AudioFlinger.cpp revision 92cd1cd127082f85cefad6fffe6671a991a52fe9
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 case AUDIO_FORMAT_DTS: return "dts"; 136 case AUDIO_FORMAT_DTS_HD: return "dts-hd"; 137 case AUDIO_FORMAT_DOLBY_TRUEHD: return "dolby-truehd"; 138 default: 139 break; 140 } 141 return "unknown"; 142} 143 144static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 145{ 146 const hw_module_t *mod; 147 int rc; 148 149 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 150 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 151 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 152 if (rc) { 153 goto out; 154 } 155 rc = audio_hw_device_open(mod, dev); 156 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 157 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 158 if (rc) { 159 goto out; 160 } 161 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 162 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 163 rc = BAD_VALUE; 164 goto out; 165 } 166 return 0; 167 168out: 169 *dev = NULL; 170 return rc; 171} 172 173// ---------------------------------------------------------------------------- 174 175AudioFlinger::AudioFlinger() 176 : BnAudioFlinger(), 177 mPrimaryHardwareDev(NULL), 178 mAudioHwDevs(NULL), 179 mHardwareStatus(AUDIO_HW_IDLE), 180 mMasterVolume(1.0f), 181 mMasterMute(false), 182 // mNextUniqueId(AUDIO_UNIQUE_ID_USE_MAX), 183 mMode(AUDIO_MODE_INVALID), 184 mBtNrecIsOff(false), 185 mIsLowRamDevice(true), 186 mIsDeviceTypeKnown(false), 187 mGlobalEffectEnableTime(0), 188 mSystemReady(false) 189{ 190 // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum 191 for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) { 192 // zero ID has a special meaning, so unavailable 193 mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX; 194 } 195 196 getpid_cached = getpid(); 197 const bool doLog = property_get_bool("ro.test_harness", false); 198 if (doLog) { 199 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 200 MemoryHeapBase::READ_ONLY); 201 } 202 203 // reset battery stats. 204 // if the audio service has crashed, battery stats could be left 205 // in bad state, reset the state upon service start. 206 BatteryNotifier::getInstance().noteResetAudio(); 207 208#ifdef TEE_SINK 209 char value[PROPERTY_VALUE_MAX]; 210 (void) property_get("ro.debuggable", value, "0"); 211 int debuggable = atoi(value); 212 int teeEnabled = 0; 213 if (debuggable) { 214 (void) property_get("af.tee", value, "0"); 215 teeEnabled = atoi(value); 216 } 217 // FIXME symbolic constants here 218 if (teeEnabled & 1) { 219 mTeeSinkInputEnabled = true; 220 } 221 if (teeEnabled & 2) { 222 mTeeSinkOutputEnabled = true; 223 } 224 if (teeEnabled & 4) { 225 mTeeSinkTrackEnabled = true; 226 } 227#endif 228} 229 230void AudioFlinger::onFirstRef() 231{ 232 Mutex::Autolock _l(mLock); 233 234 /* TODO: move all this work into an Init() function */ 235 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 236 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 237 uint32_t int_val; 238 if (1 == sscanf(val_str, "%u", &int_val)) { 239 mStandbyTimeInNsecs = milliseconds(int_val); 240 ALOGI("Using %u mSec as standby time.", int_val); 241 } else { 242 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 243 ALOGI("Using default %u mSec as standby time.", 244 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 245 } 246 } 247 248 mPatchPanel = new PatchPanel(this); 249 250 mMode = AUDIO_MODE_NORMAL; 251} 252 253AudioFlinger::~AudioFlinger() 254{ 255 while (!mRecordThreads.isEmpty()) { 256 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 257 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 258 } 259 while (!mPlaybackThreads.isEmpty()) { 260 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 261 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 262 } 263 264 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 265 // no mHardwareLock needed, as there are no other references to this 266 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 267 delete mAudioHwDevs.valueAt(i); 268 } 269 270 // Tell media.log service about any old writers that still need to be unregistered 271 if (mLogMemoryDealer != 0) { 272 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 273 if (binder != 0) { 274 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 275 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 276 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 277 mUnregisteredWriters.pop(); 278 mediaLogService->unregisterWriter(iMemory); 279 } 280 } 281 } 282} 283 284static const char * const audio_interfaces[] = { 285 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 286 AUDIO_HARDWARE_MODULE_ID_A2DP, 287 AUDIO_HARDWARE_MODULE_ID_USB, 288}; 289#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 290 291AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 292 audio_module_handle_t module, 293 audio_devices_t devices) 294{ 295 // if module is 0, the request comes from an old policy manager and we should load 296 // well known modules 297 if (module == 0) { 298 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 299 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 300 loadHwModule_l(audio_interfaces[i]); 301 } 302 // then try to find a module supporting the requested device. 303 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 304 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 305 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 306 if ((dev->get_supported_devices != NULL) && 307 (dev->get_supported_devices(dev) & devices) == devices) 308 return audioHwDevice; 309 } 310 } else { 311 // check a match for the requested module handle 312 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 313 if (audioHwDevice != NULL) { 314 return audioHwDevice; 315 } 316 } 317 318 return NULL; 319} 320 321void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 322{ 323 const size_t SIZE = 256; 324 char buffer[SIZE]; 325 String8 result; 326 327 result.append("Clients:\n"); 328 for (size_t i = 0; i < mClients.size(); ++i) { 329 sp<Client> client = mClients.valueAt(i).promote(); 330 if (client != 0) { 331 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 332 result.append(buffer); 333 } 334 } 335 336 result.append("Notification Clients:\n"); 337 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 338 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 339 result.append(buffer); 340 } 341 342 result.append("Global session refs:\n"); 343 result.append(" session pid count\n"); 344 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 345 AudioSessionRef *r = mAudioSessionRefs[i]; 346 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 347 result.append(buffer); 348 } 349 write(fd, result.string(), result.size()); 350} 351 352 353void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 354{ 355 const size_t SIZE = 256; 356 char buffer[SIZE]; 357 String8 result; 358 hardware_call_state hardwareStatus = mHardwareStatus; 359 360 snprintf(buffer, SIZE, "Hardware status: %d\n" 361 "Standby Time mSec: %u\n", 362 hardwareStatus, 363 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 364 result.append(buffer); 365 write(fd, result.string(), result.size()); 366} 367 368void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 369{ 370 const size_t SIZE = 256; 371 char buffer[SIZE]; 372 String8 result; 373 snprintf(buffer, SIZE, "Permission Denial: " 374 "can't dump AudioFlinger from pid=%d, uid=%d\n", 375 IPCThreadState::self()->getCallingPid(), 376 IPCThreadState::self()->getCallingUid()); 377 result.append(buffer); 378 write(fd, result.string(), result.size()); 379} 380 381bool AudioFlinger::dumpTryLock(Mutex& mutex) 382{ 383 bool locked = false; 384 for (int i = 0; i < kDumpLockRetries; ++i) { 385 if (mutex.tryLock() == NO_ERROR) { 386 locked = true; 387 break; 388 } 389 usleep(kDumpLockSleepUs); 390 } 391 return locked; 392} 393 394status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 395{ 396 if (!dumpAllowed()) { 397 dumpPermissionDenial(fd, args); 398 } else { 399 // get state of hardware lock 400 bool hardwareLocked = dumpTryLock(mHardwareLock); 401 if (!hardwareLocked) { 402 String8 result(kHardwareLockedString); 403 write(fd, result.string(), result.size()); 404 } else { 405 mHardwareLock.unlock(); 406 } 407 408 bool locked = dumpTryLock(mLock); 409 410 // failed to lock - AudioFlinger is probably deadlocked 411 if (!locked) { 412 String8 result(kDeadlockedString); 413 write(fd, result.string(), result.size()); 414 } 415 416 bool clientLocked = dumpTryLock(mClientLock); 417 if (!clientLocked) { 418 String8 result(kClientLockedString); 419 write(fd, result.string(), result.size()); 420 } 421 422 EffectDumpEffects(fd); 423 424 dumpClients(fd, args); 425 if (clientLocked) { 426 mClientLock.unlock(); 427 } 428 429 dumpInternals(fd, args); 430 431 // dump playback threads 432 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 433 mPlaybackThreads.valueAt(i)->dump(fd, args); 434 } 435 436 // dump record threads 437 for (size_t i = 0; i < mRecordThreads.size(); i++) { 438 mRecordThreads.valueAt(i)->dump(fd, args); 439 } 440 441 // dump orphan effect chains 442 if (mOrphanEffectChains.size() != 0) { 443 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 444 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 445 mOrphanEffectChains.valueAt(i)->dump(fd, args); 446 } 447 } 448 // dump all hardware devs 449 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 450 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 451 dev->dump(dev, fd); 452 } 453 454#ifdef TEE_SINK 455 // dump the serially shared record tee sink 456 if (mRecordTeeSource != 0) { 457 dumpTee(fd, mRecordTeeSource); 458 } 459#endif 460 461 if (locked) { 462 mLock.unlock(); 463 } 464 465 // append a copy of media.log here by forwarding fd to it, but don't attempt 466 // to lookup the service if it's not running, as it will block for a second 467 if (mLogMemoryDealer != 0) { 468 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 469 if (binder != 0) { 470 dprintf(fd, "\nmedia.log:\n"); 471 Vector<String16> args; 472 binder->dump(fd, args); 473 } 474 } 475 476 // check for optional arguments 477 bool dumpMem = false; 478 bool unreachableMemory = false; 479 for (const auto &arg : args) { 480 if (arg == String16("-m")) { 481 dumpMem = true; 482 } else if (arg == String16("--unreachable")) { 483 unreachableMemory = true; 484 } 485 } 486 487 if (dumpMem) { 488 dprintf(fd, "\nDumping memory:\n"); 489 std::string s = dumpMemoryAddresses(100 /* limit */); 490 write(fd, s.c_str(), s.size()); 491 } 492 if (unreachableMemory) { 493 dprintf(fd, "\nDumping unreachable memory:\n"); 494 // TODO - should limit be an argument parameter? 495 std::string s = GetUnreachableMemoryString(true /* contents */, 100 /* limit */); 496 write(fd, s.c_str(), s.size()); 497 } 498 } 499 return NO_ERROR; 500} 501 502sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 503{ 504 Mutex::Autolock _cl(mClientLock); 505 // If pid is already in the mClients wp<> map, then use that entry 506 // (for which promote() is always != 0), otherwise create a new entry and Client. 