AudioFlinger.cpp revision 6770c6faa3467c92eabc5ec9b23d60eb556a0d03
1/* 2** 3** Copyright 2007, The Android Open Source Project 4** 5** Licensed under the Apache License, Version 2.0 (the "License"); 6** you may not use this file except in compliance with the License. 7** You may obtain a copy of the License at 8** 9** http://www.apache.org/licenses/LICENSE-2.0 10** 11** Unless required by applicable law or agreed to in writing, software 12** distributed under the License is distributed on an "AS IS" BASIS, 13** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 14** See the License for the specific language governing permissions and 15** limitations under the License. 16*/ 17 18 19#define LOG_TAG "AudioFlinger" 20//#define LOG_NDEBUG 0 21 22#include "Configuration.h" 23#include <dirent.h> 24#include <math.h> 25#include <signal.h> 26#include <sys/time.h> 27#include <sys/resource.h> 28 29#include <binder/IPCThreadState.h> 30#include <binder/IServiceManager.h> 31#include <utils/Log.h> 32#include <utils/Trace.h> 33#include <binder/Parcel.h> 34#include <utils/String16.h> 35#include <utils/threads.h> 36#include <utils/Atomic.h> 37 38#include <cutils/bitops.h> 39#include <cutils/properties.h> 40 41#include <system/audio.h> 42#include <hardware/audio.h> 43 44#include "AudioMixer.h" 45#include "AudioFlinger.h" 46#include "ServiceUtilities.h" 47 48#include <media/AudioResamplerPublic.h> 49 50#include <media/EffectsFactoryApi.h> 51#include <audio_effects/effect_visualizer.h> 52#include <audio_effects/effect_ns.h> 53#include <audio_effects/effect_aec.h> 54 55#include <audio_utils/primitives.h> 56 57#include <powermanager/PowerManager.h> 58 59#include <common_time/cc_helper.h> 60 61#include <media/IMediaLogService.h> 62 63#include <media/nbaio/Pipe.h> 64#include <media/nbaio/PipeReader.h> 65#include <media/AudioParameter.h> 66#include <private/android_filesystem_config.h> 67 68// ---------------------------------------------------------------------------- 69 70// Note: the following macro is used for extremely verbose logging message. In 71// order to run with ALOG_ASSERT turned on, we need to have LOG_NDEBUG set to 72// 0; but one side effect of this is to turn all LOGV's as well. Some messages 73// are so verbose that we want to suppress them even when we have ALOG_ASSERT 74// turned on. Do not uncomment the #def below unless you really know what you 75// are doing and want to see all of the extremely verbose messages. 76//#define VERY_VERY_VERBOSE_LOGGING 77#ifdef VERY_VERY_VERBOSE_LOGGING 78#define ALOGVV ALOGV 79#else 80#define ALOGVV(a...) do { } while(0) 81#endif 82 83namespace android { 84 85static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n"; 86static const char kHardwareLockedString[] = "Hardware lock is taken\n"; 87static const char kClientLockedString[] = "Client lock is taken\n"; 88 89 90nsecs_t AudioFlinger::mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 91 92uint32_t AudioFlinger::mScreenState; 93 94#ifdef TEE_SINK 95bool AudioFlinger::mTeeSinkInputEnabled = false; 96bool AudioFlinger::mTeeSinkOutputEnabled = false; 97bool AudioFlinger::mTeeSinkTrackEnabled = false; 98 99size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault; 100size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault; 101size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault; 102#endif 103 104// In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off 105// we define a minimum time during which a global effect is considered enabled. 