AudioFlinger.cpp revision 5cb96b83e6bff91a5ab006ff542dc32031be9087
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 <memunreachable/memunreachable.h> 35#include <utils/String16.h> 36#include <utils/threads.h> 37#include <utils/Atomic.h> 38 39#include <cutils/bitops.h> 40#include <cutils/properties.h> 41 42#include <system/audio.h> 43#include <hardware/audio.h> 44 45#include "AudioMixer.h" 46#include "AudioFlinger.h" 47#include "ServiceUtilities.h" 48 49#include <media/AudioResamplerPublic.h> 50 51#include <media/EffectsFactoryApi.h> 52#include <audio_effects/effect_visualizer.h> 53#include <audio_effects/effect_ns.h> 54#include <audio_effects/effect_aec.h> 55 56#include <audio_utils/primitives.h> 57 58#include <powermanager/PowerManager.h> 59 60#include <media/IMediaLogService.h> 61#include <media/MemoryLeakTrackUtil.h> 62#include <media/nbaio/Pipe.h> 63#include <media/nbaio/PipeReader.h> 64#include <media/AudioParameter.h> 65#include <mediautils/BatteryNotifier.h> 66#include <private/android_filesystem_config.h> 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92uint32_t AudioFlinger::mScreenState; 93 94#ifdef TEE_SINK 95bool AudioFlinger::mTeeSinkInputEnabled = false; 96bool AudioFlinger::mTeeSinkOutputEnabled = false; 97bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 102#endif 103 104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 105// we define a minimum time during which a global effect is considered enabled. 106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108// ---------------------------------------------------------------------------- 109 110const char *formatToString(audio_format_t format) { 111 switch (audio_get_main_format(format)) { 112 case AUDIO_FORMAT_PCM: 113 switch (format) { 114 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 115 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 116 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 117 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 118 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 119 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 120 default: 121 break; 122 } 123 break; 124 case AUDIO_FORMAT_MP3: return "mp3"; 125 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 126 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 127 case AUDIO_FORMAT_AAC: return "aac"; 128 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 129 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 130 case AUDIO_FORMAT_VORBIS: return "vorbis"; 131 case AUDIO_FORMAT_OPUS: return "opus"; 132 case AUDIO_FORMAT_AC3: return "ac-3"; 133 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 134 case AUDIO_FORMAT_IEC61937: return "iec61937"; 135 default: 136 break; 137 } 138 return "unknown"; 139} 140 141static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 142{ 143 const hw_module_t *mod; 144 int rc; 145 146 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 147 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 148 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 149 if (rc) { 150 goto out; 151 } 152 rc = audio_hw_device_open(mod, dev); 153 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 154 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 155 if (rc) { 156 goto out; 157 } 158 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 159 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 160 rc = BAD_VALUE; 161 goto out; 162 } 163 return 0; 164 165out: 166 *dev = NULL; 167 return rc; 168} 169 170// ---------------------------------------------------------------------------- 171 172AudioFlinger::AudioFlinger() 173 : BnAudioFlinger(), 174 mPrimaryHardwareDev(NULL), 175 mAudioHwDevs(NULL), 176 mHardwareStatus(AUDIO_HW_IDLE), 177 mMasterVolume(1.0f), 178 mMasterMute(false), 179 // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), 180 mMode(AUDIO_MODE_INVALID), 181 mBtNrecIsOff(false), 182 mIsLowRamDevice(true), 183 mIsDeviceTypeKnown(false), 184 mGlobalEffectEnableTime(0), 185 mSystemReady(false) 186{ 187 // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum 188 for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) { 189 // zero ID has a special meaning, so unavailable 190 mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX; 191 } 192 193 getpid_cached = getpid(); 194 const bool doLog = property_get_bool("ro.test_harness", false); 195 if (doLog) { 196 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 197 MemoryHeapBase::READ_ONLY); 198 } 199 200 // reset battery stats. 201 // if the audio service has crashed, battery stats could be left 202 // in bad state, reset the state upon service start. 203 BatteryNotifier::getInstance().noteResetAudio(); 204 205#ifdef TEE_SINK 206 char value[PROPERTY_VALUE_MAX]; 207 (void) property_get("ro.debuggable", value, "0"); 208 int debuggable = atoi(value); 209 int teeEnabled = 0; 210 if (debuggable) { 211 (void) property_get("af.tee", value, "0"); 212 teeEnabled = atoi(value); 213 } 214 // FIXME symbolic constants here 215 if (teeEnabled & 1) { 216 mTeeSinkInputEnabled = true; 217 } 218 if (teeEnabled & 2) { 219 mTeeSinkOutputEnabled = true; 220 } 221 if (teeEnabled & 4) { 222 mTeeSinkTrackEnabled = true; 223 } 224#endif 225} 226 227void AudioFlinger::onFirstRef() 228{ 229 Mutex::Autolock _l(mLock); 230 231 /* TODO: move all this work into an Init() function */ 232 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 233 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 234 uint32_t int_val; 235 if (1 == sscanf(val_str, "%u", &int_val)) { 236 mStandbyTimeInNsecs = milliseconds(int_val); 237 ALOGI("Using %u mSec as standby time.", int_val); 238 } else { 239 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 240 ALOGI("Using default %u mSec as standby time.", 241 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 242 } 243 } 244 245 mPatchPanel = new PatchPanel(this); 246 247 mMode = AUDIO_MODE_NORMAL; 248} 249 250AudioFlinger::~AudioFlinger() 251{ 252 while (!mRecordThreads.isEmpty()) { 253 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 254 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 255 } 256 while (!mPlaybackThreads.isEmpty()) { 257 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 258 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 259 } 260 261 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 262 // no mHardwareLock needed, as there are no other references to this 263 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 264 delete mAudioHwDevs.valueAt(i); 265 } 266 267 // Tell media.log service about any old writers that still need to be unregistered 268 if (mLogMemoryDealer != 0) { 269 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 270 if (binder != 0) { 271 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 272 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 273 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 274 mUnregisteredWriters.pop(); 275 mediaLogService->unregisterWriter(iMemory); 276 } 277 } 278 } 279} 280 281static const char * const audio_interfaces[] = { 282 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 283 AUDIO_HARDWARE_MODULE_ID_A2DP, 284 AUDIO_HARDWARE_MODULE_ID_USB, 285}; 286#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 287 288AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 289 audio_module_handle_t module, 290 audio_devices_t devices) 291{ 292 // if module is 0, the request comes from an old policy manager and we should load 293 // well known modules 294 if (module == 0) { 295 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 296 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 297 loadHwModule_l(audio_interfaces[i]); 298 } 299 // then try to find a module supporting the requested device. 300 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 301 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 302 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 303 if ((dev->get_supported_devices != NULL) && 304 (dev->get_supported_devices(dev) & devices) == devices) 305 return audioHwDevice; 306 } 307 } else { 308 // check a match for the requested module handle 309 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 310 if (audioHwDevice != NULL) { 311 return audioHwDevice; 312 } 313 } 314 315 return NULL; 316} 317 318void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 319{ 320 const size_t SIZE = 256; 321 char buffer[SIZE]; 322 String8 result; 323 324 result.append("Clients:\n"); 325 for (size_t i = 0; i < mClients.size(); ++i) { 326 sp<Client> client = mClients.valueAt(i).promote(); 327 if (client != 0) { 328 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 329 result.append(buffer); 330 } 331 } 332 333 result.append("Notification Clients:\n"); 334 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 335 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 336 result.append(buffer); 337 } 338 339 result.append("Global session refs:\n"); 340 result.append(" session pid count\n"); 341 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 342 AudioSessionRef *r = mAudioSessionRefs[i]; 343 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 344 result.append(buffer); 345 } 346 write(fd, result.string(), result.size()); 347} 348 349 350void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 351{ 352 const size_t SIZE = 256; 353 char buffer[SIZE]; 354 String8 result; 355 hardware_call_state hardwareStatus = mHardwareStatus; 356 357 snprintf(buffer, SIZE, "Hardware status: %d\n" 358 "Standby Time mSec: %u\n", 359 hardwareStatus, 360 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 361 result.append(buffer); 362 write(fd, result.string(), result.size()); 363} 364 365void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 366{ 367 const size_t SIZE = 256; 368 char buffer[SIZE]; 369 String8 result; 370 snprintf(buffer, SIZE, "Permission Denial: " 371 "can't dump AudioFlinger from pid=%d, uid=%d\n", 372 IPCThreadState::self()->getCallingPid(), 373 IPCThreadState::self()->getCallingUid()); 374 result.append(buffer); 375 write(fd, result.string(), result.size()); 376} 377 378bool AudioFlinger::dumpTryLock(Mutex& mutex) 379{ 380 bool locked = false; 381 for (int i = 0; i < kDumpLockRetries; ++i) { 382 if (mutex.tryLock() == NO_ERROR) { 383 locked = true; 384 break; 385 } 386 usleep(kDumpLockSleepUs); 387 } 388 return locked; 389} 390 391status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 392{ 393 if (!dumpAllowed()) { 394 dumpPermissionDenial(fd, args); 395 } else { 396 // get state of hardware lock 397 bool hardwareLocked = dumpTryLock(mHardwareLock); 398 if (!hardwareLocked) { 399 String8 result(kHardwareLockedString); 400 write(fd, result.string(), result.size()); 401 } else { 402 mHardwareLock.unlock(); 403 } 404 405 bool locked = dumpTryLock(mLock); 406 407 // failed to lock - AudioFlinger is probably deadlocked 408 if (!locked) { 409 String8 result(kDeadlockedString); 410 write(fd, result.string(), result.size()); 411 } 412 413 bool clientLocked = dumpTryLock(mClientLock); 414 if (!clientLocked) { 415 String8 result(kClientLockedString); 416 write(fd, result.string(), result.size()); 417 } 418 419 EffectDumpEffects(fd); 420 421 dumpClients(fd, args); 422 if (clientLocked) { 423 mClientLock.unlock(); 424 } 425 426 dumpInternals(fd, args); 427 428 // dump playback threads 429 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 430 mPlaybackThreads.valueAt(i)->dump(fd, args); 431 } 432 433 // dump record threads 434 for (size_t i = 0; i < mRecordThreads.size(); i++) { 435 mRecordThreads.valueAt(i)->dump(fd, args); 436 } 437 438 // dump orphan effect chains 439 if (mOrphanEffectChains.size() != 0) { 440 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 441 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 442 mOrphanEffectChains.valueAt(i)->dump(fd, args); 443 } 444 } 445 // dump all hardware devs 446 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 447 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 448 dev->dump(dev, fd); 449 } 450 451#ifdef TEE_SINK 452 // dump the serially shared record tee sink 453 if (mRecordTeeSource != 0) { 454 dumpTee(fd, mRecordTeeSource); 455 } 456#endif 457 458 if (locked) { 459 mLock.unlock(); 460 } 461 462 // append a copy of media.log here by forwarding fd to it, but don't attempt 463 // to lookup the service if it's not running, as it will block for a second 464 if (mLogMemoryDealer != 0) { 465 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 466 if (binder != 0) { 467 dprintf(fd, "\nmedia.log:\n"); 468 Vector<String16> args; 469 binder->dump(fd, args); 470 } 471 } 472 473 // check for optional arguments 474 bool dumpMem = false; 475 bool unreachableMemory = false; 476 for (const auto &arg : args) { 477 if (arg == String16("-m")) { 478 dumpMem = true; 479 } else if (arg == String16("--unreachable")) { 480 unreachableMemory = true; 481 } 482 } 483 484 if (dumpMem) { 485 dprintf(fd, "\nDumping memory:\n"); 486 std::string s = dumpMemoryAddresses(100 /* limit */); 487 write(fd, s.c_str(), s.size()); 488 } 489 if (unreachableMemory) { 490 dprintf(fd, "\nDumping unreachable memory:\n"); 491 // TODO - should limit be an argument parameter? 