507 sp<Client> client = mClients.valueFor(pid).promote(); 508 if (client == 0) { 509 client = new Client(this, pid); 510 mClients.add(pid, client); 511 } 512 513 return client; 514} 515 516sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 517{ 518 // If there is no memory allocated for logs, return a dummy writer that does nothing 519 if (mLogMemoryDealer == 0) { 520 return new NBLog::Writer(); 521 } 522 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 523 // Similarly if we can't contact the media.log service, also return a dummy writer 524 if (binder == 0) { 525 return new NBLog::Writer(); 526 } 527 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 528 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 529 // If allocation fails, consult the vector of previously unregistered writers 530 // and garbage-collect one or more them until an allocation succeeds 531 if (shared == 0) { 532 Mutex::Autolock _l(mUnregisteredWritersLock); 533 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 534 { 535 // Pick the oldest stale writer to garbage-collect 536 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 537 mUnregisteredWriters.removeAt(0); 538 mediaLogService->unregisterWriter(iMemory); 539 // Now the media.log remote reference to IMemory is gone. When our last local 540 // reference to IMemory also drops to zero at end of this block, 541 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 542 } 543 // Re-attempt the allocation 544 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 545 if (shared != 0) { 546 goto success; 547 } 548 } 549 // Even after garbage-collecting all old writers, there is still not enough memory, 550 // so return a dummy writer 551 return new NBLog::Writer(); 552 } 553success: 554 mediaLogService->registerWriter(shared, size, name); 555 return new NBLog::Writer(size, shared); 556} 557 558void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 559{ 560 if (writer == 0) { 561 return; 562 } 563 sp<IMemory> iMemory(writer->getIMemory()); 564 if (iMemory == 0) { 565 return; 566 } 567 // Rather than removing the writer immediately, append it to a queue of old writers to 568 // be garbage-collected later. This allows us to continue to view old logs for a while. 569 Mutex::Autolock _l(mUnregisteredWritersLock); 570 mUnregisteredWriters.push(writer); 571} 572 573// IAudioFlinger interface 574 575 576sp<IAudioTrack> AudioFlinger::createTrack( 577 audio_stream_type_t streamType, 578 uint32_t sampleRate, 579 audio_format_t format, 580 audio_channel_mask_t channelMask, 581 size_t *frameCount, 582 audio_output_flags_t *flags, 583 const sp<IMemory>& sharedBuffer, 584 audio_io_handle_t output, 585 pid_t pid, 586 pid_t tid, 587 audio_session_t *sessionId, 588 int clientUid, 589 status_t *status) 590{ 591 sp<PlaybackThread::Track> track; 592 sp<TrackHandle> trackHandle; 593 sp<Client> client; 594 status_t lStatus; 595 audio_session_t lSessionId; 596 597 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 598 if (pid == -1 || !isTrustedCallingUid(callingUid)) { 599 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 600 ALOGW_IF(pid != -1 && pid != callingPid, 601 "%s uid %d pid %d tried to pass itself off as pid %d", 602 __func__, callingUid, callingPid, pid); 603 pid = callingPid; 604 } 605 606 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 607 // but if someone uses binder directly they could bypass that and cause us to crash 608 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 609 ALOGE("createTrack() invalid stream type %d", streamType); 610 lStatus = BAD_VALUE; 611 goto Exit; 612 } 613 614 // further sample rate checks are performed by createTrack_l() depending on the thread type 615 if (sampleRate == 0) { 616 ALOGE("createTrack() invalid sample rate %u", sampleRate); 617 lStatus = BAD_VALUE; 618 goto Exit; 619 } 620 621 // further channel mask checks are performed by createTrack_l() depending on the thread type 622 if (!audio_is_output_channel(channelMask)) { 623 ALOGE("createTrack() invalid channel mask %#x", channelMask); 624 lStatus = BAD_VALUE; 625 goto Exit; 626 } 627 628 // further format checks are performed by createTrack_l() depending on the thread type 629 if (!audio_is_valid_format(format)) { 630 ALOGE("createTrack() invalid format %#x", format); 631 lStatus = BAD_VALUE; 632 goto Exit; 633 } 634 635 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 636 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 637 lStatus = BAD_VALUE; 638 goto Exit; 639 } 640 641 { 642 Mutex::Autolock _l(mLock); 643 PlaybackThread *thread = checkPlaybackThread_l(output); 644 if (thread == NULL) { 645 ALOGE("no playback thread found for output handle %d", output); 646 lStatus = BAD_VALUE; 647 goto Exit; 648 } 649 650 client = registerPid(pid); 651 652 PlaybackThread *effectThread = NULL; 653 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 654 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 655 ALOGE("createTrack() invalid session ID %d", *sessionId); 656 lStatus = BAD_VALUE; 657 goto Exit; 658 } 659 lSessionId = *sessionId; 660 // check if an effect chain with the same session ID is present on another 661 // output thread and move it here. 662 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 663 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 664 if (mPlaybackThreads.keyAt(i) != output) { 665 uint32_t sessions = t->hasAudioSession(lSessionId); 666 if (sessions & ThreadBase::EFFECT_SESSION) { 667 effectThread = t.get(); 668 break; 669 } 670 } 671 } 672 } else { 673 // if no audio session id is provided, create one here 674 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 675 if (sessionId != NULL) { 676 *sessionId = lSessionId; 677 } 678 } 679 ALOGV("createTrack() lSessionId: %d", lSessionId); 680 681 track = thread->createTrack_l(client, streamType, sampleRate, format, 682 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 683 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 684 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 685 686 // move effect chain to this output thread if an effect on same session was waiting 687 // for a track to be created 688 if (lStatus == NO_ERROR && effectThread != NULL) { 689 // no risk of deadlock because AudioFlinger::mLock is held 690 Mutex::Autolock _dl(thread->mLock); 691 Mutex::Autolock _sl(effectThread->mLock); 692 moveEffectChain_l(lSessionId, effectThread, thread, true); 693 } 694 695 // Look for sync events awaiting for a session to be used. 696 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 697 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 698 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 699 if (lStatus == NO_ERROR) { 700 (void) track->setSyncEvent(mPendingSyncEvents[i]); 701 } else { 702 mPendingSyncEvents[i]->cancel(); 703 } 704 mPendingSyncEvents.removeAt(i); 705 i--; 706 } 707 } 708 } 709 710 setAudioHwSyncForSession_l(thread, lSessionId); 711 } 712 713 if (lStatus != NO_ERROR) { 714 // remove local strong reference to Client before deleting the Track so that the 715 // Client destructor is called by the TrackBase destructor with mClientLock held 716 // Don't hold mClientLock when releasing the reference on the track as the 717 // destructor will acquire it. 718 { 719 Mutex::Autolock _cl(mClientLock); 720 client.clear(); 721 } 722 track.clear(); 723 goto Exit; 724 } 725 726 // return handle to client 727 trackHandle = new TrackHandle(track); 728 729Exit: 730 *status = lStatus; 731 return trackHandle; 732} 733 734uint32_t AudioFlinger::sampleRate(audio_io_handle_t ioHandle) const 735{ 736 Mutex::Autolock _l(mLock); 737 ThreadBase *thread = checkThread_l(ioHandle); 738 if (thread == NULL) { 739 ALOGW("sampleRate() unknown thread %d", ioHandle); 740 return 0; 741 } 742 return thread->sampleRate(); 743} 744 745audio_format_t AudioFlinger::format(audio_io_handle_t output) const 746{ 747 Mutex::Autolock _l(mLock); 748 PlaybackThread *thread = checkPlaybackThread_l(output); 749 if (thread == NULL) { 750 ALOGW("format() unknown thread %d", output); 751 return AUDIO_FORMAT_INVALID; 752 } 753 return thread->format(); 754} 755 756size_t AudioFlinger::frameCount(audio_io_handle_t ioHandle) const 757{ 758 Mutex::Autolock _l(mLock); 759 ThreadBase *thread = checkThread_l(ioHandle); 760 if (thread == NULL) { 761 ALOGW("frameCount() unknown thread %d", ioHandle); 762 return 0; 763 } 764 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 765 // should examine all callers and fix them to handle smaller counts 766 return thread->frameCount(); 767} 768 769size_t AudioFlinger::frameCountHAL(audio_io_handle_t ioHandle) const 770{ 771 Mutex::Autolock _l(mLock); 772 ThreadBase *thread = checkThread_l(ioHandle); 773 if (thread == NULL) { 774 ALOGW("frameCountHAL() unknown thread %d", ioHandle); 775 return 0; 776 } 777 return thread->frameCountHAL(); 778} 779 780uint32_t AudioFlinger::latency(audio_io_handle_t output) const 781{ 782 Mutex::Autolock _l(mLock); 783 PlaybackThread *thread = checkPlaybackThread_l(output); 784 if (thread == NULL) { 785 ALOGW("latency(): no playback thread found for output handle %d", output); 786 return 0; 787 } 788 return thread->latency(); 789} 790 791status_t AudioFlinger::setMasterVolume(float value) 792{ 793 status_t ret = initCheck(); 794 if (ret != NO_ERROR) { 795 return ret; 796 } 797 798 // check calling permissions 799 if (!settingsAllowed()) { 800 return PERMISSION_DENIED; 801 } 802 803 Mutex::Autolock _l(mLock); 804 mMasterVolume = value; 805 806 // Set master volume in the HALs which support it. 807 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 808 AutoMutex lock(mHardwareLock); 809 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 810 811 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 812 if (dev->canSetMasterVolume()) { 813 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 814 } 815 mHardwareStatus = AUDIO_HW_IDLE; 816 } 817 818 // Now set the master volume in each playback thread. Playback threads 819 // assigned to HALs which do not have master volume support will apply 820 // master volume during the mix operation. Threads with HALs which do 821 // support master volume will simply ignore the setting. 822 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 823 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 824 continue; 825 } 826 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 827 } 828 829 return NO_ERROR; 830} 831 832status_t AudioFlinger::setMode(audio_mode_t mode) 833{ 834 status_t ret = initCheck(); 835 if (ret != NO_ERROR) { 836 return ret; 837 } 838 839 // check calling permissions 840 if (!settingsAllowed()) { 841 return PERMISSION_DENIED; 842 } 843 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 844 ALOGW("Illegal value: setMode(%d)", mode); 845 return BAD_VALUE; 846 } 847 848 { // scope for the lock 849 AutoMutex lock(mHardwareLock); 850 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 851 mHardwareStatus = AUDIO_HW_SET_MODE; 852 ret = dev->set_mode(dev, mode); 853 mHardwareStatus = AUDIO_HW_IDLE; 854 } 855 856 if (NO_ERROR == ret) { 857 Mutex::Autolock _l(mLock); 858 mMode = mode; 859 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 860 mPlaybackThreads.valueAt(i)->setMode(mode); 861 } 862 863 return ret; 864} 865 866status_t AudioFlinger::setMicMute(bool state) 867{ 868 status_t ret = initCheck(); 869 if (ret != NO_ERROR) { 870 return ret; 871 } 872 873 // check calling permissions 874 if (!settingsAllowed()) { 875 return PERMISSION_DENIED; 876 } 877 878 AutoMutex lock(mHardwareLock); 879 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 880 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 881 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 882 status_t result = dev->set_mic_mute(dev, state); 883 if (result != NO_ERROR) { 884 ret = result; 885 } 886 } 887 mHardwareStatus = AUDIO_HW_IDLE; 888 return ret; 889} 890 891bool AudioFlinger::getMicMute() const 892{ 893 status_t ret = initCheck(); 894 if (ret != NO_ERROR) { 895 return false; 896 } 897 bool mute = true; 898 bool state = AUDIO_MODE_INVALID; 899 AutoMutex lock(mHardwareLock); 900 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 901 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 902 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 903 status_t result = dev->get_mic_mute(dev, &state); 904 if (result == NO_ERROR) { 905 mute = mute && state; 906 } 907 } 908 mHardwareStatus = AUDIO_HW_IDLE; 909 910 return mute; 911} 912 913status_t AudioFlinger::setMasterMute(bool muted) 914{ 915 status_t ret = initCheck(); 916 if (ret != NO_ERROR) { 917 return ret; 918 } 919 920 // check calling permissions 921 if (!settingsAllowed()) { 922 return PERMISSION_DENIED; 923 } 924 925 Mutex::Autolock _l(mLock); 926 mMasterMute = muted; 927 928 // Set master mute in the HALs which support it. 929 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 930 AutoMutex lock(mHardwareLock); 931 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 932 933 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 934 if (dev->canSetMasterMute()) { 935 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 936 } 937 mHardwareStatus = AUDIO_HW_IDLE; 938 } 939 940 // Now set the master mute in each playback thread. Playback threads 941 // assigned to HALs which do not have master mute support will apply master 942 // mute during the mix operation. Threads with HALs which do support master 943 // mute will simply ignore the setting. 