106static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200); 107 108// ---------------------------------------------------------------------------- 109 110const char *formatToString(audio_format_t format) { 111 switch (format & AUDIO_FORMAT_MAIN_MASK) { 112 case AUDIO_FORMAT_PCM: 113 switch (format) { 114 case AUDIO_FORMAT_PCM_16_BIT: return "pcm16"; 115 case AUDIO_FORMAT_PCM_8_BIT: return "pcm8"; 116 case AUDIO_FORMAT_PCM_32_BIT: return "pcm32"; 117 case AUDIO_FORMAT_PCM_8_24_BIT: return "pcm8.24"; 118 case AUDIO_FORMAT_PCM_FLOAT: return "pcmfloat"; 119 case AUDIO_FORMAT_PCM_24_BIT_PACKED: return "pcm24"; 120 default: 121 break; 122 } 123 break; 124 case AUDIO_FORMAT_MP3: return "mp3"; 125 case AUDIO_FORMAT_AMR_NB: return "amr-nb"; 126 case AUDIO_FORMAT_AMR_WB: return "amr-wb"; 127 case AUDIO_FORMAT_AAC: return "aac"; 128 case AUDIO_FORMAT_HE_AAC_V1: return "he-aac-v1"; 129 case AUDIO_FORMAT_HE_AAC_V2: return "he-aac-v2"; 130 case AUDIO_FORMAT_VORBIS: return "vorbis"; 131 case AUDIO_FORMAT_OPUS: return "opus"; 132 case AUDIO_FORMAT_AC3: return "ac-3"; 133 case AUDIO_FORMAT_E_AC3: return "e-ac-3"; 134 default: 135 break; 136 } 137 return "unknown"; 138} 139 140static int load_audio_interface(const char *if_name, audio_hw_device_t **dev) 141{ 142 const hw_module_t *mod; 143 int rc; 144 145 rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod); 146 ALOGE_IF(rc, "%s couldn't load audio hw module %s.%s (%s)", __func__, 147 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 148 if (rc) { 149 goto out; 150 } 151 rc = audio_hw_device_open(mod, dev); 152 ALOGE_IF(rc, "%s couldn't open audio hw device in %s.%s (%s)", __func__, 153 AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc)); 154 if (rc) { 155 goto out; 156 } 157 if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) { 158 ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version); 159 rc = BAD_VALUE; 160 goto out; 161 } 162 return 0; 163 164out: 165 *dev = NULL; 166 return rc; 167} 168 169// ---------------------------------------------------------------------------- 170 171AudioFlinger::AudioFlinger() 172 : BnAudioFlinger(), 173 mPrimaryHardwareDev(NULL), 174 mAudioHwDevs(NULL), 175 mHardwareStatus(AUDIO_HW_IDLE), 176 mMasterVolume(1.0f), 177 mMasterMute(false), 178 mNextUniqueId(1), 179 mMode(AUDIO_MODE_INVALID), 180 mBtNrecIsOff(false), 181 mIsLowRamDevice(true), 182 mIsDeviceTypeKnown(false), 183 mGlobalEffectEnableTime(0), 184 mPrimaryOutputSampleRate(0) 185{ 186 getpid_cached = getpid(); 187 char value[PROPERTY_VALUE_MAX]; 188 bool doLog = (property_get("ro.test_harness", value, "0") > 0) && (atoi(value) == 1); 189 if (doLog) { 190 mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters", 191 MemoryHeapBase::READ_ONLY); 192 } 193 194#ifdef TEE_SINK 195 (void) property_get("ro.debuggable", value, "0"); 196 int debuggable = atoi(value); 197 int teeEnabled = 0; 198 if (debuggable) { 199 (void) property_get("af.tee", value, "0"); 200 teeEnabled = atoi(value); 201 } 202 // FIXME symbolic constants here 203 if (teeEnabled & 1) { 204 mTeeSinkInputEnabled = true; 205 } 206 if (teeEnabled & 2) { 207 mTeeSinkOutputEnabled = true; 208 } 209 if (teeEnabled & 4) { 210 mTeeSinkTrackEnabled = true; 211 } 212#endif 213} 214 215void AudioFlinger::onFirstRef() 216{ 217 int rc = 0; 218 219 Mutex::Autolock _l(mLock); 220 221 /* TODO: move all this work into an Init() function */ 222 char val_str[PROPERTY_VALUE_MAX] = { 0 }; 223 if (property_get("ro.audio.flinger_standbytime_ms", val_str, NULL) >= 0) { 224 uint32_t int_val; 225 if (1 == sscanf(val_str, "%u", &int_val)) { 226 mStandbyTimeInNsecs = milliseconds(int_val); 227 ALOGI("Using %u mSec as standby time.", int_val); 228 } else { 229 mStandbyTimeInNsecs = kDefaultStandbyTimeInNsecs; 230 ALOGI("Using default %u mSec as standby time.", 231 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 232 } 233 } 234 235 mPatchPanel = new PatchPanel(this); 236 237 mMode = AUDIO_MODE_NORMAL; 238} 239 240AudioFlinger::~AudioFlinger() 241{ 242 while (!mRecordThreads.isEmpty()) { 243 // closeInput_nonvirtual() will remove specified entry from mRecordThreads 244 closeInput_nonvirtual(mRecordThreads.keyAt(0)); 245 } 246 while (!mPlaybackThreads.isEmpty()) { 247 // closeOutput_nonvirtual() will remove specified entry from mPlaybackThreads 248 closeOutput_nonvirtual(mPlaybackThreads.keyAt(0)); 249 } 250 251 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 252 // no mHardwareLock needed, as there are no other references to this 253 audio_hw_device_close(mAudioHwDevs.valueAt(i)->hwDevice()); 254 delete mAudioHwDevs.valueAt(i); 255 } 256 257 // Tell media.log service about any old writers that still need to be unregistered 258 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 259 if (binder != 0) { 260 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 261 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 262 sp<IMemory> iMemory(mUnregisteredWriters.top()->getIMemory()); 263 mUnregisteredWriters.pop(); 264 mediaLogService->unregisterWriter(iMemory); 265 } 266 } 267 268} 269 270static const char * const audio_interfaces[] = { 271 AUDIO_HARDWARE_MODULE_ID_PRIMARY, 272 AUDIO_HARDWARE_MODULE_ID_A2DP, 273 AUDIO_HARDWARE_MODULE_ID_USB, 274}; 275#define ARRAY_SIZE(x) (sizeof((x))/sizeof(((x)[0]))) 276 277AudioHwDevice* AudioFlinger::findSuitableHwDev_l( 278 audio_module_handle_t module, 279 audio_devices_t devices) 280{ 281 // if module is 0, the request comes from an old policy manager and we should load 282 // well known modules 283 if (module == 0) { 284 ALOGW("findSuitableHwDev_l() loading well know audio hw modules"); 285 for (size_t i = 0; i < ARRAY_SIZE(audio_interfaces); i++) { 286 loadHwModule_l(audio_interfaces[i]); 287 } 288 // then try to find a module supporting the requested device. 289 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 290 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(i); 291 audio_hw_device_t *dev = audioHwDevice->hwDevice(); 292 if ((dev->get_supported_devices != NULL) && 293 (dev->get_supported_devices(dev) & devices) == devices) 294 return audioHwDevice; 295 } 296 } else { 297 // check a match for the requested module handle 298 AudioHwDevice *audioHwDevice = mAudioHwDevs.valueFor(module); 299 if (audioHwDevice != NULL) { 300 return audioHwDevice; 301 } 302 } 303 304 return NULL; 305} 306 307void AudioFlinger::dumpClients(int fd, const Vector<String16>& args __unused) 308{ 309 const size_t SIZE = 256; 310 char buffer[SIZE]; 311 String8 result; 312 313 result.append("Clients:\n"); 314 for (size_t i = 0; i < mClients.size(); ++i) { 315 sp<Client> client = mClients.valueAt(i).promote(); 316 if (client != 0) { 317 snprintf(buffer, SIZE, " pid: %d\n", client->pid()); 318 result.append(buffer); 319 } 320 } 321 322 result.append("Notification Clients:\n"); 323 for (size_t i = 0; i < mNotificationClients.size(); ++i) { 324 snprintf(buffer, SIZE, " pid: %d\n", mNotificationClients.keyAt(i)); 325 result.append(buffer); 326 } 327 328 result.append("Global session refs:\n"); 329 result.append(" session pid count\n"); 330 for (size_t i = 0; i < mAudioSessionRefs.size(); i++) { 331 AudioSessionRef *r = mAudioSessionRefs[i]; 332 snprintf(buffer, SIZE, " %7d %5d %5d\n", r->mSessionid, r->mPid, r->mCnt); 333 result.append(buffer); 334 } 335 write(fd, result.string(), result.size()); 336} 337 338 339void AudioFlinger::dumpInternals(int fd, const Vector<String16>& args __unused) 340{ 341 const size_t SIZE = 256; 342 char buffer[SIZE]; 343 String8 result; 344 hardware_call_state hardwareStatus = mHardwareStatus; 345 346 snprintf(buffer, SIZE, "Hardware status: %d\n" 347 "Standby Time mSec: %u\n", 348 hardwareStatus, 349 (uint32_t)(mStandbyTimeInNsecs / 1000000)); 350 result.append(buffer); 351 write(fd, result.string(), result.size()); 352} 353 354void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused) 355{ 356 const size_t SIZE = 256; 357 char buffer[SIZE]; 358 String8 result; 359 snprintf(buffer, SIZE, "Permission Denial: " 360 "can't dump AudioFlinger from pid=%d, uid=%d\n", 361 IPCThreadState::self()->getCallingPid(), 362 IPCThreadState::self()->getCallingUid()); 363 result.append(buffer); 364 write(fd, result.string(), result.size()); 365} 366 367bool AudioFlinger::dumpTryLock(Mutex& mutex) 368{ 369 bool locked = false; 370 for (int i = 0; i < kDumpLockRetries; ++i) { 371 if (mutex.tryLock() == NO_ERROR) { 372 locked = true; 373 break; 374 } 375 usleep(kDumpLockSleepUs); 376 } 377 return locked; 378} 379 380status_t AudioFlinger::dump(int fd, const Vector<String16>& args) 381{ 382 if (!dumpAllowed()) { 383 dumpPermissionDenial(fd, args); 384 } else { 385 // get state of hardware lock 386 bool hardwareLocked = dumpTryLock(mHardwareLock); 387 if (!hardwareLocked) { 388 String8 result(kHardwareLockedString); 389 write(fd, result.string(), result.size()); 390 } else { 391 mHardwareLock.unlock(); 392 } 393 394 bool locked = dumpTryLock(mLock); 395 396 // failed to lock - AudioFlinger is probably deadlocked 397 if (!locked) { 398 String8 result(kDeadlockedString); 399 write(fd, result.string(), result.size()); 400 } 401 402 bool clientLocked = dumpTryLock(mClientLock); 403 if (!clientLocked) { 404 String8 result(kClientLockedString); 405 write(fd, result.string(), result.size()); 406 } 407 408 EffectDumpEffects(fd); 409 410 dumpClients(fd, args); 411 if (clientLocked) { 412 mClientLock.unlock(); 413 } 414 415 dumpInternals(fd, args); 416 417 // dump playback threads 418 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 419 mPlaybackThreads.valueAt(i)->dump(fd, args); 420 } 421 422 // dump record threads 423 for (size_t i = 0; i < mRecordThreads.size(); i++) { 424 mRecordThreads.valueAt(i)->dump(fd, args); 425 } 426 427 // dump orphan effect chains 428 if (mOrphanEffectChains.size() != 0) { 429 write(fd, " Orphan Effect Chains\n", strlen(" Orphan Effect Chains\n")); 430 for (size_t i = 0; i < mOrphanEffectChains.size(); i++) { 431 mOrphanEffectChains.valueAt(i)->dump(fd, args); 432 } 433 } 434 // dump all hardware devs 435 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 436 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 437 dev->dump(dev, fd); 438 } 439 440#ifdef TEE_SINK 441 // dump the serially shared record tee sink 442 if (mRecordTeeSource != 0) { 443 dumpTee(fd, mRecordTeeSource); 444 } 445#endif 446 447 if (locked) { 448 mLock.unlock(); 449 } 450 451 // append a copy of