492 std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */); 493 write(fd, s.c_str(), s.size()); 494 } 495 } 496 return NO_ERROR; 497} 498 499sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 500{ 501 Mutex::Autolock _cl(mClientLock); 502 // If pid is already in the mClients wp<> map, then use that entry 503 // (for which promote() is always != 0), otherwise create a new entry and Client. 504 sp<Client> client = mClients.valueFor(pid).promote(); 505 if (client == 0) { 506 client = new Client(this, pid); 507 mClients.add(pid, client); 508 } 509 510 return client; 511} 512 513sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 514{ 515 // If there is no memory allocated for logs, return a dummy writer that does nothing 516 if (mLogMemoryDealer == 0) { 517 return new NBLog::Writer(); 518 } 519 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 520 // Similarly if we can't contact the media.log service, also return a dummy writer 521 if (binder == 0) { 522 return new NBLog::Writer(); 523 } 524 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 525 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 526 // If allocation fails, consult the vector of previously unregistered writers 527 // and garbage-collect one or more them until an allocation succeeds 528 if (shared == 0) { 529 Mutex::Autolock _l(mUnregisteredWritersLock); 530 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 531 { 532 // Pick the oldest stale writer to garbage-collect 533 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 534 mUnregisteredWriters.removeAt(0); 535 mediaLogService->unregisterWriter(iMemory); 536 // Now the media.log remote reference to IMemory is gone. When our last local 537 // reference to IMemory also drops to zero at end of this block, 538 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 539 } 540 // Re-attempt the allocation 541 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 542 if (shared != 0) { 543 goto success; 544 } 545 } 546 // Even after garbage-collecting all old writers, there is still not enough memory, 547 // so return a dummy writer 548 return new NBLog::Writer(); 549 } 550success: 551 mediaLogService->registerWriter(shared, size, name); 552 return new NBLog::Writer(size, shared); 553} 554 555void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 556{ 557 if (writer == 0) { 558 return; 559 } 560 sp<IMemory> iMemory(writer->getIMemory()); 561 if (iMemory == 0) { 562 return; 563 } 564 // Rather than removing the writer immediately, append it to a queue of old writers to 565 // be garbage-collected later. This allows us to continue to view old logs for a while. 566 Mutex::Autolock _l(mUnregisteredWritersLock); 567 mUnregisteredWriters.push(writer); 568} 569 570// IAudioFlinger interface 571 572 573sp<IAudioTrack> AudioFlinger::createTrack( 574 audio_stream_type_t streamType, 575 uint32_t sampleRate, 576 audio_format_t format, 577 audio_channel_mask_t channelMask, 578 size_t *frameCount, 579 IAudioFlinger::track_flags_t *flags, 580 const sp<IMemory>& sharedBuffer, 581 audio_io_handle_t output, 582 pid_t pid, 583 pid_t tid, 584 audio_session_t *sessionId, 585 int clientUid, 586 status_t *status) 587{ 588 sp<PlaybackThread::Track> track; 589 sp<TrackHandle> trackHandle; 590 sp<Client> client; 591 status_t lStatus; 592 audio_session_t lSessionId; 593 594 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 595 if (pid == -1 || !isTrustedCallingUid(callingUid)) { 596 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 597 ALOGW_IF(pid != -1 && pid != callingPid, 598 "%s uid %d pid %d tried to pass itself off as pid %d", 599 __func__, callingUid, callingPid, pid); 600 pid = callingPid; 601 } 602 603 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 604 // but if someone uses binder directly they could bypass that and cause us to crash 605 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 606 ALOGE("createTrack() invalid stream type %d", streamType); 607 lStatus = BAD_VALUE; 608 goto Exit; 609 } 610 611 // further sample rate checks are performed by createTrack_l() depending on the thread type 612 if (sampleRate == 0) { 613 ALOGE("createTrack() invalid sample rate %u", sampleRate); 614 lStatus = BAD_VALUE; 615 goto Exit; 616 } 617 618 // further channel mask checks are performed by createTrack_l() depending on the thread type 619 if (!audio_is_output_channel(channelMask)) { 620 ALOGE("createTrack() invalid channel mask %#x", channelMask); 621 lStatus = BAD_VALUE; 622 goto Exit; 623 } 624 625 // further format checks are performed by createTrack_l() depending on the thread type 626 if (!audio_is_valid_format(format)) { 627 ALOGE("createTrack() invalid format %#x", format); 628 lStatus = BAD_VALUE; 629 goto Exit; 630 } 631 632 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 633 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 634 lStatus = BAD_VALUE; 635 goto Exit; 636 } 637 638 { 639 Mutex::Autolock _l(mLock); 640 PlaybackThread *thread = checkPlaybackThread_l(output); 641 if (thread == NULL) { 642 ALOGE("no playback thread found for output handle %d", output); 643 lStatus = BAD_VALUE; 644 goto Exit; 645 } 646 647 client = registerPid(pid); 648 649 PlaybackThread *effectThread = NULL; 650 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 651 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 652 ALOGE("createTrack() invalid session ID %d", *sessionId); 653 lStatus = BAD_VALUE; 654 goto Exit; 655 } 656 lSessionId = *sessionId; 657 // check if an effect chain with the same session ID is present on another 658 // output thread and move it here. 659 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 660 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 661 if (mPlaybackThreads.keyAt(i) != output) { 662 uint32_t sessions = t->hasAudioSession(lSessionId); 663 if (sessions & PlaybackThread::EFFECT_SESSION) { 664 effectThread = t.get(); 665 break; 666 } 667 } 668 } 669 } else { 670 // if no audio session id is provided, create one here 671 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 672 if (sessionId != NULL) { 673 *sessionId = lSessionId; 674 } 675 } 676 ALOGV("createTrack() lSessionId: %d", lSessionId); 677 678 track = thread->createTrack_l(client, streamType, sampleRate, format, 679 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 680 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 681 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 682 683 // move effect chain to this output thread if an effect on same session was waiting 684 // for a track to be created 685 if (lStatus == NO_ERROR && effectThread != NULL) { 686 // no risk of deadlock because AudioFlinger::mLock is held 687 Mutex::Autolock _dl(thread->mLock); 688 Mutex::Autolock _sl(effectThread->mLock); 689 moveEffectChain_l(lSessionId, effectThread, thread, true); 690 } 691 692 // Look for sync events awaiting for a session to be used. 693 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 694 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 695 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 696 if (lStatus == NO_ERROR) { 697 (void) track->setSyncEvent(mPendingSyncEvents[i]); 698 } else { 699 mPendingSyncEvents[i]->cancel(); 700 } 701 mPendingSyncEvents.removeAt(i); 702 i--; 703 } 704 } 705 } 706 707 setAudioHwSyncForSession_l(thread, lSessionId); 708 } 709 710 if (lStatus != NO_ERROR) { 711 // remove local strong reference to Client before deleting the Track so that the 712 // Client destructor is called by the TrackBase destructor with mClientLock held 713 // Don't hold mClientLock when releasing the reference on the track as the 714 // destructor will acquire it. 715 { 716 Mutex::Autolock _cl(mClientLock); 717 client.clear(); 718 } 719 track.clear(); 720 goto Exit; 721 } 722 723 // return handle to client 724 trackHandle = new TrackHandle(track); 725 726Exit: 727 *status = lStatus; 728 return trackHandle; 729} 730 731uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 732{ 733 Mutex::Autolock _l(mLock); 734 ThreadBase *thread = checkThread_l(ioHandle); 735 if (thread == NULL) { 736 ALOGW("sampleRate() unknown thread %d", ioHandle); 737 return 0; 738 } 739 return thread->sampleRate(); 740} 741 742audio_format_t AudioFlinger::format(audio_io_handle_t output) const 743{ 744 Mutex::Autolock _l(mLock); 745 PlaybackThread *thread = checkPlaybackThread_l(output); 746 if (thread == NULL) { 747 ALOGW("format() unknown thread %d", output); 748 return AUDIO_FORMAT_INVALID; 749 } 750 return thread->format(); 751} 752 753size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 754{ 755 Mutex::Autolock _l(mLock); 756 ThreadBase *thread = checkThread_l(ioHandle); 757 if (thread == NULL) { 758 ALOGW("frameCount() unknown thread %d", ioHandle); 759 return 0; 760 } 761 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 762 // should examine all callers and fix them to handle smaller counts 763 return thread->frameCount(); 764} 765 766size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const 767{ 768 Mutex::Autolock _l(mLock); 769 ThreadBase *thread = checkThread_l(ioHandle); 770 if (thread == NULL) { 771 ALOGW("frameCountHAL() unknown thread %d", ioHandle); 772 return 0; 773 } 774 return thread->frameCountHAL(); 775} 776 777uint32_t AudioFlinger::latency(audio_io_handle_t output) const 778{ 779 Mutex::Autolock _l(mLock); 780 PlaybackThread *thread = checkPlaybackThread_l(output); 781 if (thread == NULL) { 782 ALOGW("latency(): no playback thread found for output handle %d", output); 783 return 0; 784 } 785 return thread->latency(); 786} 787 788status_t AudioFlinger::setMasterVolume(float value) 789{ 790 status_t ret = initCheck(); 791 if (ret != NO_ERROR) { 792 return ret; 793 } 794 795 // check calling permissions 796 if (!settingsAllowed()) { 797 return PERMISSION_DENIED; 798 } 799 800 Mutex::Autolock _l(mLock); 801 mMasterVolume = value; 802 803 // Set master volume in the HALs which support it. 804 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 805 AutoMutex lock(mHardwareLock); 806 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 807 808 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 809 if (dev->canSetMasterVolume()) { 810 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 811 } 812 mHardwareStatus = AUDIO_HW_IDLE; 813 } 814 815 // Now set the master volume in each playback thread. Playback threads 816 // assigned to HALs which do not have master volume support will apply 817 // master volume during the mix operation. Threads with HALs which do 818 // support master volume will simply ignore the setting. 