944 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 945 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 946 continue; 947 } 948 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 949 } 950 951 return NO_ERROR; 952} 953 954float AudioFlinger::masterVolume() const 955{ 956 Mutex::Autolock _l(mLock); 957 return masterVolume_l(); 958} 959 960bool AudioFlinger::masterMute() const 961{ 962 Mutex::Autolock _l(mLock); 963 return masterMute_l(); 964} 965 966float AudioFlinger::masterVolume_l() const 967{ 968 return mMasterVolume; 969} 970 971bool AudioFlinger::masterMute_l() const 972{ 973 return mMasterMute; 974} 975 976status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 977{ 978 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 979 ALOGW("setStreamVolume() invalid stream %d", stream); 980 return BAD_VALUE; 981 } 982 pid_t caller = IPCThreadState::self()->getCallingPid(); 983 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 984 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 985 return PERMISSION_DENIED; 986 } 987 988 return NO_ERROR; 989} 990 991status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 992 audio_io_handle_t output) 993{ 994 // check calling permissions 995 if (!settingsAllowed()) { 996 return PERMISSION_DENIED; 997 } 998 999 status_t status = checkStreamType(stream); 1000 if (status != NO_ERROR) { 1001 return status; 1002 } 1003 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 1004 1005 AutoMutex lock(mLock); 1006 PlaybackThread *thread = NULL; 1007 if (output != AUDIO_IO_HANDLE_NONE) { 1008 thread = checkPlaybackThread_l(output); 1009 if (thread == NULL) { 1010 return BAD_VALUE; 1011 } 1012 } 1013 1014 mStreamTypes[stream].volume = value; 1015 1016 if (thread == NULL) { 1017 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1018 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 1019 } 1020 } else { 1021 thread->setStreamVolume(stream, value); 1022 } 1023 1024 return NO_ERROR; 1025} 1026 1027status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 1028{ 1029 // check calling permissions 1030 if (!settingsAllowed()) { 1031 return PERMISSION_DENIED; 1032 } 1033 1034 status_t status = checkStreamType(stream); 1035 if (status != NO_ERROR) { 1036 return status; 1037 } 1038 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 1039 1040 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 1041 ALOGE("setStreamMute() invalid stream %d", stream); 1042 return BAD_VALUE; 1043 } 1044 1045 AutoMutex lock(mLock); 1046 mStreamTypes[stream].mute = muted; 1047 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 1048 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 1049 1050 return NO_ERROR; 1051} 1052 1053float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 1054{ 1055 status_t status = checkStreamType(stream); 1056 if (status != NO_ERROR) { 1057 return 0.0f; 1058 } 1059 1060 AutoMutex lock(mLock); 1061 float volume; 1062 if (output != AUDIO_IO_HANDLE_NONE) { 1063 PlaybackThread *thread = checkPlaybackThread_l(output); 1064 if (thread == NULL) { 1065 return 0.0f; 1066 } 1067 volume = thread->streamVolume(stream); 1068 } else { 1069 volume = streamVolume_l(stream); 1070 } 1071 1072 return volume; 1073} 1074 1075bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1076{ 1077 status_t status = checkStreamType(stream); 1078 if (status != NO_ERROR) { 1079 return true; 1080 } 1081 1082 AutoMutex lock(mLock); 1083 return streamMute_l(stream); 1084} 1085 1086 1087void AudioFlinger::broacastParametersToRecordThreads_l(const String8& keyValuePairs) 1088{ 1089 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1090 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1091 } 1092} 1093 1094status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1095{ 1096 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1097 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1098 1099 // check calling permissions 1100 if (!settingsAllowed()) { 1101 return PERMISSION_DENIED; 1102 } 1103 1104 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1105 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1106 Mutex::Autolock _l(mLock); 1107 // result will remain NO_INIT if no audio device is present 1108 status_t final_result = NO_INIT; 1109 { 1110 AutoMutex lock(mHardwareLock); 1111 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1112 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1113 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1114 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1115 // return success if at least one audio device accepts the parameters as not all 1116 // HALs are requested to support all parameters. If no audio device supports the 1117 // requested parameters, the last error is reported. 1118 if (final_result != NO_ERROR) { 1119 final_result = result; 1120 } 1121 } 1122 mHardwareStatus = AUDIO_HW_IDLE; 1123 } 1124 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1125 AudioParameter param = AudioParameter(keyValuePairs); 1126 String8 value; 1127 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1128 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1129 if (mBtNrecIsOff != btNrecIsOff) { 1130 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1131 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1132 audio_devices_t device = thread->inDevice(); 1133 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1134 // collect all of the thread's session IDs 1135 KeyedVector<audio_session_t, bool> ids = thread->sessionIds(); 1136 // suspend effects associated with those session IDs 1137 for (size_t j = 0; j < ids.size(); ++j) { 1138 audio_session_t sessionId = ids.keyAt(j); 1139 thread->setEffectSuspended(FX_IID_AEC, 1140 suspend, 1141 sessionId); 1142 thread->setEffectSuspended(FX_IID_NS, 1143 suspend, 1144 sessionId); 1145 } 1146 } 1147 mBtNrecIsOff = btNrecIsOff; 1148 } 1149 } 1150 String8 screenState; 1151 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1152 bool isOff = screenState == "off"; 1153 if (isOff != (AudioFlinger::mScreenState & 1)) { 1154 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1155 } 1156 } 1157 return final_result; 1158 } 1159 1160 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1161 // and the thread is exited once the lock is released 1162 sp<ThreadBase> thread; 1163 { 1164 Mutex::Autolock _l(mLock); 1165 thread = checkPlaybackThread_l(ioHandle); 1166 if (thread == 0) { 1167 thread = checkRecordThread_l(ioHandle); 1168 } else if (thread == primaryPlaybackThread_l()) { 1169 // indicate output device change to all input threads for pre processing 1170 AudioParameter param = AudioParameter(keyValuePairs); 1171 int value; 1172 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1173 (value != 0)) { 1174 broacastParametersToRecordThreads_l(keyValuePairs); 1175 } 1176 } 1177 } 1178 if (thread != 0) { 1179 return thread->setParameters(keyValuePairs); 1180 } 1181 return BAD_VALUE; 1182} 1183 1184String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1185{ 1186 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1187 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1188 1189 Mutex::Autolock _l(mLock); 1190 1191 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1192 String8 out_s8; 1193 1194 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1195 char *s; 1196 { 1197 AutoMutex lock(mHardwareLock); 1198 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1199 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1200 s = dev->get_parameters(dev, keys.string()); 1201 mHardwareStatus = AUDIO_HW_IDLE; 1202 } 1203 out_s8 += String8(s ? s : ""); 1204 free(s); 1205 } 1206 return out_s8; 1207 } 1208 1209 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1210 if (playbackThread != NULL) { 1211 return playbackThread->getParameters(keys); 1212 } 1213 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1214 if (recordThread != NULL) { 1215 return recordThread->getParameters(keys); 1216 } 1217 return String8(""); 1218} 1219 1220size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1221 audio_channel_mask_t channelMask) const 1222{ 1223 status_t ret = initCheck(); 1224 if (ret != NO_ERROR) { 1225 return 0; 1226 } 1227 if ((sampleRate == 0) || 1228 !audio_is_valid_format(format) || !audio_has_proportional_frames(format) || 1229 !audio_is_input_channel(channelMask)) { 1230 return 0; 1231 } 1232 1233 AutoMutex lock(mHardwareLock); 1234 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1235 audio_config_t config, proposed; 1236 memset(&proposed, 0, sizeof(proposed)); 1237 proposed.sample_rate = sampleRate; 1238 proposed.channel_mask = channelMask; 1239 proposed.format = format; 1240 1241 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1242 size_t frames; 1243 for (;;) { 1244 // Note: config is currently a const parameter for get_input_buffer_size() 1245 // but we use a copy from proposed in case config changes from the call. 1246 config = proposed; 1247 frames = dev->get_input_buffer_size(dev, &config); 1248 if (frames != 0) { 1249 break; // hal success, config is the result 1250 } 1251 // change one parameter of the configuration each iteration to a more "common" value 1252 // to see if the device will support it. 1253 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1254 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1255 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1256 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1257 } else { 1258 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1259 "format %#x, channelMask 0x%X", 1260 sampleRate, format, channelMask); 1261 break; // retries failed, break out of loop with frames == 0. 1262 } 1263 } 1264 mHardwareStatus = AUDIO_HW_IDLE; 1265 if (frames > 0 && config.sample_rate != sampleRate) { 1266 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1267 } 1268 return frames; // may be converted to bytes at the Java level. 