media.log here by forwarding fd to it, but don't attempt 452 // to lookup the service if it's not running, as it will block for a second 453 if (mLogMemoryDealer != 0) { 454 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 455 if (binder != 0) { 456 dprintf(fd, "\nmedia.log:\n"); 457 Vector<String16> args; 458 binder->dump(fd, args); 459 } 460 } 461 } 462 return NO_ERROR; 463} 464 465sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid) 466{ 467 Mutex::Autolock _cl(mClientLock); 468 // If pid is already in the mClients wp<> map, then use that entry 469 // (for which promote() is always != 0), otherwise create a new entry and Client. 470 sp<Client> client = mClients.valueFor(pid).promote(); 471 if (client == 0) { 472 client = new Client(this, pid); 473 mClients.add(pid, client); 474 } 475 476 return client; 477} 478 479sp<NBLog::Writer> AudioFlinger::newWriter_l(size_t size, const char *name) 480{ 481 // If there is no memory allocated for logs, return a dummy writer that does nothing 482 if (mLogMemoryDealer == 0) { 483 return new NBLog::Writer(); 484 } 485 sp<IBinder> binder = defaultServiceManager()->getService(String16("media.log")); 486 // Similarly if we can't contact the media.log service, also return a dummy writer 487 if (binder == 0) { 488 return new NBLog::Writer(); 489 } 490 sp<IMediaLogService> mediaLogService(interface_cast<IMediaLogService>(binder)); 491 sp<IMemory> shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 492 // If allocation fails, consult the vector of previously unregistered writers 493 // and garbage-collect one or more them until an allocation succeeds 494 if (shared == 0) { 495 Mutex::Autolock _l(mUnregisteredWritersLock); 496 for (size_t count = mUnregisteredWriters.size(); count > 0; count--) { 497 { 498 // Pick the oldest stale writer to garbage-collect 499 sp<IMemory> iMemory(mUnregisteredWriters[0]->getIMemory()); 500 mUnregisteredWriters.removeAt(0); 501 mediaLogService->unregisterWriter(iMemory); 502 // Now the media.log remote reference to IMemory is gone. When our last local 503 // reference to IMemory also drops to zero at end of this block, 504 // the IMemory destructor will deallocate the region from mLogMemoryDealer. 505 } 506 // Re-attempt the allocation 507 shared = mLogMemoryDealer->allocate(NBLog::Timeline::sharedSize(size)); 508 if (shared != 0) { 509 goto success; 510 } 511 } 512 // Even after garbage-collecting all old writers, there is still not enough memory, 513 // so return a dummy writer 514 return new NBLog::Writer(); 515 } 516success: 517 mediaLogService->registerWriter(shared, size, name); 518 return new NBLog::Writer(size, shared); 519} 520 521void AudioFlinger::unregisterWriter(const sp<NBLog::Writer>& writer) 522{ 523 if (writer == 0) { 524 return; 525 } 526 sp<IMemory> iMemory(writer->getIMemory()); 527 if (iMemory == 0) { 528 return; 529 } 530 // Rather than removing the writer immediately, append it to a queue of old writers to 531 // be garbage-collected later. This allows us to continue to view old logs for a while. 