819 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 820 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 821 continue; 822 } 823 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 824 } 825 826 return NO_ERROR; 827} 828 829status_t AudioFlinger::setMode(audio_mode_t mode) 830{ 831 status_t ret = initCheck(); 832 if (ret != NO_ERROR) { 833 return ret; 834 } 835 836 // check calling permissions 837 if (!settingsAllowed()) { 838 return PERMISSION_DENIED; 839 } 840 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 841 ALOGW("Illegal value: setMode(%d)", mode); 842 return BAD_VALUE; 843 } 844 845 { // scope for the lock 846 AutoMutex lock(mHardwareLock); 847 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 848 mHardwareStatus = AUDIO_HW_SET_MODE; 849 ret = dev->set_mode(dev, mode); 850 mHardwareStatus = AUDIO_HW_IDLE; 851 } 852 853 if (NO_ERROR == ret) { 854 Mutex::Autolock _l(mLock); 855 mMode = mode; 856 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 857 mPlaybackThreads.valueAt(i)->setMode(mode); 858 } 859 860 return ret; 861} 862 863status_t AudioFlinger::setMicMute(bool state) 864{ 865 status_t ret = initCheck(); 866 if (ret != NO_ERROR) { 867 return ret; 868 } 869 870 // check calling permissions 871 if (!settingsAllowed()) { 872 return PERMISSION_DENIED; 873 } 874 875 AutoMutex lock(mHardwareLock); 876 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 877 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 878 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 879 status_t result = dev->set_mic_mute(dev, state); 880 if (result != NO_ERROR) { 881 ret = result; 882 } 883 } 884 mHardwareStatus = AUDIO_HW_IDLE; 885 return ret; 886} 887 888bool AudioFlinger::getMicMute() const 889{ 890 status_t ret = initCheck(); 891 if (ret != NO_ERROR) { 892 return false; 893 } 894 bool mute = true; 895 bool state = AUDIO_MODE_INVALID; 896 AutoMutex lock(mHardwareLock); 897 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 898 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 899 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 900 status_t result = dev->get_mic_mute(dev, &state); 901 if (result == NO_ERROR) { 902 mute = mute && state; 903 } 904 } 905 mHardwareStatus = AUDIO_HW_IDLE; 906 907 return mute; 908} 909 910status_t AudioFlinger::setMasterMute(bool muted) 911{ 912 status_t ret = initCheck(); 913 if (ret != NO_ERROR) { 914 return ret; 915 } 916 917 // check calling permissions 918 if (!settingsAllowed()) { 919 return PERMISSION_DENIED; 920 } 921 922 Mutex::Autolock _l(mLock); 923 mMasterMute = muted; 924 925 // Set master mute in the HALs which support it. 926 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 927 AutoMutex lock(mHardwareLock); 928 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 929 930 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 931 if (dev->canSetMasterMute()) { 932 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 933 } 934 mHardwareStatus = AUDIO_HW_IDLE; 935 } 936 937 // Now set the master mute in each playback thread. Playback threads 938 // assigned to HALs which do not have master mute support will apply master 939 // mute during the mix operation. Threads with HALs which do support master 940 // mute will simply ignore the setting. 941 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 942 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 943 continue; 944 } 945 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 946 } 947 948 return NO_ERROR; 949} 950 951float AudioFlinger::masterVolume() const 952{ 953 Mutex::Autolock _l(mLock); 954 return masterVolume_l(); 955} 956 957bool AudioFlinger::masterMute() const 958{ 959 Mutex::Autolock _l(mLock); 960 return masterMute_l(); 961} 962 963float AudioFlinger::masterVolume_l() const 964{ 965 return mMasterVolume; 966} 967 968bool AudioFlinger::masterMute_l() const 969{ 970 return mMasterMute; 971} 972 973status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 974{ 975 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 976 ALOGW("setStreamVolume() invalid stream %d", stream); 977 return BAD_VALUE; 978 } 979 pid_t caller = IPCThreadState::self()->getCallingPid(); 980 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 981 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 982 return PERMISSION_DENIED; 983 } 984 985 return NO_ERROR; 986} 987 988status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 989 audio_io_handle_t output) 990{ 991 // check calling permissions 992 if (!settingsAllowed()) { 993 return PERMISSION_DENIED; 994 } 995 996 status_t status = checkStreamType(stream); 997 if (status != NO_ERROR) { 998 return status; 999 } 1000 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 1001 1002 AutoMutex lock(mLock); 1003 PlaybackThread *thread = NULL; 1004 if (output != AUDIO_IO_HANDLE_NONE) { 1005 thread = checkPlaybackThread_l(output); 1006 if (thread == NULL) { 1007 return BAD_VALUE; 1008 } 1009 } 1010 1011 mStreamTypes[stream].volume = value; 1012 1013 if (thread == NULL) { 1014 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1015 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 1016 } 1017 } else { 1018 thread->setStreamVolume(stream, value); 1019 } 1020 1021 return NO_ERROR; 1022} 1023 1024status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 1025{ 1026 // check calling permissions 1027 if (!settingsAllowed()) { 1028 return PERMISSION_DENIED; 1029 } 1030 1031 status_t status = checkStreamType(stream); 1032 if (status != NO_ERROR) { 1033 return status; 1034 } 1035 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 1036 1037 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 1038 ALOGE("setStreamMute() invalid stream %d", stream); 1039 return BAD_VALUE; 1040 } 1041 1042 AutoMutex lock(mLock); 1043 mStreamTypes[stream].mute = muted; 1044 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 1045 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 1046 1047 return NO_ERROR; 1048} 1049 1050float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1051{ 1052 status_t status = checkStreamType(stream); 1053 if (status != NO_ERROR) { 1054 return 0.0f; 1055 } 1056 1057 AutoMutex lock(mLock); 1058 float volume; 1059 if (output != AUDIO_IO_HANDLE_NONE) { 1060 PlaybackThread *thread = checkPlaybackThread_l(output); 1061 if (thread == NULL) { 1062 return 0.0f; 1063 } 1064 volume = thread->streamVolume(stream); 1065 } else { 1066 volume = streamVolume_l(stream); 1067 } 1068 1069 return volume; 1070} 1071 1072bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1073{ 1074 status_t status = checkStreamType(stream); 1075 if (status != NO_ERROR) { 1076 return true; 1077 } 1078 1079 AutoMutex lock(mLock); 1080 return streamMute_l(stream); 1081} 1082 1083 1084void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1085{ 1086 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1087 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1088 } 1089} 1090 1091status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1092{ 1093 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1094 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1095 1096 // check calling permissions 1097 if (!settingsAllowed()) { 1098 return PERMISSION_DENIED; 1099 } 1100 1101 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1102 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1103 Mutex::Autolock _l(mLock); 1104 status_t final_result = NO_ERROR; 1105 { 1106 AutoMutex lock(mHardwareLock); 1107 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1108 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1109 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1110 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1111 final_result = result ?: final_result; 1112 } 1113 mHardwareStatus = AUDIO_HW_IDLE; 1114 } 1115 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1116 AudioParameter param = AudioParameter(keyValuePairs); 1117 String8 value; 1118 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1119 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1120 if (mBtNrecIsOff != btNrecIsOff) { 1121 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1122 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1123 audio_devices_t device = thread->inDevice(); 1124 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1125 // collect all of the thread's session IDs 1126 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1127 // suspend effects associated with those session IDs 1128 for (size_t j = 0; j < ids.size(); ++j) { 1129 audio_session_t sessionId = ids.keyAt(j); 1130 thread->setEffectSuspended(FX_IID_AEC, 1131 suspend, 1132 sessionId); 1133 thread->setEffectSuspended(FX_IID_NS, 1134 suspend, 1135 sessionId); 1136 } 1137 } 1138 mBtNrecIsOff = btNrecIsOff; 1139 } 1140 } 1141 String8 screenState; 1142 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1143 bool isOff = screenState == "off"; 1144 if (isOff != (AudioFlinger::mScreenState & 1)) { 1145 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1146 } 1147 } 1148 return final_result; 1149 } 1150 1151 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1152 // and the thread is exited once the lock is released 1153 sp<ThreadBase> thread; 1154 { 1155 Mutex::Autolock _l(mLock); 1156 thread = checkPlaybackThread_l(ioHandle); 1157 if (thread == 0) { 1158 thread = checkRecordThread_l(ioHandle); 1159 } else if (thread == primaryPlaybackThread_l()) { 1160 // indicate output device change to all input threads for pre processing 1161 AudioParameter param = AudioParameter(keyValuePairs); 1162 int value; 1163 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1164 (value != 0)) { 1165 broacastParametersToRecordThreads_l(keyValuePairs); 1166 } 1167 } 1168 } 1169 if (thread != 0) { 1170 return thread->setParameters(keyValuePairs); 1171 } 1172 return BAD_VALUE; 1173} 1174 1175String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1176{ 1177 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1178 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1179 1180 Mutex::Autolock _l(mLock); 1181 1182 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1183 String8 out_s8; 1184 1185 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1186 char *s; 1187 { 1188 AutoMutex lock(mHardwareLock); 1189 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1190 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1191 s = dev->get_parameters(dev, keys.string()); 1192 mHardwareStatus = AUDIO_HW_IDLE; 1193 } 1194 out_s8 += String8(s ? s : ""); 1195 free(s); 1196 } 1197 return out_s8; 1198 } 1199 1200 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1201 if (playbackThread != NULL) { 1202 return playbackThread->getParameters(keys); 1203 } 1204 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1205 if (recordThread != NULL) { 1206 return recordThread->getParameters(keys); 1207 } 1208 return String8(""); 1209} 1210 1211size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1212 audio_channel_mask_t channelMask) const 1213{ 1214 status_t ret = initCheck(); 1215 if (ret != NO_ERROR) { 1216 return 0; 1217 } 1218 if ((sampleRate == 0) || 1219 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1220 !audio_is_input_channel(channelMask)) { 1221 return 0; 1222 } 1223 1224 AutoMutex lock(mHardwareLock); 1225 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1226 audio_config_t config, proposed; 1227 memset(&proposed, 0, sizeof(proposed)); 1228 proposed.sample_rate = sampleRate; 1229 proposed.channel_mask = channelMask; 1230 proposed.format = format; 1231 1232 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1233 size_t frames; 1234 for (;;) { 1235 // Note: config is currently a const parameter for get_input_buffer_size() 1236 // but we use a copy from proposed in case config changes from the call. 1237 config = proposed; 1238 frames = dev->get_input_buffer_size(dev, &config); 1239 if (frames != 0) { 1240 break; // hal success, config is the result 1241 } 1242 // change one parameter of the configuration each iteration to a more "common" value 1243 // to see if the device will support it. 1244 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1245 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1246 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1247 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1248 } else { 1249 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1250 "format %#x, channelMask 0x%X", 1251 sampleRate, format, channelMask); 1252 break; // retries failed, break out of loop with frames == 0. 