1269} 1270 1271uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1272{ 1273 Mutex::Autolock _l(mLock); 1274 1275 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1276 if (recordThread != NULL) { 1277 return recordThread->getInputFramesLost(); 1278 } 1279 return 0; 1280} 1281 1282status_t AudioFlinger::setVoiceVolume(float value) 1283{ 1284 status_t ret = initCheck(); 1285 if (ret != NO_ERROR) { 1286 return ret; 1287 } 1288 1289 // check calling permissions 1290 if (!settingsAllowed()) { 1291 return PERMISSION_DENIED; 1292 } 1293 1294 AutoMutex lock(mHardwareLock); 1295 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1296 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1297 ret = dev->set_voice_volume(dev, value); 1298 mHardwareStatus = AUDIO_HW_IDLE; 1299 1300 return ret; 1301} 1302 1303status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1304 audio_io_handle_t output) const 1305{ 1306 Mutex::Autolock _l(mLock); 1307 1308 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1309 if (playbackThread != NULL) { 1310 return playbackThread->getRenderPosition(halFrames, dspFrames); 1311 } 1312 1313 return BAD_VALUE; 1314} 1315 1316void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1317{ 1318 Mutex::Autolock _l(mLock); 1319 if (client == 0) { 1320 return; 1321 } 1322 pid_t pid = IPCThreadState::self()->getCallingPid(); 1323 { 1324 Mutex::Autolock _cl(mClientLock); 1325 if (mNotificationClients.indexOfKey(pid) < 0) { 1326 sp<NotificationClient> notificationClient = new NotificationClient(this, 1327 client, 1328 pid); 1329 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1330 1331 mNotificationClients.add(pid, notificationClient); 1332 1333 sp<IBinder> binder = IInterface::asBinder(client); 1334 binder->linkToDeath(notificationClient); 1335 } 1336 } 1337 1338 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1339 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1340 // the config change is always sent from playback or record threads to avoid deadlock 1341 // with AudioSystem::gLock 1342 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1343 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AUDIO_OUTPUT_OPENED, pid); 1344 } 1345 1346 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1347 mRecordThreads.valueAt(i)->sendIoConfigEvent(AUDIO_INPUT_OPENED, pid); 1348 } 1349} 1350 1351void AudioFlinger::removeNotificationClient(pid_t pid) 1352{ 1353 Mutex::Autolock _l(mLock); 1354 { 1355 Mutex::Autolock _cl(mClientLock); 1356 mNotificationClients.removeItem(pid); 1357 } 1358 1359 ALOGV("%d died, releasing its sessions", pid); 1360 size_t num = mAudioSessionRefs.size(); 1361 bool removed = false; 1362 for (size_t i = 0; i< num; ) { 1363 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1364 ALOGV(" pid %d @ %zu", ref->mPid, i); 1365 if (ref->mPid == pid) { 1366 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1367 mAudioSessionRefs.removeAt(i); 1368 delete ref; 1369 removed = true; 1370 num--; 1371 } else { 1372 i++; 1373 } 1374 } 1375 if (removed) { 1376 purgeStaleEffects_l(); 1377 } 1378} 1379 1380void AudioFlinger::ioConfigChanged(audio_io_config_event event, 1381 const sp<AudioIoDescriptor>& ioDesc, 1382 pid_t pid) 1383{ 1384 Mutex::Autolock _l(mClientLock); 1385 size_t size = mNotificationClients.size(); 1386 for (size_t i = 0; i < size; i++) { 1387 if ((pid == 0) || (mNotificationClients.keyAt(i) == pid)) { 1388 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, ioDesc); 1389 } 1390 } 1391} 1392 1393// removeClient_l() must be called with AudioFlinger::mClientLock held 1394void AudioFlinger::removeClient_l(pid_t pid) 1395{ 1396 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1397 IPCThreadState::self()->getCallingPid()); 1398 mClients.removeItem(pid); 1399} 1400 1401// getEffectThread_l() must be called with AudioFlinger::mLock held 1402sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(audio_session_t sessionId, 1403 int EffectId) 1404{ 1405 sp<PlaybackThread> thread; 1406 1407 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1408 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1409 ALOG_ASSERT(thread == 0); 1410 thread = mPlaybackThreads.valueAt(i); 1411 } 1412 } 1413 1414 return thread; 1415} 1416 1417 1418 1419// ---------------------------------------------------------------------------- 1420 1421AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1422 : RefBase(), 1423 mAudioFlinger(audioFlinger), 1424 mPid(pid) 1425{ 1426 size_t heapSize = property_get_int32("ro.af.client_heap_size_kbyte", 0); 1427 heapSize *= 1024; 1428 if (!heapSize) { 1429 heapSize = kClientSharedHeapSizeBytes; 1430 // Increase heap size on non low ram devices to limit risk of reconnection failure for 1431 // invalidated tracks 1432 if (!audioFlinger->isLowRamDevice()) { 1433 heapSize *= kClientSharedHeapSizeMultiplier; 1434 } 1435 } 1436 mMemoryDealer = new MemoryDealer(heapSize, "AudioFlinger::Client"); 1437} 1438 1439// Client destructor must be called with AudioFlinger::mClientLock held 1440AudioFlinger::Client::~Client() 1441{ 1442 mAudioFlinger->removeClient_l(mPid); 1443} 1444 1445sp<MemoryDealer> AudioFlinger::Client::heap() const 1446{ 1447 return mMemoryDealer; 1448} 1449 1450// ---------------------------------------------------------------------------- 1451 1452AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1453 const sp<IAudioFlingerClient>& client, 1454 pid_t pid) 1455 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1456{ 1457} 1458 1459AudioFlinger::NotificationClient::~NotificationClient() 1460{ 1461} 1462 1463void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1464{ 1465 sp<NotificationClient> keep(this); 1466 mAudioFlinger->removeNotificationClient(mPid); 1467} 1468 1469 1470// ---------------------------------------------------------------------------- 1471 1472sp<IAudioRecord> AudioFlinger::openRecord( 1473 audio_io_handle_t input, 1474 uint32_t sampleRate, 1475 audio_format_t format, 1476 audio_channel_mask_t channelMask, 1477 const String16& opPackageName, 1478 size_t *frameCount, 1479 audio_input_flags_t *flags, 1480 pid_t pid, 1481 pid_t tid, 1482 int clientUid, 1483 audio_session_t *sessionId, 1484 size_t *notificationFrames, 1485 sp<IMemory>& cblk, 1486 sp<IMemory>& buffers, 1487 status_t *status) 1488{ 1489 sp<RecordThread::RecordTrack> recordTrack; 1490 sp<RecordHandle> recordHandle; 1491 sp<Client> client; 1492 status_t lStatus; 1493 audio_session_t lSessionId; 1494 1495 cblk.clear(); 1496 buffers.clear(); 1497 1498 bool updatePid = (pid == -1); 1499 const uid_t callingUid = IPCThreadState::self()->getCallingUid(); 1500 if (!isTrustedCallingUid(callingUid)) { 1501 ALOGW_IF((uid_t)clientUid != callingUid, 1502 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid); 1503 clientUid = callingUid; 1504 updatePid = true; 1505 } 1506 1507 if (updatePid) { 1508 const pid_t callingPid = IPCThreadState::self()->getCallingPid(); 1509 ALOGW_IF(pid != -1 && pid != callingPid, 1510 "%s uid %d pid %d tried to pass itself off as pid %d", 1511 __func__, callingUid, callingPid, pid); 1512 pid = callingPid; 1513 } 1514 1515 // check calling permissions 1516 if (!recordingAllowed(opPackageName, tid, clientUid)) { 1517 ALOGE("openRecord() permission denied: recording not allowed"); 1518 lStatus = PERMISSION_DENIED; 1519 goto Exit; 1520 } 1521 1522 // further sample rate checks are performed by createRecordTrack_l() 1523 if (sampleRate == 0) { 1524 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1525 lStatus = BAD_VALUE; 1526 goto Exit; 1527 } 1528 1529 // we don't yet support anything other than linear PCM 1530 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1531 ALOGE("openRecord() invalid format %#x", format); 1532 lStatus = BAD_VALUE; 1533 goto Exit; 1534 } 1535 1536 // further channel mask checks are performed by createRecordTrack_l() 1537 if (!audio_is_input_channel(channelMask)) { 1538 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1539 lStatus = BAD_VALUE; 1540 goto Exit; 1541 } 1542 1543 { 1544 Mutex::Autolock _l(mLock); 1545 RecordThread *thread = checkRecordThread_l(input); 1546 if (thread == NULL) { 1547 ALOGE("openRecord() checkRecordThread_l failed"); 1548 lStatus = BAD_VALUE; 1549 goto Exit; 1550 } 1551 1552 client = registerPid(pid); 1553 1554 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1555 if (audio_unique_id_get_use(*sessionId) != AUDIO_UNIQUE_ID_USE_SESSION) { 1556 lStatus = BAD_VALUE; 1557 goto Exit; 1558 } 1559 lSessionId = *sessionId; 1560 } else { 1561 // if no audio session id is provided, create one here 1562 lSessionId = (audio_session_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_SESSION); 1563 if (sessionId != NULL) { 1564 *sessionId = lSessionId; 1565 } 1566 } 1567 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1568 1569 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1570 frameCount, lSessionId, notificationFrames, 1571 clientUid, flags, tid, &lStatus); 1572 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1573 1574 if (lStatus == NO_ERROR) { 1575 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1576 // session and move it to this thread. 1577 sp<EffectChain> chain = getOrphanEffectChain_l(lSessionId); 1578 if (chain != 0) { 1579 Mutex::Autolock _l(thread->mLock); 1580 thread->addEffectChain_l(chain); 1581 } 1582 } 1583 } 1584 1585 if (lStatus != NO_ERROR) { 1586 // remove local strong reference to Client before deleting the RecordTrack so that the 1587 // Client destructor is called by the TrackBase destructor with mClientLock held 1588 // Don't hold mClientLock when releasing the reference on the track as the 1589 // destructor will acquire it. 1590 { 1591 Mutex::Autolock _cl(mClientLock); 1592 client.clear(); 1593 } 1594 recordTrack.clear(); 1595 goto Exit; 1596 } 1597 1598 cblk = recordTrack->getCblk(); 1599 buffers = recordTrack->getBuffers(); 1600 1601 // return handle to client 1602 recordHandle = new RecordHandle(recordTrack); 1603 1604Exit: 1605 *status = lStatus; 1606 return recordHandle; 1607} 1608 1609 1610 1611// ---------------------------------------------------------------------------- 1612 1613audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1614{ 1615 if (name == NULL) { 1616 return AUDIO_MODULE_HANDLE_NONE; 1617 } 1618 if (!settingsAllowed()) { 1619 return AUDIO_MODULE_HANDLE_NONE; 1620 } 1621 Mutex::Autolock _l(mLock); 1622 return loadHwModule_l(name); 1623} 1624 1625// loadHwModule_l() must be called with AudioFlinger::mLock held 1626audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1627{ 1628 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1629 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1630 ALOGW("loadHwModule() module %s already loaded", name); 1631 return mAudioHwDevs.keyAt(i); 1632 } 1633 } 1634 1635 audio_hw_device_t *dev; 1636 1637 int rc = load_audio_interface(name, &dev); 1638 if (rc) { 1639 ALOGE("loadHwModule() error %d loading module %s", rc, name); 1640 return AUDIO_MODULE_HANDLE_NONE; 1641 } 1642 1643 mHardwareStatus = AUDIO_HW_INIT; 1644 rc = dev->init_check(dev); 1645 mHardwareStatus = AUDIO_HW_IDLE; 1646 if (rc) { 1647 ALOGE("loadHwModule() init check error %d for module %s", rc, name); 1648 return AUDIO_MODULE_HANDLE_NONE; 1649 } 1650 1651 // Check and cache this HAL's level of support for master mute and master 1652 // volume. If this is the first HAL opened, and it supports the get 1653 // methods, use the initial values provided by the HAL as the current 1654 // master mute and volume settings. 