532 Mutex::Autolock _l(mUnregisteredWritersLock); 533 mUnregisteredWriters.push(writer); 534} 535 536// IAudioFlinger interface 537 538 539sp<IAudioTrack> AudioFlinger::createTrack( 540 audio_stream_type_t streamType, 541 uint32_t sampleRate, 542 audio_format_t format, 543 audio_channel_mask_t channelMask, 544 size_t *frameCount, 545 IAudioFlinger::track_flags_t *flags, 546 const sp<IMemory>& sharedBuffer, 547 audio_io_handle_t output, 548 pid_t tid, 549 int *sessionId, 550 int clientUid, 551 status_t *status) 552{ 553 sp<PlaybackThread::Track> track; 554 sp<TrackHandle> trackHandle; 555 sp<Client> client; 556 status_t lStatus; 557 int lSessionId; 558 559 // client AudioTrack::set already implements AUDIO_STREAM_DEFAULT => AUDIO_STREAM_MUSIC, 560 // but if someone uses binder directly they could bypass that and cause us to crash 561 if (uint32_t(streamType) >= AUDIO_STREAM_CNT) { 562 ALOGE("createTrack() invalid stream type %d", streamType); 563 lStatus = BAD_VALUE; 564 goto Exit; 565 } 566 567 // further sample rate checks are performed by createTrack_l() depending on the thread type 568 if (sampleRate == 0) { 569 ALOGE("createTrack() invalid sample rate %u", sampleRate); 570 lStatus = BAD_VALUE; 571 goto Exit; 572 } 573 574 // further channel mask checks are performed by createTrack_l() depending on the thread type 575 if (!audio_is_output_channel(channelMask)) { 576 ALOGE("createTrack() invalid channel mask %#x", channelMask); 577 lStatus = BAD_VALUE; 578 goto Exit; 579 } 580 581 // further format checks are performed by createTrack_l() depending on the thread type 582 if (!audio_is_valid_format(format)) { 583 ALOGE("createTrack() invalid format %#x", format); 584 lStatus = BAD_VALUE; 585 goto Exit; 586 } 587 588 if (sharedBuffer != 0 && sharedBuffer->pointer() == NULL) { 589 ALOGE("createTrack() sharedBuffer is non-0 but has NULL pointer()"); 590 lStatus = BAD_VALUE; 591 goto Exit; 592 } 593 594 { 595 Mutex::Autolock _l(mLock); 596 PlaybackThread *thread = checkPlaybackThread_l(output); 597 if (thread == NULL) { 598 ALOGE("no playback thread found for output handle %d", output); 599 lStatus = BAD_VALUE; 600 goto Exit; 601 } 602 603 pid_t pid = IPCThreadState::self()->getCallingPid(); 604 client = registerPid(pid); 605 606 PlaybackThread *effectThread = NULL; 607 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 608 lSessionId = *sessionId; 609 // check if an effect chain with the same session ID is present on another 610 // output thread and move it here. 611 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 612 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 613 if (mPlaybackThreads.keyAt(i) != output) { 614 uint32_t sessions = t->hasAudioSession(lSessionId); 615 if (sessions & PlaybackThread::EFFECT_SESSION) { 616 effectThread = t.get(); 617 break; 618 } 619 } 620 } 621 } else { 622 // if no audio session id is provided, create one here 623 lSessionId = nextUniqueId(); 624 if (sessionId != NULL) { 625 *sessionId = lSessionId; 626 } 627 } 628 ALOGV("createTrack() lSessionId: %d", lSessionId); 629 630 track = thread->createTrack_l(client, streamType, sampleRate, format, 631 channelMask, frameCount, sharedBuffer, lSessionId, flags, tid, clientUid, &lStatus); 632 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0)); 633 // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless 634 635 // move effect chain to this output thread if an effect on same session was waiting 636 // for a track to be created 637 if (lStatus == NO_ERROR && effectThread != NULL) { 638 // no risk of deadlock because AudioFlinger::mLock is held 639 Mutex::Autolock _dl(thread->mLock); 640 Mutex::Autolock _sl(effectThread->mLock); 641 moveEffectChain_l(lSessionId, effectThread, thread, true); 642 } 643 644 // Look for sync events awaiting for a session to be used. 645 for (size_t i = 0; i < mPendingSyncEvents.size(); i++) { 646 if (mPendingSyncEvents[i]->triggerSession() == lSessionId) { 647 if (thread->isValidSyncEvent(mPendingSyncEvents[i])) { 648 if (lStatus == NO_ERROR) { 649 (void) track->setSyncEvent(mPendingSyncEvents[i]); 650 } else { 651 mPendingSyncEvents[i]->cancel(); 652 } 653 mPendingSyncEvents.removeAt(i); 654 i--; 655 } 656 } 657 } 658 659 setAudioHwSyncForSession_l(thread, (audio_session_t)lSessionId); 660 } 661 662 if (lStatus != NO_ERROR) { 663 // remove local strong reference to Client before deleting the Track so that the 664 // Client destructor is called by the TrackBase destructor with mClientLock held 665 // Don't hold mClientLock when releasing the reference on the track as the 666 // destructor will acquire it. 667 { 668 Mutex::Autolock _cl(mClientLock); 669 client.clear(); 670 } 671 track.clear(); 672 goto Exit; 673 } 674 675 // return handle to client 676 trackHandle = new TrackHandle(track); 677 678Exit: 679 *status = lStatus; 680 return trackHandle; 681} 682 683uint32_t AudioFlinger::sampleRate(audio_io_handle_t output) const 684{ 685 Mutex::Autolock _l(mLock); 686 PlaybackThread *thread = checkPlaybackThread_l(output); 687 if (thread == NULL) { 688 ALOGW("sampleRate() unknown thread %d", output); 689 return 0; 690 } 691 return thread->sampleRate(); 692} 693 694audio_format_t AudioFlinger::format(audio_io_handle_t output) const 695{ 696 Mutex::Autolock _l(mLock); 697 PlaybackThread *thread = checkPlaybackThread_l(output); 698 if (thread == NULL) { 699 ALOGW("format() unknown thread %d", output); 700 return AUDIO_FORMAT_INVALID; 701 } 702 return thread->format(); 703} 704 705size_t AudioFlinger::frameCount(audio_io_handle_t output) const 706{ 707 Mutex::Autolock _l(mLock); 708 PlaybackThread *thread = checkPlaybackThread_l(output); 709 if (thread == NULL) { 710 ALOGW("frameCount() unknown thread %d", output); 711 return 0; 712 } 713 // FIXME currently returns the normal mixer's frame count to avoid confusing legacy callers; 714 // should examine all callers and fix them to handle smaller counts 715 return thread->frameCount(); 716} 717 718uint32_t AudioFlinger::latency(audio_io_handle_t output) const 719{ 720 Mutex::Autolock _l(mLock); 721 PlaybackThread *thread = checkPlaybackThread_l(output); 722 if (thread == NULL) { 723 ALOGW("latency(): no playback thread found for output handle %d", output); 724 return 0; 725 } 726 return thread->latency(); 727} 728 729status_t AudioFlinger::setMasterVolume(float value) 730{ 731 status_t ret = initCheck(); 732 if (ret != NO_ERROR) { 733 return ret; 734 } 735 736 // check calling permissions 737 if (!settingsAllowed()) { 738 return PERMISSION_DENIED; 739 } 740 741 Mutex::Autolock _l(mLock); 742 mMasterVolume = value; 743 744 // Set master volume in the HALs which support it. 745 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 746 AutoMutex lock(mHardwareLock); 747 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 748 749 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 750 if (dev->canSetMasterVolume()) { 751 dev->hwDevice()->set_master_volume(dev->hwDevice(), value); 752 } 753 mHardwareStatus = AUDIO_HW_IDLE; 754 } 755 756 // Now set the master volume in each playback thread. Playback threads 757 // assigned to HALs which do not have master volume support will apply 758 // master volume during the mix operation. Threads with HALs which do 759 // support master volume will simply ignore the setting. 