1253 } 1254 } 1255 mHardwareStatus = AUDIO_HW_IDLE; 1256 if (frames > 0 && config.sample_rate != sampleRate) { 1257 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1258 } 1259 return frames; // may be converted to bytes at the Java level. 1260} 1261 1262uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1263{ 1264 Mutex::Autolock _l(mLock); 1265 1266 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1267 if (recordThread != NULL) { 1268 return recordThread->getInputFramesLost(); 1269 } 1270 return 0; 1271} 1272 1273status_t AudioFlinger::setVoiceVolume(float value) 1274{ 1275 status_t ret = initCheck(); 1276 if (ret != NO_ERROR) { 1277 return ret; 1278 } 1279 1280 // check calling permissions 1281 if (!settingsAllowed()) { 1282 return PERMISSION_DENIED; 1283 } 1284 1285 AutoMutex lock(mHardwareLock); 1286 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1287 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1288 ret = dev->set_voice_volume(dev, value); 1289 mHardwareStatus = AUDIO_HW_IDLE; 1290 1291 return ret; 1292} 1293 1294status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1295 audio_io_handle_t output) const 1296{ 1297 Mutex::Autolock _l(mLock); 1298 1299 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1300 if (playbackThread != NULL) { 1301 return playbackThread->getRenderPosition(halFrames, dspFrames); 1302 } 1303 1304 return BAD_VALUE; 1305} 1306 1307void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1308{ 1309 Mutex::Autolock _l(mLock); 1310 if (client == 0) { 1311 return; 1312 } 1313 pid_t pid = IPCThreadState::self()->getCallingPid(); 1314 { 1315 Mutex::Autolock _cl(mClientLock); 1316 if (mNotificationClients.indexOfKey(pid) < 0) { 1317 sp<NotificationClient> notificationClient = new NotificationClient(this, 1318 client, 1319 pid); 1320 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1321 1322 mNotificationClients.add(pid, notificationClient); 1323 1324 sp<IBinder> binder = IInterface::asBinder(client); 1325 binder->linkToDeath(notificationClient); 1326 } 1327 } 1328 1329 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1330 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1331 // the config change is always sent from playback or record threads to avoid deadlock 1332 // with AudioSystem::gLock 1333 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1334 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1335 } 1336 1337 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1338 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1339 } 1340} 1341 1342void AudioFlinger::removeNotificationClient(pid_t pid) 1343{ 1344 Mutex::Autolock _l(mLock); 1345 { 1346 Mutex::Autolock _cl(mClientLock); 1347 mNotificationClients.removeItem(pid); 1348 } 1349 1350 ALOGV("%d died, releasing its sessions", pid); 1351 size_t num = mAudioSessionRefs.size(); 1352 bool removed = false; 1353 for (size_t i = 0; i< num; ) { 1354 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1355 ALOGV(" pid %d @ %zu", ref->mPid, i); 1356 if (ref->mPid == pid) { 1357 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1358 mAudioSessionRefs.removeAt(i); 1359 delete ref; 1360 removed = true; 1361 num--; 1362 } else { 1363 i++; 1364 } 1365 } 1366 if (removed) { 1367 purgeStaleEffects_l(); 1368 } 1369} 1370 1371void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1372 const sp<AudioIoDescriptor>& ioDesc, 1373 pid_t pid) 1374{ 1375 Mutex::Autolock _l(mClientLock); 1376 size_t size = mNotificationClients.size(); 1377 for (size_t i = 0; i < size; i++) { 1378 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1379 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1380 } 1381 } 1382} 1383 1384// removeClient_l() must be called with AudioFlinger::mClientLock held 1385void AudioFlinger::removeClient_l(pid_t pid) 1386{ 1387 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1388 IPCThreadState::self()->getCallingPid()); 1389 mClients.removeItem(pid); 1390} 1391 1392// getEffectThread_l() must be called with AudioFlinger::mLock held 1393sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1394 int EffectId) 1395{ 1396 sp<PlaybackThread> thread; 1397 1398 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1399 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1400 ALOG_ASSERT(thread == 0); 1401 thread = mPlaybackThreads.valueAt(i); 1402 } 1403 } 1404 1405 return thread; 1406} 1407 1408 1409 1410// ---------------------------------------------------------------------------- 1411 1412AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1413 : RefBase(), 1414 mAudioFlinger(audioFlinger), 1415 mPid(pid) 1416{ 1417 size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0); 1418 heapSize *= 1024; 1419 if (!heapSize) { 1420 heapSize = kClientSharedHeapSizeBytes; 1421 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1422 // invalidated tracks 1423 if (!audioFlinger->isLowRamDevice()) { 1424 heapSize *= kClientSharedHeapSizeMultiplier; 1425 } 1426 } 1427 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1428} 1429 1430// Client destructor must be called with AudioFlinger::mClientLock held 1431AudioFlinger::Client::~Client() 1432{ 1433 mAudioFlinger->removeClient_l(mPid); 1434} 1435 1436sp<MemoryDealer> AudioFlinger::Client::heap() const 1437{ 1438 return mMemoryDealer; 1439} 1440 1441// ---------------------------------------------------------------------------- 1442 1443AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1444 const sp<IAudioFlingerClient>& client, 1445 pid_t pid) 1446 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1447{ 1448} 1449 1450AudioFlinger::NotificationClient::~NotificationClient() 1451{ 1452} 1453 1454void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1455{ 1456 sp<NotificationClient> keep(this); 1457 mAudioFlinger->removeNotificationClient(mPid); 1458} 1459 1460 1461// ---------------------------------------------------------------------------- 1462 1463sp<IAudioRecord> AudioFlinger::openRecord( 1464 audio_io_handle_t input, 1465 uint32_t sampleRate, 1466 audio_format_t format, 1467 audio_channel_mask_t channelMask, 1468 const String16& opPackageName, 1469 size_t *frameCount, 1470 IAudioFlinger::track_flags_t *flags, 1471 pid_t pid, 1472 pid_t tid, 1473 int clientUid, 1474 audio_session_t *sessionId, 1475 size_t *notificationFrames, 1476 sp<IMemory>& cblk, 1477 sp<IMemory>& buffers, 1478 status_t *status) 1479{ 1480 sp<RecordThread::RecordTrack> recordTrack; 1481 sp<RecordHandle> recordHandle; 1482 sp<Client> client; 1483 status_t lStatus; 1484 audio_session_t lSessionId; 1485 1486 cblk.clear(); 1487 buffers.clear(); 1488 1489 bool updatePid = (pid == -1); 1490 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1491 if (!isTrustedCallingUid(callingUid)) { 1492 ALOGW_IF((uid_t)clientUid != callingUid, 1493 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1494 clientUid = callingUid; 1495 updatePid = true; 1496 } 1497 1498 if (updatePid) { 1499 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1500 ALOGW_IF(pid != -1 && pid != callingPid, 1501 "%s uid %d pid %d tried to pass itself off as pid %d", 1502 __func__, callingUid, callingPid, pid); 1503 pid = callingPid; 1504 } 1505 1506 // check calling permissions 1507 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1508 ALOGE("openRecord() permission denied: recording not allowed"); 1509 lStatus = PERMISSION_DENIED; 1510 goto Exit; 1511 } 1512 1513 // further sample rate checks are performed by createRecordTrack_l() 1514 if (sampleRate == 0) { 1515 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1516 lStatus = BAD_VALUE; 1517 goto Exit; 1518 } 1519 1520 // we don't yet support anything other than linear PCM 1521 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1522 ALOGE("openRecord() invalid format %#x", format); 1523 lStatus = BAD_VALUE; 1524 goto Exit; 1525 } 1526 1527 // further channel mask checks are performed by createRecordTrack_l() 1528 if (!audio_is_input_channel(channelMask)) { 1529 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1530 lStatus = BAD_VALUE; 1531 goto Exit; 1532 } 1533 1534 { 1535 Mutex::Autolock _l(mLock); 1536 RecordThread *thread = checkRecordThread_l(input); 1537 if (thread == NULL) { 1538 ALOGE("openRecord() checkRecordThread_l failed"); 1539 lStatus = BAD_VALUE; 1540 goto Exit; 1541 } 1542 1543 client = registerPid(pid); 1544 1545 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1546 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1547 lStatus = BAD_VALUE; 1548 goto Exit; 1549 } 1550 lSessionId = *sessionId; 1551 } else { 1552 // if no audio session id is provided, create one here 1553 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1554 if (sessionId != NULL) { 1555 *sessionId = lSessionId; 1556 } 1557 } 1558 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1559 1560 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1561 frameCount, lSessionId, notificationFrames, 1562 clientUid, flags, tid, &lStatus); 1563 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1564 1565 if (lStatus == NO_ERROR) { 1566 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1567 // session and move it to this thread. 1568 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1569 if (chain != 0) { 1570 Mutex::Autolock _l(thread->mLock); 1571 thread->addEffectChain_l(chain); 1572 } 1573 } 1574 } 1575 1576 if (lStatus != NO_ERROR) { 1577 // remove local strong reference to Client before deleting the RecordTrack so that the 1578 // Client destructor is called by the TrackBase destructor with mClientLock held 1579 // Don't hold mClientLock when releasing the reference on the track as the 1580 // destructor will acquire it. 1581 { 1582 Mutex::Autolock _cl(mClientLock); 1583 client.clear(); 1584 } 1585 recordTrack.clear(); 1586 goto Exit; 1587 } 1588 1589 cblk = recordTrack->getCblk(); 1590 buffers = recordTrack->getBuffers(); 1591 1592 // return handle to client 1593 recordHandle = new RecordHandle(recordTrack); 1594 1595Exit: 1596 *status = lStatus; 1597 return recordHandle; 1598} 1599 1600 1601 1602// ---------------------------------------------------------------------------- 1603 1604audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1605{ 1606 if (name == NULL) { 1607 return AUDIO_MODULE_HANDLE_NONE; 1608 } 1609 if (!settingsAllowed()) { 1610 return AUDIO_MODULE_HANDLE_NONE; 1611 } 1612 Mutex::Autolock _l(mLock); 1613 return loadHwModule_l(name); 1614} 1615 1616// loadHwModule_l() must be called with AudioFlinger::mLock held 1617audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1618{ 1619 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1620 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1621 ALOGW("loadHwModule() module %s already loaded", name); 1622 return mAudioHwDevs.keyAt(i); 1623 } 1624 } 1625 1626 audio_hw_device_t *dev; 1627 1628 int rc = load_audio_interface(name, &dev); 1629 if (rc) { 1630 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1631 return AUDIO_MODULE_HANDLE_NONE; 1632 } 1633 1634 mHardwareStatus = AUDIO_HW_INIT; 1635 rc = dev->init_check(dev); 1636 mHardwareStatus = AUDIO_HW_IDLE; 1637 if (rc) { 1638 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1639 return AUDIO_MODULE_HANDLE_NONE; 1640 } 1641 1642 // Check and cache this HAL's level of support for master mute and master 1643 // volume. If this is the first HAL opened, and it supports the get 1644 // methods, use the initial values provided by the HAL as the current 1645 // master mute and volume settings. 