(), 2899 strategy, 2900 sessionId, 2901 effect->id()); 2902 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2903 } 2904 effect = chain->getEffectFromId_l(0); 2905 } 2906 2907 if (status != NO_ERROR) { 2908 for (size_t i = 0; i < removed.size(); i++) { 2909 srcThread->addEffect_l(removed[i]); 2910 if (dstChain != 0 && reRegister) { 2911 AudioSystem::unregisterEffect(removed[i]->id()); 2912 AudioSystem::registerEffect(&removed[i]->desc(), 2913 srcThread->id(), 2914 strategy, 2915 sessionId, 2916 removed[i]->id()); 2917 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2918 } 2919 } 2920 } 2921 2922 return status; 2923} 2924 2925bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2926{ 2927 if (mGlobalEffectEnableTime != 0 && 2928 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2929 return true; 2930 } 2931 2932 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2933 sp<EffectChain> ec = 2934 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2935 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2936 return true; 2937 } 2938 } 2939 return false; 2940} 2941 2942void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2943{ 2944 Mutex::Autolock _l(mLock); 2945 2946 mGlobalEffectEnableTime = systemTime(); 2947 2948 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2949 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2950 if (t->mType == ThreadBase::OFFLOAD) { 2951 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2952 } 2953 } 2954 2955} 2956 2957status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2958{ 2959 audio_session_t session = chain->sessionId(); 2960 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2961 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2962 if (index >= 0) { 2963 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2964 return ALREADY_EXISTS; 2965 } 2966 mOrphanEffectChains.add(session, chain); 2967 return NO_ERROR; 2968} 2969 2970sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2971{ 2972 sp<EffectChain> chain; 2973 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2974 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2975 if (index >= 0) { 2976 chain = mOrphanEffectChains.valueAt(index); 2977 mOrphanEffectChains.removeItemsAt(index); 2978 } 2979 return chain; 2980} 2981 2982bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2983{ 2984 Mutex::Autolock _l(mLock); 2985 audio_session_t session = effect->sessionId(); 2986 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2987 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2988 if (index >= 0) { 2989 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2990 if (chain->removeEffect_l(effect) == 0) { 2991 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 2992 mOrphanEffectChains.removeItemsAt(index); 2993 } 2994 return true; 2995 } 2996 return false; 2997} 2998 2999 3000struct Entry { 3001#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 3002 char mFileName[TEE_MAX_FILENAME]; 3003}; 3004 3005int comparEntry(const void *p1, const void *p2) 3006{ 3007 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 3008} 3009 3010#ifdef TEE_SINK 3011void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3012{ 3013 NBAIO_Source *teeSource = source.get(); 3014 if (teeSource != NULL) { 3015 // .wav rotation 3016 // There is a benign race condition if 2 threads call this simultaneously. 3017 // They would both traverse the directory, but the result would simply be 3018 // failures at unlink() which are ignored. It's also unlikely since 3019 // normally dumpsys is only done by bugreport or from the command line. 3020 char teePath[32+256]; 3021 strcpy(teePath, "/data/misc/audioserver"); 3022 size_t teePathLen = strlen(teePath); 3023 DIR *dir = opendir(teePath); 3024 teePath[teePathLen++] = '/'; 3025 if (dir != NULL) { 3026#define TEE_MAX_SORT 20 // number of entries to sort 3027#define TEE_MAX_KEEP 10 // number of entries to keep 3028 struct Entry entries[TEE_MAX_SORT]; 3029 size_t entryCount = 0; 3030 while (entryCount < TEE_MAX_SORT) { 3031 struct dirent de; 3032 struct dirent *result = NULL; 3033 int rc = readdir_r(dir, &de, &result); 3034 if (rc != 0) { 3035 ALOGW("readdir_r failed %d", rc); 3036 break; 3037 } 3038 if (result == NULL) { 3039 break; 3040 } 3041 if (result != &de) { 3042 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 3043 break; 3044 } 3045 // ignore non .wav file entries 3046 size_t nameLen = strlen(de.d_name); 3047 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 3048 strcmp(&de.d_name[nameLen - 4], ".wav")) { 3049 continue; 3050 } 3051 strcpy(entries[entryCount++].mFileName, de.d_name); 3052 } 3053 (void) closedir(dir); 3054 if (entryCount > TEE_MAX_KEEP) { 3055 qsort(entries, entryCount, sizeof(Entry), comparEntry); 3056 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 3057 strcpy(&teePath[teePathLen], entries[i].mFileName); 3058 (void) unlink(teePath); 3059 } 3060 } 3061 } else { 3062 if (fd >= 0) { 3063 dprintf(fd, "unable to rotate tees in %.*s: %s\n", (int) teePathLen, teePath, 3064 strerror(errno)); 3065 } 3066 } 3067 char teeTime[16]; 3068 struct timeval tv; 3069 gettimeofday(&tv, NULL); 3070 struct tm tm; 3071 localtime_r(&tv.tv_sec, &tm); 3072 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 3073 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 3074 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 3075 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 3076 if (teeFd >= 0) { 3077 // FIXME use libsndfile 3078 char wavHeader[44]; 3079 memcpy(wavHeader, 3080 "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", 3081 sizeof(wavHeader)); 3082 NBAIO_Format format = teeSource->format(); 3083 unsigned channelCount = Format_channelCount(format); 3084 uint32_t sampleRate = Format_sampleRate(format); 3085 size_t frameSize = Format_frameSize(format); 3086 wavHeader[22] = channelCount; // number of channels 3087 wavHeader[24] = sampleRate; // sample rate 3088 wavHeader[25] = sampleRate >> 8; 3089 wavHeader[32] = frameSize; // block alignment 3090 wavHeader[33] = frameSize >> 8; 3091 write(teeFd, wavHeader, sizeof(wavHeader)); 3092 size_t total = 0; 3093 bool firstRead = true; 3094#define TEE_SINK_READ 1024 // frames per I/O operation 3095 void *buffer = malloc(TEE_SINK_READ * frameSize); 3096 for (;;) { 3097 size_t count = TEE_SINK_READ; 3098 ssize_t actual = teeSource->read(buffer, count); 3099 bool wasFirstRead = firstRead; 3100 firstRead = false; 3101 if (actual <= 0) { 3102 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3103 continue; 3104 } 3105 break; 3106 } 3107 ALOG_ASSERT(actual <= (ssize_t)count); 3108 write(teeFd, buffer, actual * frameSize); 3109 total += actual; 3110 } 3111 free(buffer); 3112 lseek(teeFd, (off_t) 4, SEEK_SET); 3113 uint32_t temp = 44 + total * frameSize - 8; 3114 // FIXME not big-endian safe 3115 write(teeFd, &temp, sizeof(temp)); 3116 lseek(teeFd, (off_t) 40, SEEK_SET); 3117 temp = total * frameSize; 3118 // FIXME not big-endian safe 3119 write(teeFd, &temp, sizeof(temp)); 3120 close(teeFd); 3121 if (fd >= 0) { 3122 dprintf(fd, "tee copied to %s\n", teePath); 3123 } 3124 } else { 3125 if (fd >= 0) { 3126 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3127 } 3128 } 3129 } 3130} 3131#endif 3132 3133// ---------------------------------------------------------------------------- 3134 3135status_t AudioFlinger::onTransact( 3136 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3137{ 3138 return BnAudioFlinger::onTransact(code, data, reply, flags); 3139} 3140 3141} // namespace android 3142