1655 1656 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1657 { // scope for auto-lock pattern 1658 AutoMutex lock(mHardwareLock); 1659 1660 if (0 == mAudioHwDevs.size()) { 1661 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1662 if (NULL != dev->get_master_volume) { 1663 float mv; 1664 if (OK == dev->get_master_volume(dev, &mv)) { 1665 mMasterVolume = mv; 1666 } 1667 } 1668 1669 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1670 if (NULL != dev->get_master_mute) { 1671 bool mm; 1672 if (OK == dev->get_master_mute(dev, &mm)) { 1673 mMasterMute = mm; 1674 } 1675 } 1676 } 1677 1678 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1679 if ((NULL != dev->set_master_volume) && 1680 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1681 flags = static_cast<AudioHwDevice::Flags>(flags | 1682 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1683 } 1684 1685 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1686 if ((NULL != dev->set_master_mute) && 1687 (OK == dev->set_master_mute(dev, mMasterMute))) { 1688 flags = static_cast<AudioHwDevice::Flags>(flags | 1689 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1690 } 1691 1692 mHardwareStatus = AUDIO_HW_IDLE; 1693 } 1694 1695 audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE); 1696 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1697 1698 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1699 name, dev->common.module->name, dev->common.module->id, handle); 1700 1701 return handle; 1702 1703} 1704 1705// ---------------------------------------------------------------------------- 1706 1707uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1708{ 1709 Mutex::Autolock _l(mLock); 1710 PlaybackThread *thread = fastPlaybackThread_l(); 1711 return thread != NULL ? thread->sampleRate() : 0; 1712} 1713 1714size_t AudioFlinger::getPrimaryOutputFrameCount() 1715{ 1716 Mutex::Autolock _l(mLock); 1717 PlaybackThread *thread = fastPlaybackThread_l(); 1718 return thread != NULL ? thread->frameCountHAL() : 0; 1719} 1720 1721// ---------------------------------------------------------------------------- 1722 1723status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1724{ 1725 uid_t uid = IPCThreadState::self()->getCallingUid(); 1726 if (uid != AID_SYSTEM) { 1727 return PERMISSION_DENIED; 1728 } 1729 Mutex::Autolock _l(mLock); 1730 if (mIsDeviceTypeKnown) { 1731 return INVALID_OPERATION; 1732 } 1733 mIsLowRamDevice = isLowRamDevice; 1734 mIsDeviceTypeKnown = true; 1735 return NO_ERROR; 1736} 1737 1738audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1739{ 1740 Mutex::Autolock _l(mLock); 1741 1742 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1743 if (index >= 0) { 1744 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1745 mHwAvSyncIds.valueAt(index), sessionId); 1746 return mHwAvSyncIds.valueAt(index); 1747 } 1748 1749 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1750 if (dev == NULL) { 1751 return AUDIO_HW_SYNC_INVALID; 1752 } 1753 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1754 AudioParameter param = AudioParameter(String8(reply)); 1755 free(reply); 1756 1757 int value; 1758 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1759 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1760 return AUDIO_HW_SYNC_INVALID; 1761 } 1762 1763 // allow only one session for a given HW A/V sync ID. 1764 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1765 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1766 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1767 value, mHwAvSyncIds.keyAt(i)); 1768 mHwAvSyncIds.removeItemsAt(i); 1769 break; 1770 } 1771 } 1772 1773 mHwAvSyncIds.add(sessionId, value); 1774 1775 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1776 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1777 uint32_t sessions = thread->hasAudioSession(sessionId); 1778 if (sessions & ThreadBase::TRACK_SESSION) { 1779 AudioParameter param = AudioParameter(); 1780 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1781 thread->setParameters(param.toString()); 1782 break; 1783 } 1784 } 1785 1786 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1787 return (audio_hw_sync_t)value; 1788} 1789 1790status_t AudioFlinger::systemReady() 1791{ 1792 Mutex::Autolock _l(mLock); 1793 ALOGI("%s", __FUNCTION__); 1794 if (mSystemReady) { 1795 ALOGW("%s called twice", __FUNCTION__); 1796 return NO_ERROR; 1797 } 1798 mSystemReady = true; 1799 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1800 ThreadBase *thread = (ThreadBase *)mPlaybackThreads.valueAt(i).get(); 1801 thread->systemReady(); 1802 } 1803 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1804 ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get(); 1805 thread->systemReady(); 1806 } 1807 return NO_ERROR; 1808} 1809 1810// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1811void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1812{ 1813 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1814 if (index >= 0) { 1815 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1816 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1817 AudioParameter param = AudioParameter(); 1818 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1819 thread->setParameters(param.toString()); 1820 } 1821} 1822 1823 1824// ---------------------------------------------------------------------------- 1825 1826 1827sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1828 audio_io_handle_t *output, 1829 audio_config_t *config, 1830 audio_devices_t devices, 1831 const String8& address, 1832 audio_output_flags_t flags) 1833{ 1834 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1835 if (outHwDev == NULL) { 1836 return 0; 1837 } 1838 1839 if (*output == AUDIO_IO_HANDLE_NONE) { 1840 *output = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1841 } else { 1842 // Audio Policy does not currently request a specific output handle. 1843 // If this is ever needed, see openInput_l() for example code. 1844 ALOGE("openOutput_l requested output handle %d is not AUDIO_IO_HANDLE_NONE", *output); 1845 return 0; 1846 } 1847 1848 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1849 1850 // FOR TESTING ONLY: 1851 // This if statement allows overriding the audio policy settings 1852 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1853 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1854 // Check only for Normal Mixing mode 1855 if (kEnableExtendedPrecision) { 1856 // Specify format (uncomment one below to choose) 1857 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1858 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1859 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1860 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1861 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1862 } 1863 if (kEnableExtendedChannels) { 1864 // Specify channel mask (uncomment one below to choose) 1865 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1866 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1867 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1868 } 1869 } 1870 1871 AudioStreamOut *outputStream = NULL; 1872 status_t status = outHwDev->openOutputStream( 1873 &outputStream, 1874 *output, 1875 devices, 1876 flags, 1877 config, 1878 address.string()); 1879 1880 mHardwareStatus = AUDIO_HW_IDLE; 1881 1882 if (status == NO_ERROR) { 1883 1884 PlaybackThread *thread; 1885 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1886 thread = new OffloadThread(this, outputStream, *output, devices, mSystemReady); 1887 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1888 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1889 || !isValidPcmSinkFormat(config->format) 1890 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1891 thread = new DirectOutputThread(this, outputStream, *output, devices, mSystemReady); 1892 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1893 } else { 1894 thread = new MixerThread(this, outputStream, *output, devices, mSystemReady); 1895 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1896 } 1897 mPlaybackThreads.add(*output, thread); 1898 return thread; 1899 } 1900 1901 return 0; 1902} 1903 1904status_t AudioFlinger::openOutput(audio_module_handle_t module, 1905 audio_io_handle_t *output, 1906 audio_config_t *config, 1907 audio_devices_t *devices, 1908 const String8& address, 1909 uint32_t *latencyMs, 1910 audio_output_flags_t flags) 1911{ 1912 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1913 module, 1914 (devices != NULL) ? *devices : 0, 1915 config->sample_rate, 1916 config->format, 1917 config->channel_mask, 1918 flags); 1919 1920 if (*devices == AUDIO_DEVICE_NONE) { 1921 return BAD_VALUE; 1922 } 1923 1924 Mutex::Autolock _l(mLock); 1925 1926 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1927 if (thread != 0) { 1928 *latencyMs = thread->latency(); 1929 1930 // notify client processes of the new output creation 1931 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1932 1933 // the first primary output opened designates the primary hw device 1934 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1935 ALOGI("Using module %d has the primary audio interface", module); 1936 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1937 1938 AutoMutex lock(mHardwareLock); 1939 mHardwareStatus = AUDIO_HW_SET_MODE; 1940 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1941 mHardwareStatus = AUDIO_HW_IDLE; 1942 } 1943 return NO_ERROR; 1944 } 1945 1946 return NO_INIT; 1947} 1948 1949audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1950 audio_io_handle_t output2) 1951{ 1952 Mutex::Autolock _l(mLock); 1953 MixerThread *thread1 = checkMixerThread_l(output1); 1954 MixerThread *thread2 = checkMixerThread_l(output2); 1955 1956 if (thread1 == NULL || thread2 == NULL) { 1957 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1958 output2); 1959 return AUDIO_IO_HANDLE_NONE; 1960 } 1961 1962 audio_io_handle_t id = nextUniqueId(AUDIO_UNIQUE_ID_USE_OUTPUT); 1963 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id, mSystemReady); 1964 thread->addOutputTrack(thread2); 1965 mPlaybackThreads.add(id, thread); 1966 // notify client processes of the new output creation 1967 thread->ioConfigChanged(AUDIO_OUTPUT_OPENED); 1968 return id; 1969} 1970 1971status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1972{ 1973 return closeOutput_nonvirtual(output); 1974} 1975 1976status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1977{ 1978 // keep strong reference on the playback thread so that 1979 // it is not destroyed while exit() is executed 1980 sp<PlaybackThread> thread; 1981 { 1982 Mutex::Autolock _l(mLock); 1983 thread = checkPlaybackThread_l(output); 1984 if (thread == NULL) { 1985 return BAD_VALUE; 1986 } 1987 1988 ALOGV("closeOutput() %d", output); 1989 1990 if (thread->type() == ThreadBase::MIXER) { 1991 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1992 if (mPlaybackThreads.valueAt(i)->isDuplicating()) { 1993 DuplicatingThread *dupThread = 1994 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1995 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1996 } 1997 } 1998 } 1999 2000 2001 mPlaybackThreads.removeItem(output); 2002 // save all effects to the default thread 2003 if (mPlaybackThreads.size()) { 2004 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 2005 if (dstThread != NULL) { 2006 // audioflinger lock is held here so the acquisition order of thread locks does not 2007 // matter 2008 Mutex::Autolock _dl(dstThread->mLock); 2009 Mutex::Autolock _sl(thread->mLock); 2010 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2011 for (size_t i = 0; i < effectChains.size(); i ++) { 2012 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 2013 } 2014 } 2015 } 2016 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2017 ioDesc->mIoHandle = output; 2018 ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc); 2019 } 2020 thread->exit(); 2021 // The thread entity (active unit of execution) is no longer running here, 2022 // but the ThreadBase container still exists. 2023 2024 if (!thread->isDuplicating()) { 2025 closeOutputFinish(thread); 2026 } 2027 2028 return NO_ERROR; 2029} 2030 2031void AudioFlinger::closeOutputFinish(const sp<PlaybackThread>& thread) 2032{ 2033 AudioStreamOut *out = thread->clearOutput(); 2034 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 2035 // from now on thread->mOutput is NULL 2036 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 2037 delete out; 2038} 2039 2040void AudioFlinger::closeOutputInternal_l(const sp<PlaybackThread>& thread) 2041{ 2042 mPlaybackThreads.removeItem(thread->mId); 2043 thread->exit(); 2044 closeOutputFinish(thread); 2045} 2046 2047status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 2048{ 2049 Mutex::Autolock _l(mLock); 2050 PlaybackThread *thread = checkPlaybackThread_l(output); 2051 2052 if (thread == NULL) { 2053 return BAD_VALUE; 2054 } 2055 2056 ALOGV("suspendOutput() %d", output); 2057 thread->suspend(); 2058 2059 return NO_ERROR; 2060} 2061 2062status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 2063{ 2064 Mutex::Autolock _l(mLock); 2065 PlaybackThread *thread = checkPlaybackThread_l(output); 2066 2067 if (thread == NULL) { 2068 return BAD_VALUE; 2069 } 2070 2071 ALOGV("restoreOutput() %d", output); 2072 2073 thread->restore(); 2074 2075 return NO_ERROR; 2076} 2077 2078status_t AudioFlinger::openInput(audio_module_handle_t module, 2079 audio_io_handle_t *input, 2080 audio_config_t *config, 2081 audio_devices_t *devices, 2082 const String8& address, 2083 audio_source_t source, 2084 audio_input_flags_t flags) 2085{ 2086 Mutex::Autolock _l(mLock); 2087 2088 if (*devices == AUDIO_DEVICE_NONE) { 2089 return BAD_VALUE; 2090 } 2091 2092 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 2093 2094 if (thread != 0) { 2095 // notify client processes of the new input creation 2096 thread->ioConfigChanged(AUDIO_INPUT_OPENED); 2097 return NO_ERROR; 2098 } 2099 return NO_INIT; 2100} 2101 2102sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2103 audio_io_handle_t *input, 2104 audio_config_t *config, 2105 audio_devices_t devices, 2106 const String8& address, 2107 audio_source_t source, 2108 audio_input_flags_t flags) 2109{ 2110 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2111 if (inHwDev == NULL) { 2112 *input = AUDIO_IO_HANDLE_NONE; 2113 return 0; 2114 } 2115 2116 // Audio Policy can request a specific handle for hardware hotword. 