1646 1647 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1648 { // scope for auto-lock pattern 1649 AutoMutex lock(mHardwareLock); 1650 1651 if (0 == mAudioHwDevs.size()) { 1652 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1653 if (NULL != dev->get_master_volume) { 1654 float mv; 1655 if (OK == dev->get_master_volume(dev, &mv)) { 1656 mMasterVolume = mv; 1657 } 1658 } 1659 1660 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1661 if (NULL != dev->get_master_mute) { 1662 bool mm; 1663 if (OK == dev->get_master_mute(dev, &mm)) { 1664 mMasterMute = mm; 1665 } 1666 } 1667 } 1668 1669 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1670 if ((NULL != dev->set_master_volume) && 1671 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1672 flags = static_cast<AudioHwDevice::Flags>(flags | 1673 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1674 } 1675 1676 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1677 if ((NULL != dev->set_master_mute) && 1678 (OK == dev->set_master_mute(dev, mMasterMute))) { 1679 flags = static_cast<AudioHwDevice::Flags>(flags | 1680 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1681 } 1682 1683 mHardwareStatus = AUDIO_HW_IDLE; 1684 } 1685 1686 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1687 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1688 1689 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1690 name, dev->common.module->name, dev->common.module->id, handle); 1691 1692 return handle; 1693 1694} 1695 1696// ---------------------------------------------------------------------------- 1697 1698uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1699{ 1700 Mutex::Autolock _l(mLock); 1701 PlaybackThread *thread = primaryPlaybackThread_l(); 1702 return thread != NULL ? thread->sampleRate() : 0; 1703} 1704 1705size_t AudioFlinger::getPrimaryOutputFrameCount() 1706{ 1707 Mutex::Autolock _l(mLock); 1708 PlaybackThread *thread = primaryPlaybackThread_l(); 1709 return thread != NULL ? thread->frameCountHAL() : 0; 1710} 1711 1712// ---------------------------------------------------------------------------- 1713 1714status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1715{ 1716 uid_t uid = IPCThreadState::self()->getCallingUid(); 1717 if (uid != AID_SYSTEM) { 1718 return PERMISSION_DENIED; 1719 } 1720 Mutex::Autolock _l(mLock); 1721 if (mIsDeviceTypeKnown) { 1722 return INVALID_OPERATION; 1723 } 1724 mIsLowRamDevice = isLowRamDevice; 1725 mIsDeviceTypeKnown = true; 1726 return NO_ERROR; 1727} 1728 1729audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1730{ 1731 Mutex::Autolock _l(mLock); 1732 1733 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1734 if (index >= 0) { 1735 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1736 mHwAvSyncIds.valueAt(index), sessionId); 1737 return mHwAvSyncIds.valueAt(index); 1738 } 1739 1740 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1741 if (dev == NULL) { 1742 return AUDIO_HW_SYNC_INVALID; 1743 } 1744 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1745 AudioParameter param = AudioParameter(String8(reply)); 1746 free(reply); 1747 1748 int value; 1749 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1750 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1751 return AUDIO_HW_SYNC_INVALID; 1752 } 1753 1754 // allow only one session for a given HW A/V sync ID. 1755 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1756 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1757 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1758 value, mHwAvSyncIds.keyAt(i)); 1759 mHwAvSyncIds.removeItemsAt(i); 1760 break; 1761 } 1762 } 1763 1764 mHwAvSyncIds.add(sessionId, value); 1765 1766 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1767 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1768 uint32_t sessions = thread->hasAudioSession(sessionId); 1769 if (sessions & PlaybackThread::TRACK_SESSION) { 1770 AudioParameter param = AudioParameter(); 1771 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1772 thread->setParameters(param.toString()); 1773 break; 1774 } 1775 } 1776 1777 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1778 return (audio_hw_sync_t)value; 1779} 1780 1781status_t AudioFlinger::systemReady() 1782{ 1783 Mutex::Autolock _l(mLock); 1784 ALOGI("%s", __FUNCTION__); 1785 if (mSystemReady) { 1786 ALOGW("%s called twice", __FUNCTION__); 1787 return NO_ERROR; 1788 } 1789 mSystemReady = true; 1790 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1791 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1792 thread->systemReady(); 1793 } 1794 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1795 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1796 thread->systemReady(); 1797 } 1798 return NO_ERROR; 1799} 1800 1801// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1802void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1803{ 1804 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1805 if (index >= 0) { 1806 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1807 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1808 AudioParameter param = AudioParameter(); 1809 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1810 thread->setParameters(param.toString()); 1811 } 1812} 1813 1814 1815// ---------------------------------------------------------------------------- 1816 1817 1818sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1819 audio_io_handle_t *output, 1820 audio_config_t *config, 1821 audio_devices_t devices, 1822 const String8& address, 1823 audio_output_flags_t flags) 1824{ 1825 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1826 if (outHwDev == NULL) { 1827 return 0; 1828 } 1829 1830 if (*output == AUDIO_IO_HANDLE_NONE) { 1831 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1832 } else { 1833 // Audio Policy does not currently request a specific output handle. 1834 // If this is ever needed, see openInput_l() for example code. 1835 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1836 return 0; 1837 } 1838 1839 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1840 1841 // FOR TESTING ONLY: 1842 // This if statement allows overriding the audio policy settings 1843 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1844 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1845 // Check only for Normal Mixing mode 1846 if (kEnableExtendedPrecision) { 1847 // Specify format (uncomment one below to choose) 1848 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1849 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1850 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1851 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1852 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1853 } 1854 if (kEnableExtendedChannels) { 1855 // Specify channel mask (uncomment one below to choose) 1856 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1857 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1858 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1859 } 1860 } 1861 1862 AudioStreamOut *outputStream = NULL; 1863 status_t status = outHwDev->openOutputStream( 1864 &outputStream, 1865 *output, 1866 devices, 1867 flags, 1868 config, 1869 address.string()); 1870 1871 mHardwareStatus = AUDIO_HW_IDLE; 1872 1873 if (status == NO_ERROR) { 1874 1875 PlaybackThread *thread; 1876 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1877 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1878 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1879 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1880 || !isValidPcmSinkFormat(config->format) 1881 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1882 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1883 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1884 } else { 1885 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1886 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1887 } 1888 mPlaybackThreads.add(*output, thread); 1889 return thread; 1890 } 1891 1892 return 0; 1893} 1894 1895status_t AudioFlinger::openOutput(audio_module_handle_t module, 1896 audio_io_handle_t *output, 1897 audio_config_t *config, 1898 audio_devices_t *devices, 1899 const String8& address, 1900 uint32_t *latencyMs, 1901 audio_output_flags_t flags) 1902{ 1903 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1904 module, 1905 (devices != NULL) ? *devices : 0, 1906 config->sample_rate, 1907 config->format, 1908 config->channel_mask, 1909 flags); 1910 1911 if (*devices == AUDIO_DEVICE_NONE) { 1912 return BAD_VALUE; 1913 } 1914 1915 Mutex::Autolock _l(mLock); 1916 1917 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1918 if (thread != 0) { 1919 *latencyMs = thread->latency(); 1920 1921 // notify client processes of the new output creation 1922 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1923 1924 // the first primary output opened designates the primary hw device 1925 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1926 ALOGI("Using module %d has the primary audio interface", module); 1927 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1928 1929 AutoMutex lock(mHardwareLock); 1930 mHardwareStatus = AUDIO_HW_SET_MODE; 1931 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1932 mHardwareStatus = AUDIO_HW_IDLE; 1933 } 1934 return NO_ERROR; 1935 } 1936 1937 return NO_INIT; 1938} 1939 1940audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1941 audio_io_handle_t output2) 1942{ 1943 Mutex::Autolock _l(mLock); 1944 MixerThread *thread1 = checkMixerThread_l(output1); 1945 MixerThread *thread2 = checkMixerThread_l(output2); 1946 1947 if (thread1 == NULL || thread2 == NULL) { 1948 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1949 output2); 1950 return AUDIO_IO_HANDLE_NONE; 1951 } 1952 1953 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1954 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1955 thread->addOutputTrack(thread2); 1956 mPlaybackThreads.add(id, thread); 1957 // notify client processes of the new output creation 1958 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1959 return id; 1960} 1961 1962status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1963{ 1964 return closeOutput_nonvirtual(output); 1965} 1966 1967status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1968{ 1969 // keep strong reference on the playback thread so that 1970 // it is not destroyed while exit() is executed 1971 sp<PlaybackThread> thread; 1972 { 1973 Mutex::Autolock _l(mLock); 1974 thread = checkPlaybackThread_l(output); 1975 if (thread == NULL) { 1976 return BAD_VALUE; 1977 } 1978 1979 ALOGV("closeOutput() %d", output); 1980 1981 if (thread->type() == ThreadBase::MIXER) { 1982 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1983 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1984 DuplicatingThread *dupThread = 1985 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1986 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1987 } 1988 } 1989 } 1990 1991 1992 mPlaybackThreads.removeItem(output); 1993 // save all effects to the default thread 1994 if (mPlaybackThreads.size()) { 1995 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1996 if (dstThread != NULL) { 1997 // audioflinger lock is held here so the acquisition order of thread locks does not 1998 // matter 1999 Mutex::Autolock _dl(dstThread->mLock); 2000 Mutex::Autolock _sl(thread->mLock); 2001 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2002 for (size_t i = 0; i < effectChains.size(); i ++) { 2003 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 2004 } 2005 } 2006 } 2007 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2008 ioDesc->mIoHandle = output; 2009 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 2010 } 2011 thread->exit(); 2012 // The thread entity (active unit of execution) is no longer running here, 2013 // but the ThreadBase container still exists. 2014 2015 if (!thread->isDuplicating()) { 2016 closeOutputFinish(thread); 2017 } 2018 2019 return NO_ERROR; 2020} 2021 2022void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 2023{ 2024 AudioStreamOut *out = thread->clearOutput(); 2025 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 2026 // from now on thread->mOutput is NULL 2027 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 2028 delete out; 2029} 2030 2031void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 2032{ 2033 mPlaybackThreads.removeItem(thread->mId); 2034 thread->exit(); 2035 closeOutputFinish(thread); 2036} 2037 2038status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 2039{ 2040 Mutex::Autolock _l(mLock); 2041 PlaybackThread *thread = checkPlaybackThread_l(output); 2042 2043 if (thread == NULL) { 2044 return BAD_VALUE; 2045 } 2046 2047 ALOGV("suspendOutput() %d", output); 2048 thread->suspend(); 2049 2050 return NO_ERROR; 2051} 2052 2053status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2054{ 2055 Mutex::Autolock _l(mLock); 2056 PlaybackThread *thread = checkPlaybackThread_l(output); 2057 2058 if (thread == NULL) { 2059 return BAD_VALUE; 2060 } 2061 2062 ALOGV("restoreOutput() %d", output); 2063 2064 thread->restore(); 2065 2066 return NO_ERROR; 2067} 2068 2069status_t AudioFlinger::openInput(audio_module_handle_t module, 2070 audio_io_handle_t *input, 2071 audio_config_t *config, 2072 audio_devices_t *devices, 2073 const String8& address, 2074 audio_source_t source, 2075 audio_input_flags_t flags) 2076{ 2077 Mutex::Autolock _l(mLock); 2078 2079 if (*devices == AUDIO_DEVICE_NONE) { 2080 return BAD_VALUE; 2081 } 2082 2083 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2084 2085 if (thread != 0) { 2086 // notify client processes of the new input creation 2087 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2088 return NO_ERROR; 2089 } 2090 return NO_INIT; 2091} 2092 2093sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2094 audio_io_handle_t *input, 2095 audio_config_t *config, 2096 audio_devices_t devices, 2097 const String8& address, 2098 audio_source_t source, 2099 audio_input_flags_t flags) 2100{ 2101 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2102 if (inHwDev == NULL) { 2103 *input = AUDIO_IO_HANDLE_NONE; 2104 return 0; 2105 } 2106 2107 // Audio Policy can request a specific handle for hardware hotword. 2108 // The goal here is not to re-open an already opened input. 2109 // It is to use a pre-assigned I/O handle. 2110 if (*input == AUDIO_IO_HANDLE_NONE) { 2111 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2112 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2113 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2114 return 0; 2115 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2116 // This should not happen in a transient state with current design. 2117 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2118 return 0; 2119 } 2120 2121 audio_config_t halconfig = *config; 2122 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2123 audio_stream_in_t *inStream = NULL; 2124 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2125 &inStream, flags, address.string(), source); 2126 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2127 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2128 inStream, 2129 halconfig.sample_rate, 2130 halconfig.format, 2131 halconfig.channel_mask, 2132 flags, 2133 status, address.string()); 2134 2135 // If the input could not be opened with the requested parameters and we can handle the 2136 // conversion internally, try to open again with the proposed parameters. 2137 if (status == BAD_VALUE && 2138 audio_is_linear_pcm(config->format) && 2139 audio_is_linear_pcm(halconfig.format) && 2140 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2141 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) && 2142 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) { 2143 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2144 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2145 inStream = NULL; 2146 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2147 &inStream, flags, address.string(), source); 2148 // FIXME log this new status; HAL should not propose any further changes 2149 } 2150 2151 if (status == NO_ERROR && inStream != NULL) { 2152 2153#ifdef TEE_SINK 2154 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2155 // or (re-)create if current Pipe is idle and does not match the new format 2156 sp<NBAIO_Sink> teeSink; 2157 enum { 2158 TEE_SINK_NO, // don't copy input 2159 TEE_SINK_NEW, // copy input using a new pipe 2160 TEE_SINK_OLD, // copy input using an existing pipe 2161 } kind; 2162 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2163 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2164 if (!mTeeSinkInputEnabled) { 2165 kind = TEE_SINK_NO; 2166 } else if (!Format_isValid(format)) { 2167 kind = TEE_SINK_NO; 2168 } else if (mRecordTeeSink == 0) { 2169 kind = TEE_SINK_NEW; 2170 } else if (mRecordTeeSink->getStrongCount() != 1) { 2171 kind = TEE_SINK_NO; 2172 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2173 kind = TEE_SINK_OLD; 2174 } else { 2175 kind = TEE_SINK_NEW; 2176 } 2177 switch (kind) { 2178 case TEE_SINK_NEW: { 2179 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2180 size_t numCounterOffers = 0; 2181 const NBAIO_Format offers[1] = {format}; 2182 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2183 ALOG_ASSERT(index == 0); 2184 PipeReader *pipeReader = new PipeReader(*pipe); 2185 numCounterOffers = 0; 2186 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2187 ALOG_ASSERT(index == 0); 2188 mRecordTeeSink = pipe; 2189 mRecordTeeSource = pipeReader; 2190 teeSink = pipe; 2191 } 2192 break; 2193 case TEE_SINK_OLD: 2194 teeSink = mRecordTeeSink; 2195 break; 2196 case TEE_SINK_NO: 2197 default: 2198 break; 2199 } 2200#endif 2201 2202 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2203 2204 // Start record thread 2205 // RecordThread requires both input and output device indication to forward to audio 2206 // pre processing modules 2207 sp<RecordThread> thread = new RecordThread(this, 2208 inputStream, 2209 *input, 2210 primaryOutputDevice_l(), 2211 devices, 2212 mSystemReady 2213#ifdef TEE_SINK 2214 , teeSink 2215#endif 2216 ); 2217 mRecordThreads.add(*input, thread); 2218 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2219 return thread; 2220 } 2221 2222 *input = AUDIO_IO_HANDLE_NONE; 2223 return 0; 2224} 2225 2226status_t AudioFlinger::closeInput(audio_io_handle_t input) 2227{ 2228 return closeInput_nonvirtual(input); 2229} 2230 2231status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2232{ 2233 // keep strong reference on the record thread so that 2234 // it is not destroyed while exit() is executed 2235 sp<RecordThread> thread; 2236 { 2237 Mutex::Autolock _l(mLock); 2238 thread = checkRecordThread_l(input); 2239 if (thread == 0) { 2240 return BAD_VALUE; 2241 } 2242 2243 ALOGV("closeInput() %d", input); 2244 2245 // If we still have effect chains, it means that a client still holds a handle 2246 // on at least one effect. We must either move the chain to an existing thread with the 2247 // same session ID or put it aside in case a new record thread is opened for a 2248 // new capture on the same session 2249 sp<EffectChain> chain; 2250 { 2251 Mutex::Autolock _sl(thread->mLock); 2252 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2253 // Note: maximum one chain per record thread 2254 if (effectChains.size() != 0) { 2255 chain = effectChains[0]; 2256 } 2257 } 2258 if (chain != 0) { 2259 // first check if a record thread is already opened with a client on the same session. 2260 // This should only happen in case of overlap between one thread tear down and the 2261 // creation of its replacement 2262 size_t i; 2263 for (i = 0; i < mRecordThreads.size(); i++) { 2264 sp<RecordThread> t = mRecordThreads.valueAt(i); 2265 if (t == thread) { 2266 continue; 2267 } 2268 if (t->hasAudioSession(chain->sessionId()) != 0) { 2269 Mutex::Autolock _l(t->mLock); 2270 ALOGV("closeInput() found thread %d for effect session %d", 2271 t->id(), chain->sessionId()); 2272 t->addEffectChain_l(chain); 2273 break; 2274 } 2275 } 2276 // put the chain aside if we could not find a record thread with the same session id. 2277 if (i == mRecordThreads.size()) { 2278 putOrphanEffectChain_l(chain); 2279 } 2280 } 2281 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2282 ioDesc->mIoHandle = input; 2283 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2284 mRecordThreads.removeItem(input); 2285 } 2286 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2287 // we have a different lock for notification client 2288 closeInputFinish(thread); 2289 return NO_ERROR; 2290} 2291 2292void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2293{ 2294 thread->exit(); 2295 AudioStreamIn *in = thread->clearInput(); 2296 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2297 // from now on thread->mInput is NULL 2298 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2299 delete in; 2300} 2301 2302void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2303{ 2304 mRecordThreads.removeItem(thread->mId); 2305 closeInputFinish(thread); 2306} 2307 2308status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2309{ 2310 Mutex::Autolock _l(mLock); 2311 ALOGV("invalidateStream() stream %d", stream); 2312 2313 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2314 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2315 thread->invalidateTracks(stream); 2316 } 2317 2318 return NO_ERROR; 2319} 2320 2321 2322audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2323{ 2324 // This is a binder API, so a malicious client could pass in a bad parameter. 2325 // Check for that before calling the internal API nextUniqueId(). 2326 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2327 ALOGE("newAudioUniqueId invalid use %d", use); 2328 return AUDIO_UNIQUE_ID_ALLOCATE; 2329 } 2330 return nextUniqueId(use); 2331} 2332 2333void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2334{ 2335 Mutex::Autolock _l(mLock); 2336 pid_t caller = IPCThreadState::self()->getCallingPid(); 2337 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2338 if (pid != -1 && (caller == getpid_cached)) { 2339 caller = pid; 2340 } 2341 2342 { 2343 Mutex::Autolock _cl(mClientLock); 2344 // Ignore requests received from processes not known as notification client. The request 2345 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2346 // called from a different pid leaving a stale session reference. Also we don't know how 2347 // to clear this reference if the client process dies. 2348 if (mNotificationClients.indexOfKey(caller) < 0) { 2349 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2350 return; 2351 } 2352 } 2353 2354 size_t num = mAudioSessionRefs.size(); 2355 for (size_t i = 0; i< num; i++) { 2356 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2357 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2358 ref->mCnt++; 2359 ALOGV(" incremented refcount to %d", ref->mCnt); 2360 return; 2361 } 2362 } 2363 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2364 ALOGV(" added new entry for %d", audioSession); 2365} 2366 2367void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2368{ 2369 Mutex::Autolock _l(mLock); 2370 pid_t caller = IPCThreadState::self()->getCallingPid(); 2371 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2372 if (pid != -1 && (caller == getpid_cached)) { 2373 caller = pid; 2374 } 2375 size_t num = mAudioSessionRefs.size(); 2376 for (size_t i = 0; i< num; i++) { 2377 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2378 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2379 ref->mCnt--; 2380 ALOGV(" decremented refcount to %d", ref->mCnt); 2381 if (ref->mCnt == 0) { 2382 mAudioSessionRefs.removeAt(i); 2383 delete ref; 2384 purgeStaleEffects_l(); 2385 } 2386 return; 2387 } 2388 } 2389 // If the caller is mediaserver it is likely that the session being released was acquired 2390 // on behalf of a process not in notification clients and we ignore the warning. 