2117 // The goal here is not to re-open an already opened input. 2118 // It is to use a pre-assigned I/O handle. 2119 if (*input == AUDIO_IO_HANDLE_NONE) { 2120 *input = nextUniqueId(AUDIO_UNIQUE_ID_USE_INPUT); 2121 } else if (audio_unique_id_get_use(*input) != AUDIO_UNIQUE_ID_USE_INPUT) { 2122 ALOGE("openInput_l() requested input handle %d is invalid", *input); 2123 return 0; 2124 } else if (mRecordThreads.indexOfKey(*input) >= 0) { 2125 // This should not happen in a transient state with current design. 2126 ALOGE("openInput_l() requested input handle %d is already assigned", *input); 2127 return 0; 2128 } 2129 2130 audio_config_t halconfig = *config; 2131 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2132 audio_stream_in_t *inStream = NULL; 2133 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2134 &inStream, flags, address.string(), source); 2135 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2136 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2137 inStream, 2138 halconfig.sample_rate, 2139 halconfig.format, 2140 halconfig.channel_mask, 2141 flags, 2142 status, address.string()); 2143 2144 // If the input could not be opened with the requested parameters and we can handle the 2145 // conversion internally, try to open again with the proposed parameters. 2146 if (status == BAD_VALUE && 2147 audio_is_linear_pcm(config->format) && 2148 audio_is_linear_pcm(halconfig.format) && 2149 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2150 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) && 2151 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) { 2152 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2153 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2154 inStream = NULL; 2155 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2156 &inStream, flags, address.string(), source); 2157 // FIXME log this new status; HAL should not propose any further changes 2158 } 2159 2160 if (status == NO_ERROR && inStream != NULL) { 2161 2162#ifdef TEE_SINK 2163 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2164 // or (re-)create if current Pipe is idle and does not match the new format 2165 sp<NBAIO_Sink> teeSink; 2166 enum { 2167 TEE_SINK_NO, // don't copy input 2168 TEE_SINK_NEW, // copy input using a new pipe 2169 TEE_SINK_OLD, // copy input using an existing pipe 2170 } kind; 2171 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2172 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2173 if (!mTeeSinkInputEnabled) { 2174 kind = TEE_SINK_NO; 2175 } else if (!Format_isValid(format)) { 2176 kind = TEE_SINK_NO; 2177 } else if (mRecordTeeSink == 0) { 2178 kind = TEE_SINK_NEW; 2179 } else if (mRecordTeeSink->getStrongCount() != 1) { 2180 kind = TEE_SINK_NO; 2181 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2182 kind = TEE_SINK_OLD; 2183 } else { 2184 kind = TEE_SINK_NEW; 2185 } 2186 switch (kind) { 2187 case TEE_SINK_NEW: { 2188 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2189 size_t numCounterOffers = 0; 2190 const NBAIO_Format offers[1] = {format}; 2191 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2192 ALOG_ASSERT(index == 0); 2193 PipeReader *pipeReader = new PipeReader(*pipe); 2194 numCounterOffers = 0; 2195 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2196 ALOG_ASSERT(index == 0); 2197 mRecordTeeSink = pipe; 2198 mRecordTeeSource = pipeReader; 2199 teeSink = pipe; 2200 } 2201 break; 2202 case TEE_SINK_OLD: 2203 teeSink = mRecordTeeSink; 2204 break; 2205 case TEE_SINK_NO: 2206 default: 2207 break; 2208 } 2209#endif 2210 2211 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream, flags); 2212 2213 // Start record thread 2214 // RecordThread requires both input and output device indication to forward to audio 2215 // pre processing modules 2216 sp<RecordThread> thread = new RecordThread(this, 2217 inputStream, 2218 *input, 2219 primaryOutputDevice_l(), 2220 devices, 2221 mSystemReady 2222#ifdef TEE_SINK 2223 , teeSink 2224#endif 2225 ); 2226 mRecordThreads.add(*input, thread); 2227 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2228 return thread; 2229 } 2230 2231 *input = AUDIO_IO_HANDLE_NONE; 2232 return 0; 2233} 2234 2235status_t AudioFlinger::closeInput(audio_io_handle_t input) 2236{ 2237 return closeInput_nonvirtual(input); 2238} 2239 2240status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2241{ 2242 // keep strong reference on the record thread so that 2243 // it is not destroyed while exit() is executed 2244 sp<RecordThread> thread; 2245 { 2246 Mutex::Autolock _l(mLock); 2247 thread = checkRecordThread_l(input); 2248 if (thread == 0) { 2249 return BAD_VALUE; 2250 } 2251 2252 ALOGV("closeInput() %d", input); 2253 2254 // If we still have effect chains, it means that a client still holds a handle 2255 // on at least one effect. We must either move the chain to an existing thread with the 2256 // same session ID or put it aside in case a new record thread is opened for a 2257 // new capture on the same session 2258 sp<EffectChain> chain; 2259 { 2260 Mutex::Autolock _sl(thread->mLock); 2261 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2262 // Note: maximum one chain per record thread 2263 if (effectChains.size() != 0) { 2264 chain = effectChains[0]; 2265 } 2266 } 2267 if (chain != 0) { 2268 // first check if a record thread is already opened with a client on the same session. 2269 // This should only happen in case of overlap between one thread tear down and the 2270 // creation of its replacement 2271 size_t i; 2272 for (i = 0; i < mRecordThreads.size(); i++) { 2273 sp<RecordThread> t = mRecordThreads.valueAt(i); 2274 if (t == thread) { 2275 continue; 2276 } 2277 if (t->hasAudioSession(chain->sessionId()) != 0) { 2278 Mutex::Autolock _l(t->mLock); 2279 ALOGV("closeInput() found thread %d for effect session %d", 2280 t->id(), chain->sessionId()); 2281 t->addEffectChain_l(chain); 2282 break; 2283 } 2284 } 2285 // put the chain aside if we could not find a record thread with the same session id. 2286 if (i == mRecordThreads.size()) { 2287 putOrphanEffectChain_l(chain); 2288 } 2289 } 2290 const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor(); 2291 ioDesc->mIoHandle = input; 2292 ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc); 2293 mRecordThreads.removeItem(input); 2294 } 2295 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2296 // we have a different lock for notification client 2297 closeInputFinish(thread); 2298 return NO_ERROR; 2299} 2300 2301void AudioFlinger::closeInputFinish(const sp<RecordThread>& thread) 2302{ 2303 thread->exit(); 2304 AudioStreamIn *in = thread->clearInput(); 2305 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2306 // from now on thread->mInput is NULL 2307 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2308 delete in; 2309} 2310 2311void AudioFlinger::closeInputInternal_l(const sp<RecordThread>& thread) 2312{ 2313 mRecordThreads.removeItem(thread->mId); 2314 closeInputFinish(thread); 2315} 2316 2317status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2318{ 2319 Mutex::Autolock _l(mLock); 2320 ALOGV("invalidateStream() stream %d", stream); 2321 2322 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2323 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2324 thread->invalidateTracks(stream); 2325 } 2326 2327 return NO_ERROR; 2328} 2329 2330 2331audio_unique_id_t AudioFlinger::newAudioUniqueId(audio_unique_id_use_t use) 2332{ 2333 // This is a binder API, so a malicious client could pass in a bad parameter. 2334 // Check for that before calling the internal API nextUniqueId(). 2335 if ((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX) { 2336 ALOGE("newAudioUniqueId invalid use %d", use); 2337 return AUDIO_UNIQUE_ID_ALLOCATE; 2338 } 2339 return nextUniqueId(use); 2340} 2341 2342void AudioFlinger::acquireAudioSessionId(audio_session_t audioSession, pid_t pid) 2343{ 2344 Mutex::Autolock _l(mLock); 2345 pid_t caller = IPCThreadState::self()->getCallingPid(); 2346 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2347 if (pid != -1 && (caller == getpid_cached)) { 2348 caller = pid; 2349 } 2350 2351 { 2352 Mutex::Autolock _cl(mClientLock); 2353 // Ignore requests received from processes not known as notification client. The request 2354 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2355 // called from a different pid leaving a stale session reference. Also we don't know how 2356 // to clear this reference if the client process dies. 2357 if (mNotificationClients.indexOfKey(caller) < 0) { 2358 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2359 return; 2360 } 2361 } 2362 2363 size_t num = mAudioSessionRefs.size(); 2364 for (size_t i = 0; i< num; i++) { 2365 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2366 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2367 ref->mCnt++; 2368 ALOGV(" incremented refcount to %d", ref->mCnt); 2369 return; 2370 } 2371 } 2372 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2373 ALOGV(" added new entry for %d", audioSession); 2374} 2375 2376void AudioFlinger::releaseAudioSessionId(audio_session_t audioSession, pid_t pid) 2377{ 2378 Mutex::Autolock _l(mLock); 2379 pid_t caller = IPCThreadState::self()->getCallingPid(); 2380 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2381 if (pid != -1 && (caller == getpid_cached)) { 2382 caller = pid; 2383 } 2384 size_t num = mAudioSessionRefs.size(); 2385 for (size_t i = 0; i< num; i++) { 2386 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2387 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2388 ref->mCnt--; 2389 ALOGV(" decremented refcount to %d", ref->mCnt); 2390 if (ref->mCnt == 0) { 2391 mAudioSessionRefs.removeAt(i); 2392 delete ref; 2393 purgeStaleEffects_l(); 2394 } 2395 return; 2396 } 2397 } 2398 // If the caller is mediaserver it is likely that the session being released was acquired 2399 // on behalf of a process not in notification clients and we ignore the warning. 