2391 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2392} 2393 2394void AudioFlinger::purgeStaleEffects_l() { 2395 2396 ALOGV("purging stale effects"); 2397 2398 Vector< sp<EffectChain> > chains; 2399 2400 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2401 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2402 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2403 sp<EffectChain> ec = t->mEffectChains[j]; 2404 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2405 chains.push(ec); 2406 } 2407 } 2408 } 2409 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2410 sp<RecordThread> t = mRecordThreads.valueAt(i); 2411 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2412 sp<EffectChain> ec = t->mEffectChains[j]; 2413 chains.push(ec); 2414 } 2415 } 2416 2417 for (size_t i = 0; i < chains.size(); i++) { 2418 sp<EffectChain> ec = chains[i]; 2419 int sessionid = ec->sessionId(); 2420 sp<ThreadBase> t = ec->mThread.promote(); 2421 if (t == 0) { 2422 continue; 2423 } 2424 size_t numsessionrefs = mAudioSessionRefs.size(); 2425 bool found = false; 2426 for (size_t k = 0; k < numsessionrefs; k++) { 2427 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2428 if (ref->mSessionid == sessionid) { 2429 ALOGV(" session %d still exists for %d with %d refs", 2430 sessionid, ref->mPid, ref->mCnt); 2431 found = true; 2432 break; 2433 } 2434 } 2435 if (!found) { 2436 Mutex::Autolock _l(t->mLock); 2437 // remove all effects from the chain 2438 while (ec->mEffects.size()) { 2439 sp<EffectModule> effect = ec->mEffects[0]; 2440 effect->unPin(); 2441 t->removeEffect_l(effect); 2442 if (effect->purgeHandles()) { 2443 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2444 } 2445 AudioSystem::unregisterEffect(effect->id()); 2446 } 2447 } 2448 } 2449 return; 2450} 2451 2452// checkThread_l() must be called with AudioFlinger::mLock held 2453AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2454{ 2455 ThreadBase *thread = NULL; 2456 switch (audio_unique_id_get_use(ioHandle)) { 2457 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2458 thread = checkPlaybackThread_l(ioHandle); 2459 break; 2460 case AUDIO_UNIQUE_ID_USE_INPUT: 2461 thread = checkRecordThread_l(ioHandle); 2462 break; 2463 default: 2464 break; 2465 } 2466 return thread; 2467} 2468 2469// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2470AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2471{ 2472 return mPlaybackThreads.valueFor(output).get(); 2473} 2474 2475// checkMixerThread_l() must be called with AudioFlinger::mLock held 2476AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2477{ 2478 PlaybackThread *thread = checkPlaybackThread_l(output); 2479 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2480} 2481 2482// checkRecordThread_l() must be called with AudioFlinger::mLock held 2483AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2484{ 2485 return mRecordThreads.valueFor(input).get(); 2486} 2487 2488audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2489{ 2490 // This is the internal API, so it is OK to assert on bad parameter. 2491 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2492 const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1; 2493 for (int retry = 0; retry < maxRetries; retry++) { 2494 // The cast allows wraparound from max positive to min negative instead of abort 2495 uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use], 2496 (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel); 2497 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2498 // allow wrap by skipping 0 and -1 for session ids 2499 if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) { 2500 ALOGW_IF(retry != 0, "unique ID overflow for use %d", use); 2501 return (audio_unique_id_t) (base | use); 2502 } 2503 } 2504 // We have no way of recovering from wraparound 2505 LOG_ALWAYS_FATAL("unique ID overflow for use %d", use); 2506 // TODO Use a floor after wraparound. This may need a mutex. 2507} 2508 2509AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2510{ 2511 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2512 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2513 if(thread->isDuplicating()) { 2514 continue; 2515 } 2516 AudioStreamOut *output = thread->getOutput(); 2517 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2518 return thread; 2519 } 2520 } 2521 return NULL; 2522} 2523 2524audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2525{ 2526 PlaybackThread *thread = primaryPlaybackThread_l(); 2527 2528 if (thread == NULL) { 2529 return 0; 2530 } 2531 2532 return thread->outDevice(); 2533} 2534 2535sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2536 audio_session_t triggerSession, 2537 audio_session_t listenerSession, 2538 sync_event_callback_t callBack, 2539 wp<RefBase> cookie) 2540{ 2541 Mutex::Autolock _l(mLock); 2542 2543 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2544 status_t playStatus = NAME_NOT_FOUND; 2545 status_t recStatus = NAME_NOT_FOUND; 2546 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2547 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2548 if (playStatus == NO_ERROR) { 2549 return event; 2550 } 2551 } 2552 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2553 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2554 if (recStatus == NO_ERROR) { 2555 return event; 2556 } 2557 } 2558 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2559 mPendingSyncEvents.add(event); 2560 } else { 2561 ALOGV("createSyncEvent() invalid event %d", event->type()); 2562 event.clear(); 2563 } 2564 return event; 2565} 2566 2567// ---------------------------------------------------------------------------- 2568// Effect management 2569// ---------------------------------------------------------------------------- 2570 2571 2572status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2573{ 2574 Mutex::Autolock _l(mLock); 2575 return EffectQueryNumberEffects(numEffects); 2576} 2577 2578status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2579{ 2580 Mutex::Autolock _l(mLock); 2581 return EffectQueryEffect(index, descriptor); 2582} 2583 2584status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2585 effect_descriptor_t *descriptor) const 2586{ 2587 Mutex::Autolock _l(mLock); 2588 return EffectGetDescriptor(pUuid, descriptor); 2589} 2590 2591 2592sp<IEffect> AudioFlinger::createEffect( 2593 effect_descriptor_t *pDesc, 2594 const sp<IEffectClient>& effectClient, 2595 int32_t priority, 2596 audio_io_handle_t io, 2597 audio_session_t sessionId, 2598 const String16& opPackageName, 2599 status_t *status, 2600 int *id, 2601 int *enabled) 2602{ 2603 status_t lStatus = NO_ERROR; 2604 sp<EffectHandle> handle; 2605 effect_descriptor_t desc; 2606 2607 pid_t pid = IPCThreadState::self()->getCallingPid(); 2608 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2609 pid, effectClient.get(), priority, sessionId, io); 2610 2611 if (pDesc == NULL) { 2612 lStatus = BAD_VALUE; 2613 goto Exit; 2614 } 2615 2616 // check audio settings permission for global effects 2617 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2618 lStatus = PERMISSION_DENIED; 2619 goto Exit; 2620 } 2621 2622 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2623 // that can only be created by audio policy manager (running in same process) 2624 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2625 lStatus = PERMISSION_DENIED; 2626 goto Exit; 2627 } 2628 2629 { 2630 if (!EffectIsNullUuid(&pDesc->uuid)) { 2631 // if uuid is specified, request effect descriptor 2632 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2633 if (lStatus < 0) { 2634 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2635 goto Exit; 2636 } 2637 } else { 2638 // if uuid is not specified, look for an available implementation 2639 // of the required type in effect factory 2640 if (EffectIsNullUuid(&pDesc->type)) { 2641 ALOGW("createEffect() no effect type"); 2642 lStatus = BAD_VALUE; 2643 goto Exit; 2644 } 2645 uint32_t numEffects = 0; 2646 effect_descriptor_t d; 2647 d.flags = 0; // prevent compiler warning 2648 bool found = false; 2649 2650 lStatus = EffectQueryNumberEffects(&numEffects); 2651 if (lStatus < 0) { 2652 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2653 goto Exit; 2654 } 2655 for (uint32_t i = 0; i < numEffects; i++) { 2656 lStatus = EffectQueryEffect(i, &desc); 2657 if (lStatus < 0) { 2658 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2659 continue; 2660 } 2661 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2662 // If matching type found save effect descriptor. If the session is 2663 // 0 and the effect is not auxiliary, continue enumeration in case 2664 // an auxiliary version of this effect type is available 2665 found = true; 2666 d = desc; 2667 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2668 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2669 break; 2670 } 2671 } 2672 } 2673 if (!found) { 2674 lStatus = BAD_VALUE; 2675 ALOGW("createEffect() effect not found"); 2676 goto Exit; 2677 } 2678 // For same effect type, chose auxiliary version over insert version if 2679 // connect to output mix (Compliance to OpenSL ES) 2680 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2681 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2682 desc = d; 2683 } 2684 } 2685 2686 // Do not allow auxiliary effects on a session different from 0 (output mix) 2687 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2688 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2689 lStatus = INVALID_OPERATION; 2690 goto Exit; 2691 } 2692 2693 // check recording permission for visualizer 2694 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2695 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2696 lStatus = PERMISSION_DENIED; 2697 goto Exit; 2698 } 2699 2700 // return effect descriptor 2701 *pDesc = desc; 2702 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2703 // if the output returned by getOutputForEffect() is removed before we lock the 2704 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2705 // and we will exit safely 2706 io = AudioSystem::getOutputForEffect(&desc); 2707 ALOGV("createEffect got output %d", io); 2708 } 2709 2710 Mutex::Autolock _l(mLock); 2711 2712 // If output is not specified try to find a matching audio session ID in one of the 2713 // output threads. 2714 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2715 // because of code checking output when entering the function. 2716 // Note: io is never 0 when creating an effect on an input 2717 if (io == AUDIO_IO_HANDLE_NONE) { 2718 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2719 // output must be specified by AudioPolicyManager when using session 2720 // AUDIO_SESSION_OUTPUT_STAGE 2721 lStatus = BAD_VALUE; 2722 goto Exit; 2723 } 2724 // look for the thread where the specified audio session is present 2725 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2726 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2727 io = mPlaybackThreads.keyAt(i); 2728 break; 2729 } 2730 } 2731 if (io == 0) { 2732 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2733 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2734 io = mRecordThreads.keyAt(i); 2735 break; 2736 } 2737 } 2738 } 2739 // If no output thread contains the requested session ID, default to 2740 // first output. The effect chain will be moved to the correct output 2741 // thread when a track with the same session ID is created 2742 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2743 io = mPlaybackThreads.keyAt(0); 2744 } 2745 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2746 } 2747 ThreadBase *thread = checkRecordThread_l(io); 2748 if (thread == NULL) { 2749 thread = checkPlaybackThread_l(io); 2750 if (thread == NULL) { 2751 ALOGE("createEffect() unknown output thread"); 2752 lStatus = BAD_VALUE; 2753 goto Exit; 2754 } 2755 } else { 2756 // Check if one effect chain was awaiting for an effect to be created on this 2757 // session and used it instead of creating a new one. 2758 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2759 if (chain != 0) { 2760 Mutex::Autolock _l(thread->mLock); 2761 thread->addEffectChain_l(chain); 2762 } 2763 } 2764 2765 sp<Client> client = registerPid(pid); 2766 2767 // create effect on selected output thread 2768 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2769 &desc, enabled, &lStatus); 2770 if (handle != 0 && id != NULL) { 2771 *id = handle->id(); 2772 } 2773 if (handle == 0) { 2774 // remove local strong reference to Client with mClientLock held 2775 Mutex::Autolock _cl(mClientLock); 2776 client.clear(); 2777 } 2778 } 2779 2780Exit: 2781 *status = lStatus; 2782 return handle; 2783} 2784 2785status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2786 audio_io_handle_t dstOutput) 2787{ 2788 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2789 sessionId, srcOutput, dstOutput); 2790 Mutex::Autolock _l(mLock); 2791 if (srcOutput == dstOutput) { 2792 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2793 return NO_ERROR; 2794 } 2795 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2796 if (srcThread == NULL) { 2797 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2798 return BAD_VALUE; 2799 } 2800 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2801 if (dstThread == NULL) { 2802 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2803 return BAD_VALUE; 2804 } 2805 2806 Mutex::Autolock _dl(dstThread->mLock); 2807 Mutex::Autolock _sl(srcThread->mLock); 2808 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2809} 2810 2811// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2812status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2813 AudioFlinger::PlaybackThread *srcThread, 2814 AudioFlinger::PlaybackThread *dstThread, 2815 bool reRegister) 2816{ 2817 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2818 sessionId, srcThread, dstThread); 2819 2820 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2821 if (chain == 0) { 2822 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2823 sessionId, srcThread); 2824 return INVALID_OPERATION; 2825 } 2826 2827 // Check whether the destination thread has a channel count of FCC_2, which is 2828 // currently required for (most) effects. Prevent moving the effect chain here rather 2829 // than disabling the addEffect_l() call in dstThread below. 2830 if ((dstThread->type() == ThreadBase::MIXER || dstThread->isDuplicating()) && 2831 dstThread->mChannelCount != FCC_2) { 2832 ALOGW("moveEffectChain_l() effect chain failed because" 2833 " destination thread %p channel count(%u) != %u", 2834 dstThread, dstThread->mChannelCount, FCC_2); 2835 return INVALID_OPERATION; 2836 } 2837 2838 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2839 // so that a new chain is created with correct parameters when first effect is added. This is 2840 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2841 // removed. 2842 srcThread->removeEffectChain_l(chain); 2843 2844 // transfer all effects one by one so that new effect chain is created on new thread with 2845 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2846 sp<EffectChain> dstChain; 2847 uint32_t strategy = 0; // prevent compiler warning 2848 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2849 Vector< sp<EffectModule> > removed; 2850 status_t status = NO_ERROR; 2851 while (effect != 0) { 2852 srcThread->removeEffect_l(effect); 2853 removed.add(effect); 2854 status = dstThread->addEffect_l(effect); 2855 if (status != NO_ERROR) { 2856 break; 2857 } 2858 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2859 if (effect->state() == EffectModule::ACTIVE || 2860 effect->state() == EffectModule::STOPPING) { 2861 effect->start(); 2862 } 2863 // if the move request is not received from audio policy manager, the effect must be 2864 // re-registered with the new strategy and output 2865 if (dstChain == 0) { 2866 dstChain = effect->chain().promote(); 2867 if (dstChain == 0) { 2868 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2869 status = NO_INIT; 2870 break; 2871 } 2872 strategy = dstChain->strategy(); 2873 } 2874 if (reRegister) { 2875 AudioSystem::unregisterEffect(effect->id()); 2876 AudioSystem::registerEffect(&effect->desc(), 2877 dstThread->id(), 2878 strategy, 2879 sessionId, 2880 effect->id()); 2881 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2882 } 2883 effect = chain->getEffectFromId_l(0); 2884 } 2885 2886 if (status != NO_ERROR) { 2887 for (size_t i = 0; i < removed.size(); i++) { 2888 srcThread->addEffect_l(removed[i]); 2889 if (dstChain != 0 && reRegister) { 2890 AudioSystem::unregisterEffect(removed[i]->id()); 2891 AudioSystem::registerEffect(&removed[i]->desc(), 2892 srcThread->id(), 2893 strategy, 2894 sessionId, 2895 removed[i]->id()); 2896 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2897 } 2898 } 2899 } 2900 2901 return status; 2902} 2903 2904bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2905{ 2906 if (mGlobalEffectEnableTime != 0 && 2907 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2908 return true; 2909 } 2910 2911 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2912 sp<EffectChain> ec = 2913 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2914 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2915 return true; 2916 } 2917 } 2918 return false; 2919} 2920 2921void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2922{ 2923 Mutex::Autolock _l(mLock); 2924 2925 mGlobalEffectEnableTime = systemTime(); 2926 2927 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2928 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2929 if (t->mType == ThreadBase::OFFLOAD) { 2930 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2931 } 2932 } 2933 2934} 2935 2936status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2937{ 2938 audio_session_t session = chain->sessionId(); 2939 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2940 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2941 if (index >= 0) { 2942 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2943 return ALREADY_EXISTS; 2944 } 2945 mOrphanEffectChains.add(session, chain); 2946 return NO_ERROR; 2947} 2948 2949sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2950{ 2951 sp<EffectChain> chain; 2952 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2953 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2954 if (index >= 0) { 2955 chain = mOrphanEffectChains.valueAt(index); 2956 mOrphanEffectChains.removeItemsAt(index); 2957 } 2958 return chain; 2959} 2960 2961bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2962{ 2963 Mutex::Autolock _l(mLock); 2964 audio_session_t session = effect->sessionId(); 2965 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2966 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2967 if (index >= 0) { 2968 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2969 if (chain->removeEffect_l(effect) == 0) { 2970 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 2971 mOrphanEffectChains.removeItemsAt(index); 2972 } 2973 return true; 2974 } 2975 return false; 2976} 2977 2978 2979struct Entry { 2980#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2981 char mFileName[TEE_MAX_FILENAME]; 2982}; 2983 2984int comparEntry(const void *p1, const void *p2) 2985{ 2986 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2987} 2988 2989#ifdef TEE_SINK 2990void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2991{ 2992 NBAIO_Source *teeSource = source.get(); 2993 if (teeSource != NULL) { 2994 // .wav rotation 2995 // There is a benign race condition if 2 threads call this simultaneously. 2996 // They would both traverse the directory, but the result would simply be 2997 // failures at unlink() which are ignored. It's also unlikely since 2998 // normally dumpsys is only done by bugreport or from the command line. 2999 char teePath[32+256]; 3000 strcpy(teePath, "/data/misc/audioserver"); 3001 size_t teePathLen = strlen(teePath); 3002 DIR *dir = opendir(teePath); 3003 teePath[teePathLen++] = '/'; 3004 if (dir != NULL) { 3005#define TEE_MAX_SORT 20 // number of entries to sort 3006#define TEE_MAX_KEEP 10 // number of entries to keep 3007 struct Entry entries[TEE_MAX_SORT]; 3008 size_t entryCount = 0; 3009 while (entryCount < TEE_MAX_SORT) { 3010 struct dirent de; 3011 struct dirent *result = NULL; 3012 int rc = readdir_r(dir, &de, &result); 3013 if (rc != 0) { 3014 ALOGW("readdir_r failed %d", rc); 3015 break; 3016 } 3017 if (result == NULL) { 3018 break; 3019 } 3020 if (result != &de) { 3021 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 3022 break; 3023 } 3024 // ignore non .wav file entries 3025 size_t nameLen = strlen(de.d_name); 3026 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 3027 strcmp(&de.d_name[nameLen - 4], ".wav")) { 3028 continue; 3029 } 3030 strcpy(entries[entryCount++].mFileName, de.d_name); 3031 } 3032 (void) closedir(dir); 3033 if (entryCount > TEE_MAX_KEEP) { 3034 qsort(entries, entryCount, sizeof(Entry), comparEntry); 3035 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 3036 strcpy(&teePath[teePathLen], entries[i].mFileName); 3037 (void) unlink(teePath); 3038 } 3039 } 3040 } else { 3041 if (fd >= 0) { 3042 dprintf(fd, "unable to rotate tees in %.*s: %s\n", teePathLen, teePath, 3043 strerror(errno)); 3044 } 3045 } 3046 char teeTime[16]; 3047 struct timeval tv; 3048 gettimeofday(&tv, NULL); 3049 struct tm tm; 3050 localtime_r(&tv.tv_sec, &tm); 3051 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 3052 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 3053 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 3054 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 3055 if (teeFd >= 0) { 3056 // FIXME use libsndfile 3057 char wavHeader[44]; 3058 memcpy(wavHeader, 3059 "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", 3060 sizeof(wavHeader)); 3061 NBAIO_Format format = teeSource->format(); 3062 unsigned channelCount = Format_channelCount(format); 3063 uint32_t sampleRate = Format_sampleRate(format); 3064 size_t frameSize = Format_frameSize(format); 3065 wavHeader[22] = channelCount; // number of channels 3066 wavHeader[24] = sampleRate; // sample rate 3067 wavHeader[25] = sampleRate >> 8; 3068 wavHeader[32] = frameSize; // block alignment 3069 wavHeader[33] = frameSize >> 8; 3070 write(teeFd, wavHeader, sizeof(wavHeader)); 3071 size_t total = 0; 3072 bool firstRead = true; 3073#define TEE_SINK_READ 1024 // frames per I/O operation 3074 void *buffer = malloc(TEE_SINK_READ * frameSize); 3075 for (;;) { 3076 size_t count = TEE_SINK_READ; 3077 ssize_t actual = teeSource->read(buffer, count); 3078 bool wasFirstRead = firstRead; 3079 firstRead = false; 3080 if (actual <= 0) { 3081 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3082 continue; 3083 } 3084 break; 3085 } 3086 ALOG_ASSERT(actual <= (ssize_t)count); 3087 write(teeFd, buffer, actual * frameSize); 3088 total += actual; 3089 } 3090 free(buffer); 3091 lseek(teeFd, (off_t) 4, SEEK_SET); 3092 uint32_t temp = 44 + total * frameSize - 8; 3093 // FIXME not big-endian safe 3094 write(teeFd, &temp, sizeof(temp)); 3095 lseek(teeFd, (off_t) 40, SEEK_SET); 3096 temp = total * frameSize; 3097 // FIXME not big-endian safe 3098 write(teeFd, &temp, sizeof(temp)); 3099 close(teeFd); 3100 if (fd >= 0) { 3101 dprintf(fd, "tee copied to %s\n", teePath); 3102 } 3103 } else { 3104 if (fd >= 0) { 3105 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3106 } 3107 } 3108 } 3109} 3110#endif 3111 3112// ---------------------------------------------------------------------------- 3113 3114status_t AudioFlinger::onTransact( 3115 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3116{ 3117 return BnAudioFlinger::onTransact(code, data, reply, flags); 3118} 3119 3120} // namespace android 3121