2400 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2401} 2402 2403void AudioFlinger::purgeStaleEffects_l() { 2404 2405 ALOGV("purging stale effects"); 2406 2407 Vector< sp<EffectChain> > chains; 2408 2409 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2410 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2411 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2412 sp<EffectChain> ec = t->mEffectChains[j]; 2413 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2414 chains.push(ec); 2415 } 2416 } 2417 } 2418 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2419 sp<RecordThread> t = mRecordThreads.valueAt(i); 2420 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2421 sp<EffectChain> ec = t->mEffectChains[j]; 2422 chains.push(ec); 2423 } 2424 } 2425 2426 for (size_t i = 0; i < chains.size(); i++) { 2427 sp<EffectChain> ec = chains[i]; 2428 int sessionid = ec->sessionId(); 2429 sp<ThreadBase> t = ec->mThread.promote(); 2430 if (t == 0) { 2431 continue; 2432 } 2433 size_t numsessionrefs = mAudioSessionRefs.size(); 2434 bool found = false; 2435 for (size_t k = 0; k < numsessionrefs; k++) { 2436 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2437 if (ref->mSessionid == sessionid) { 2438 ALOGV(" session %d still exists for %d with %d refs", 2439 sessionid, ref->mPid, ref->mCnt); 2440 found = true; 2441 break; 2442 } 2443 } 2444 if (!found) { 2445 Mutex::Autolock _l(t->mLock); 2446 // remove all effects from the chain 2447 while (ec->mEffects.size()) { 2448 sp<EffectModule> effect = ec->mEffects[0]; 2449 effect->unPin(); 2450 t->removeEffect_l(effect); 2451 if (effect->purgeHandles()) { 2452 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2453 } 2454 AudioSystem::unregisterEffect(effect->id()); 2455 } 2456 } 2457 } 2458 return; 2459} 2460 2461// checkThread_l() must be called with AudioFlinger::mLock held 2462AudioFlinger::ThreadBase *AudioFlinger::checkThread_l(audio_io_handle_t ioHandle) const 2463{ 2464 ThreadBase *thread = NULL; 2465 switch (audio_unique_id_get_use(ioHandle)) { 2466 case AUDIO_UNIQUE_ID_USE_OUTPUT: 2467 thread = checkPlaybackThread_l(ioHandle); 2468 break; 2469 case AUDIO_UNIQUE_ID_USE_INPUT: 2470 thread = checkRecordThread_l(ioHandle); 2471 break; 2472 default: 2473 break; 2474 } 2475 return thread; 2476} 2477 2478// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2479AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2480{ 2481 return mPlaybackThreads.valueFor(output).get(); 2482} 2483 2484// checkMixerThread_l() must be called with AudioFlinger::mLock held 2485AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2486{ 2487 PlaybackThread *thread = checkPlaybackThread_l(output); 2488 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2489} 2490 2491// checkRecordThread_l() must be called with AudioFlinger::mLock held 2492AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2493{ 2494 return mRecordThreads.valueFor(input).get(); 2495} 2496 2497audio_unique_id_t AudioFlinger::nextUniqueId(audio_unique_id_use_t use) 2498{ 2499 // This is the internal API, so it is OK to assert on bad parameter. 2500 LOG_ALWAYS_FATAL_IF((unsigned) use >= (unsigned) AUDIO_UNIQUE_ID_USE_MAX); 2501 const int maxRetries = use == AUDIO_UNIQUE_ID_USE_SESSION ? 3 : 1; 2502 for (int retry = 0; retry < maxRetries; retry++) { 2503 // The cast allows wraparound from max positive to min negative instead of abort 2504 uint32_t base = (uint32_t) atomic_fetch_add_explicit(&mNextUniqueIds[use], 2505 (uint_fast32_t) AUDIO_UNIQUE_ID_USE_MAX, memory_order_acq_rel); 2506 ALOG_ASSERT(audio_unique_id_get_use(base) == AUDIO_UNIQUE_ID_USE_UNSPECIFIED); 2507 // allow wrap by skipping 0 and -1 for session ids 2508 if (!(base == 0 || base == (~0u & ~AUDIO_UNIQUE_ID_USE_MASK))) { 2509 ALOGW_IF(retry != 0, "unique ID overflow for use %d", use); 2510 return (audio_unique_id_t) (base | use); 2511 } 2512 } 2513 // We have no way of recovering from wraparound 2514 LOG_ALWAYS_FATAL("unique ID overflow for use %d", use); 2515 // TODO Use a floor after wraparound. This may need a mutex. 2516} 2517 2518AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2519{ 2520 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2521 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2522 if(thread->isDuplicating()) { 2523 continue; 2524 } 2525 AudioStreamOut *output = thread->getOutput(); 2526 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2527 return thread; 2528 } 2529 } 2530 return NULL; 2531} 2532 2533audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2534{ 2535 PlaybackThread *thread = primaryPlaybackThread_l(); 2536 2537 if (thread == NULL) { 2538 return 0; 2539 } 2540 2541 return thread->outDevice(); 2542} 2543 2544AudioFlinger::PlaybackThread *AudioFlinger::fastPlaybackThread_l() const 2545{ 2546 size_t minFrameCount = 0; 2547 PlaybackThread *minThread = NULL; 2548 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2549 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2550 if (!thread->isDuplicating()) { 2551 size_t frameCount = thread->frameCountHAL(); 2552 if (frameCount != 0 && (minFrameCount == 0 || frameCount < minFrameCount || 2553 (frameCount == minFrameCount && thread->hasFastMixer() && 2554 /*minThread != NULL &&*/ !minThread->hasFastMixer()))) { 2555 minFrameCount = frameCount; 2556 minThread = thread; 2557 } 2558 } 2559 } 2560 return minThread; 2561} 2562 2563sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2564 audio_session_t triggerSession, 2565 audio_session_t listenerSession, 2566 sync_event_callback_t callBack, 2567 const wp<RefBase>& cookie) 2568{ 2569 Mutex::Autolock _l(mLock); 2570 2571 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2572 status_t playStatus = NAME_NOT_FOUND; 2573 status_t recStatus = NAME_NOT_FOUND; 2574 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2575 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2576 if (playStatus == NO_ERROR) { 2577 return event; 2578 } 2579 } 2580 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2581 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2582 if (recStatus == NO_ERROR) { 2583 return event; 2584 } 2585 } 2586 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2587 mPendingSyncEvents.add(event); 2588 } else { 2589 ALOGV("createSyncEvent() invalid event %d", event->type()); 2590 event.clear(); 2591 } 2592 return event; 2593} 2594 2595// ---------------------------------------------------------------------------- 2596// Effect management 2597// ---------------------------------------------------------------------------- 2598 2599 2600status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2601{ 2602 Mutex::Autolock _l(mLock); 2603 return EffectQueryNumberEffects(numEffects); 2604} 2605 2606status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2607{ 2608 Mutex::Autolock _l(mLock); 2609 return EffectQueryEffect(index, descriptor); 2610} 2611 2612status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2613 effect_descriptor_t *descriptor) const 2614{ 2615 Mutex::Autolock _l(mLock); 2616 return EffectGetDescriptor(pUuid, descriptor); 2617} 2618 2619 2620sp<IEffect> AudioFlinger::createEffect( 2621 effect_descriptor_t *pDesc, 2622 const sp<IEffectClient>& effectClient, 2623 int32_t priority, 2624 audio_io_handle_t io, 2625 audio_session_t sessionId, 2626 const String16& opPackageName, 2627 status_t *status, 2628 int *id, 2629 int *enabled) 2630{ 2631 status_t lStatus = NO_ERROR; 2632 sp<EffectHandle> handle; 2633 effect_descriptor_t desc; 2634 2635 pid_t pid = IPCThreadState::self()->getCallingPid(); 2636 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2637 pid, effectClient.get(), priority, sessionId, io); 2638 2639 if (pDesc == NULL) { 2640 lStatus = BAD_VALUE; 2641 goto Exit; 2642 } 2643 2644 // check audio settings permission for global effects 2645 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2646 lStatus = PERMISSION_DENIED; 2647 goto Exit; 2648 } 2649 2650 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2651 // that can only be created by audio policy manager (running in same process) 2652 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2653 lStatus = PERMISSION_DENIED; 2654 goto Exit; 2655 } 2656 2657 { 2658 if (!EffectIsNullUuid(&pDesc->uuid)) { 2659 // if uuid is specified, request effect descriptor 2660 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2661 if (lStatus < 0) { 2662 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2663 goto Exit; 2664 } 2665 } else { 2666 // if uuid is not specified, look for an available implementation 2667 // of the required type in effect factory 2668 if (EffectIsNullUuid(&pDesc->type)) { 2669 ALOGW("createEffect() no effect type"); 2670 lStatus = BAD_VALUE; 2671 goto Exit; 2672 } 2673 uint32_t numEffects = 0; 2674 effect_descriptor_t d; 2675 d.flags = 0; // prevent compiler warning 2676 bool found = false; 2677 2678 lStatus = EffectQueryNumberEffects(&numEffects); 2679 if (lStatus < 0) { 2680 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2681 goto Exit; 2682 } 2683 for (uint32_t i = 0; i < numEffects; i++) { 2684 lStatus = EffectQueryEffect(i, &desc); 2685 if (lStatus < 0) { 2686 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2687 continue; 2688 } 2689 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2690 // If matching type found save effect descriptor. If the session is 2691 // 0 and the effect is not auxiliary, continue enumeration in case 2692 // an auxiliary version of this effect type is available 2693 found = true; 2694 d = desc; 2695 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2696 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2697 break; 2698 } 2699 } 2700 } 2701 if (!found) { 2702 lStatus = BAD_VALUE; 2703 ALOGW("createEffect() effect not found"); 2704 goto Exit; 2705 } 2706 // For same effect type, chose auxiliary version over insert version if 2707 // connect to output mix (Compliance to OpenSL ES) 2708 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2709 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2710 desc = d; 2711 } 2712 } 2713 2714 // Do not allow auxiliary effects on a session different from 0 (output mix) 2715 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2716 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2717 lStatus = INVALID_OPERATION; 2718 goto Exit; 2719 } 2720 2721 // check recording permission for visualizer 2722 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2723 !recordingAllowed(opPackageName, pid, IPCThreadState::self()->getCallingUid())) { 2724 lStatus = PERMISSION_DENIED; 2725 goto Exit; 2726 } 2727 2728 // return effect descriptor 2729 *pDesc = desc; 2730 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2731 // if the output returned by getOutputForEffect() is removed before we lock the 2732 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2733 // and we will exit safely 2734 io = AudioSystem::getOutputForEffect(&desc); 2735 ALOGV("createEffect got output %d", io); 2736 } 2737 2738 Mutex::Autolock _l(mLock); 2739 2740 // If output is not specified try to find a matching audio session ID in one of the 2741 // output threads. 2742 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2743 // because of code checking output when entering the function. 2744 // Note: io is never 0 when creating an effect on an input 2745 if (io == AUDIO_IO_HANDLE_NONE) { 2746 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2747 // output must be specified by AudioPolicyManager when using session 2748 // AUDIO_SESSION_OUTPUT_STAGE 2749 lStatus = BAD_VALUE; 2750 goto Exit; 2751 } 2752 // look for the thread where the specified audio session is present 2753 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2754 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2755 io = mPlaybackThreads.keyAt(i); 2756 break; 2757 } 2758 } 2759 if (io == 0) { 2760 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2761 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2762 io = mRecordThreads.keyAt(i); 2763 break; 2764 } 2765 } 2766 } 2767 // If no output thread contains the requested session ID, default to 2768 // first output. The effect chain will be moved to the correct output 2769 // thread when a track with the same session ID is created 2770 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2771 io = mPlaybackThreads.keyAt(0); 2772 } 2773 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2774 } 2775 ThreadBase *thread = checkRecordThread_l(io); 2776 if (thread == NULL) { 2777 thread = checkPlaybackThread_l(io); 2778 if (thread == NULL) { 2779 ALOGE("createEffect() unknown output thread"); 2780 lStatus = BAD_VALUE; 2781 goto Exit; 2782 } 2783 } else { 2784 // Check if one effect chain was awaiting for an effect to be created on this 2785 // session and used it instead of creating a new one. 2786 sp<EffectChain> chain = getOrphanEffectChain_l(sessionId); 2787 if (chain != 0) { 2788 Mutex::Autolock _l(thread->mLock); 2789 thread->addEffectChain_l(chain); 2790 } 2791 } 2792 2793 sp<Client> client = registerPid(pid); 2794 2795 // create effect on selected output thread 2796 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2797 &desc, enabled, &lStatus); 2798 if (handle != 0 && id != NULL) { 2799 *id = handle->id(); 2800 } 2801 if (handle == 0) { 2802 // remove local strong reference to Client with mClientLock held 2803 Mutex::Autolock _cl(mClientLock); 2804 client.clear(); 2805 } 2806 } 2807 2808Exit: 2809 *status = lStatus; 2810 return handle; 2811} 2812 2813status_t AudioFlinger::moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 2814 audio_io_handle_t dstOutput) 2815{ 2816 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2817 sessionId, srcOutput, dstOutput); 2818 Mutex::Autolock _l(mLock); 2819 if (srcOutput == dstOutput) { 2820 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2821 return NO_ERROR; 2822 } 2823 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2824 if (srcThread == NULL) { 2825 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2826 return BAD_VALUE; 2827 } 2828 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2829 if (dstThread == NULL) { 2830 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2831 return BAD_VALUE; 2832 } 2833 2834 Mutex::Autolock _dl(dstThread->mLock); 2835 Mutex::Autolock _sl(srcThread->mLock); 2836 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2837} 2838 2839// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2840status_t AudioFlinger::moveEffectChain_l(audio_session_t sessionId, 2841 AudioFlinger::PlaybackThread *srcThread, 2842 AudioFlinger::PlaybackThread *dstThread, 2843 bool reRegister) 2844{ 2845 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2846 sessionId, srcThread, dstThread); 2847 2848 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2849 if (chain == 0) { 2850 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2851 sessionId, srcThread); 2852 return INVALID_OPERATION; 2853 } 2854 2855 // Check whether the destination thread and all effects in the chain are compatible 2856 if (!chain->isCompatibleWithThread_l(dstThread)) { 2857 ALOGW("moveEffectChain_l() effect chain failed because" 2858 " destination thread %p is not compatible with effects in the chain", 2859 dstThread); 2860 return INVALID_OPERATION; 2861 } 2862 2863 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2864 // so that a new chain is created with correct parameters when first effect is added. This is 2865 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2866 // removed. 2867 srcThread->removeEffectChain_l(chain); 2868 2869 // transfer all effects one by one so that new effect chain is created on new thread with 2870 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2871 sp<EffectChain> dstChain; 2872 uint32_t strategy = 0; // prevent compiler warning 2873 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2874 Vector< sp<EffectModule> > removed; 2875 status_t status = NO_ERROR; 2876 while (effect != 0) { 2877 srcThread->removeEffect_l(effect); 2878 removed.add(effect); 2879 status = dstThread->addEffect_l(effect); 2880 if (status != NO_ERROR) { 2881 break; 2882 } 2883 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2884 if (effect->state() == EffectModule::ACTIVE || 2885 effect->state() == EffectModule::STOPPING) { 2886 effect->start(); 2887 } 2888 // if the move request is not received from audio policy manager, the effect must be 2889 // re-registered with the new strategy and output 2890 if (dstChain == 0) { 2891 dstChain = effect->chain().promote(); 2892 if (dstChain == 0) { 2893 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2894 status = NO_INIT; 2895 break; 2896 } 2897 strategy = dstChain->strategy(); 2898 } 2899 if (reRegister) { 2900 AudioSystem::unregisterEffect(effect->id()); 2901 AudioSystem::registerEffect(&effect->desc(), 2902 dstThread->id(), 2903 strategy, 2904 sessionId, 2905 effect->id()); 2906 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2907 } 2908 effect = chain->getEffectFromId_l(0); 2909 } 2910 2911 if (status != NO_ERROR) { 2912 for (size_t i = 0; i < removed.size(); i++) { 2913 srcThread->addEffect_l(removed[i]); 2914 if (dstChain != 0 && reRegister) { 2915 AudioSystem::unregisterEffect(removed[i]->id()); 2916 AudioSystem::registerEffect(&removed[i]->desc(), 2917 srcThread->id(), 2918 strategy, 2919 sessionId, 2920 removed[i]->id()); 2921 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2922 } 2923 } 2924 } 2925 2926 return status; 2927} 2928 2929bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2930{ 2931 if (mGlobalEffectEnableTime != 0 && 2932 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2933 return true; 2934 } 2935 2936 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2937 sp<EffectChain> ec = 2938 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2939 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2940 return true; 2941 } 2942 } 2943 return false; 2944} 2945 2946void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2947{ 2948 Mutex::Autolock _l(mLock); 2949 2950 mGlobalEffectEnableTime = systemTime(); 2951 2952 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2953 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2954 if (t->mType == ThreadBase::OFFLOAD) { 2955 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2956 } 2957 } 2958 2959} 2960 2961status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2962{ 2963 audio_session_t session = chain->sessionId(); 2964 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2965 ALOGV("putOrphanEffectChain_l session %d index %zd", session, index); 2966 if (index >= 0) { 2967 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2968 return ALREADY_EXISTS; 2969 } 2970 mOrphanEffectChains.add(session, chain); 2971 return NO_ERROR; 2972} 2973 2974sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2975{ 2976 sp<EffectChain> chain; 2977 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2978 ALOGV("getOrphanEffectChain_l session %d index %zd", session, index); 2979 if (index >= 0) { 2980 chain = mOrphanEffectChains.valueAt(index); 2981 mOrphanEffectChains.removeItemsAt(index); 2982 } 2983 return chain; 2984} 2985 2986bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2987{ 2988 Mutex::Autolock _l(mLock); 2989 audio_session_t session = effect->sessionId(); 2990 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2991 ALOGV("updateOrphanEffectChains session %d index %zd", session, index); 2992 if (index >= 0) { 2993 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2994 if (chain->removeEffect_l(effect) == 0) { 2995 ALOGV("updateOrphanEffectChains removing effect chain at index %zd", index); 2996 mOrphanEffectChains.removeItemsAt(index); 2997 } 2998 return true; 2999 } 3000 return false; 3001} 3002 3003 3004struct Entry { 3005#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 3006 char mFileName[TEE_MAX_FILENAME]; 3007}; 3008 3009int comparEntry(const void *p1, const void *p2) 3010{ 3011 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 3012} 3013 3014#ifdef TEE_SINK 3015void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 3016{ 3017 NBAIO_Source *teeSource = source.get(); 3018 if (teeSource != NULL) { 3019 // .wav rotation 3020 // There is a benign race condition if 2 threads call this simultaneously. 3021 // They would both traverse the directory, but the result would simply be 3022 // failures at unlink() which are ignored. It's also unlikely since 3023 // normally dumpsys is only done by bugreport or from the command line. 3024 char teePath[32+256]; 3025 strcpy(teePath, "/data/misc/audioserver"); 3026 size_t teePathLen = strlen(teePath); 3027 DIR *dir = opendir(teePath); 3028 teePath[teePathLen++] = '/'; 3029 if (dir != NULL) { 3030#define TEE_MAX_SORT 20 // number of entries to sort 3031#define TEE_MAX_KEEP 10 // number of entries to keep 3032 struct Entry entries[TEE_MAX_SORT]; 3033 size_t entryCount = 0; 3034 while (entryCount < TEE_MAX_SORT) { 3035 struct dirent de; 3036 struct dirent *result = NULL; 3037 int rc = readdir_r(dir, &de, &result); 3038 if (rc != 0) { 3039 ALOGW("readdir_r failed %d", rc); 3040 break; 3041 } 3042 if (result == NULL) { 3043 break; 3044 } 3045 if (result != &de) { 3046 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 3047 break; 3048 } 3049 // ignore non .wav file entries 3050 size_t nameLen = strlen(de.d_name); 3051 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 3052 strcmp(&de.d_name[nameLen - 4], ".wav")) { 3053 continue; 3054 } 3055 strcpy(entries[entryCount++].mFileName, de.d_name); 3056 } 3057 (void) closedir(dir); 3058 if (entryCount > TEE_MAX_KEEP) { 3059 qsort(entries, entryCount, sizeof(Entry), comparEntry); 3060 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 3061 strcpy(&teePath[teePathLen], entries[i].mFileName); 3062 (void) unlink(teePath); 3063 } 3064 } 3065 } else { 3066 if (fd >= 0) { 3067 dprintf(fd, "unable to rotate tees in %.*s: %s\n", (int) teePathLen, teePath, 3068 strerror(errno)); 3069 } 3070 } 3071 char teeTime[16]; 3072 struct timeval tv; 3073 gettimeofday(&tv, NULL); 3074 struct tm tm; 3075 localtime_r(&tv.tv_sec, &tm); 3076 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 3077 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 3078 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 3079 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 3080 if (teeFd >= 0) { 3081 // FIXME use libsndfile 3082 char wavHeader[44]; 3083 memcpy(wavHeader, 3084 "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", 3085 sizeof(wavHeader)); 3086 NBAIO_Format format = teeSource->format(); 3087 unsigned channelCount = Format_channelCount(format); 3088 uint32_t sampleRate = Format_sampleRate(format); 3089 size_t frameSize = Format_frameSize(format); 3090 wavHeader[22] = channelCount; // number of channels 3091 wavHeader[24] = sampleRate; // sample rate 3092 wavHeader[25] = sampleRate >> 8; 3093 wavHeader[32] = frameSize; // block alignment 3094 wavHeader[33] = frameSize >> 8; 3095 write(teeFd, wavHeader, sizeof(wavHeader)); 3096 size_t total = 0; 3097 bool firstRead = true; 3098#define TEE_SINK_READ 1024 // frames per I/O operation 3099 void *buffer = malloc(TEE_SINK_READ * frameSize); 3100 for (;;) { 3101 size_t count = TEE_SINK_READ; 3102 ssize_t actual = teeSource->read(buffer, count); 3103 bool wasFirstRead = firstRead; 3104 firstRead = false; 3105 if (actual <= 0) { 3106 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 3107 continue; 3108 } 3109 break; 3110 } 3111 ALOG_ASSERT(actual <= (ssize_t)count); 3112 write(teeFd, buffer, actual * frameSize); 3113 total += actual; 3114 } 3115 free(buffer); 3116 lseek(teeFd, (off_t) 4, SEEK_SET); 3117 uint32_t temp = 44 + total * frameSize - 8; 3118 // FIXME not big-endian safe 3119 write(teeFd, &temp, sizeof(temp)); 3120 lseek(teeFd, (off_t) 40, SEEK_SET); 3121 temp = total * frameSize; 3122 // FIXME not big-endian safe 3123 write(teeFd, &temp, sizeof(temp)); 3124 close(teeFd); 3125 if (fd >= 0) { 3126 dprintf(fd, "tee copied to %s\n", teePath); 3127 } 3128 } else { 3129 if (fd >= 0) { 3130 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 3131 } 3132 } 3133 } 3134} 3135#endif 3136 3137// ---------------------------------------------------------------------------- 3138 3139status_t AudioFlinger::onTransact( 3140 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 3141{ 3142 return BnAudioFlinger::onTransact(code, data, reply, flags); 3143} 3144 3145} // namespace android 3146