760 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 761 mPlaybackThreads.valueAt(i)->setMasterVolume(value); 762 763 return NO_ERROR; 764} 765 766status_t AudioFlinger::setMode(audio_mode_t mode) 767{ 768 status_t ret = initCheck(); 769 if (ret != NO_ERROR) { 770 return ret; 771 } 772 773 // check calling permissions 774 if (!settingsAllowed()) { 775 return PERMISSION_DENIED; 776 } 777 if (uint32_t(mode) >= AUDIO_MODE_CNT) { 778 ALOGW("Illegal value: setMode(%d)", mode); 779 return BAD_VALUE; 780 } 781 782 { // scope for the lock 783 AutoMutex lock(mHardwareLock); 784 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 785 mHardwareStatus = AUDIO_HW_SET_MODE; 786 ret = dev->set_mode(dev, mode); 787 mHardwareStatus = AUDIO_HW_IDLE; 788 } 789 790 if (NO_ERROR == ret) { 791 Mutex::Autolock _l(mLock); 792 mMode = mode; 793 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 794 mPlaybackThreads.valueAt(i)->setMode(mode); 795 } 796 797 return ret; 798} 799 800status_t AudioFlinger::setMicMute(bool state) 801{ 802 status_t ret = initCheck(); 803 if (ret != NO_ERROR) { 804 return ret; 805 } 806 807 // check calling permissions 808 if (!settingsAllowed()) { 809 return PERMISSION_DENIED; 810 } 811 812 AutoMutex lock(mHardwareLock); 813 mHardwareStatus = AUDIO_HW_SET_MIC_MUTE; 814 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 815 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 816 status_t result = dev->set_mic_mute(dev, state); 817 if (result != NO_ERROR) { 818 ret = result; 819 } 820 } 821 mHardwareStatus = AUDIO_HW_IDLE; 822 return ret; 823} 824 825bool AudioFlinger::getMicMute() const 826{ 827 status_t ret = initCheck(); 828 if (ret != NO_ERROR) { 829 return false; 830 } 831 bool mute = true; 832 bool state = AUDIO_MODE_INVALID; 833 AutoMutex lock(mHardwareLock); 834 mHardwareStatus = AUDIO_HW_GET_MIC_MUTE; 835 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 836 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 837 status_t result = dev->get_mic_mute(dev, &state); 838 if (result == NO_ERROR) { 839 mute = mute && state; 840 } 841 } 842 mHardwareStatus = AUDIO_HW_IDLE; 843 844 return mute; 845} 846 847status_t AudioFlinger::setMasterMute(bool muted) 848{ 849 status_t ret = initCheck(); 850 if (ret != NO_ERROR) { 851 return ret; 852 } 853 854 // check calling permissions 855 if (!settingsAllowed()) { 856 return PERMISSION_DENIED; 857 } 858 859 Mutex::Autolock _l(mLock); 860 mMasterMute = muted; 861 862 // Set master mute in the HALs which support it. 863 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 864 AutoMutex lock(mHardwareLock); 865 AudioHwDevice *dev = mAudioHwDevs.valueAt(i); 866 867 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 868 if (dev->canSetMasterMute()) { 869 dev->hwDevice()->set_master_mute(dev->hwDevice(), muted); 870 } 871 mHardwareStatus = AUDIO_HW_IDLE; 872 } 873 874 // Now set the master mute in each playback thread. Playback threads 875 // assigned to HALs which do not have master mute support will apply master 876 // mute during the mix operation. Threads with HALs which do support master 877 // mute will simply ignore the setting. 878 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 879 mPlaybackThreads.valueAt(i)->setMasterMute(muted); 880 881 return NO_ERROR; 882} 883 884float AudioFlinger::masterVolume() const 885{ 886 Mutex::Autolock _l(mLock); 887 return masterVolume_l(); 888} 889 890bool AudioFlinger::masterMute() const 891{ 892 Mutex::Autolock _l(mLock); 893 return masterMute_l(); 894} 895 896float AudioFlinger::masterVolume_l() const 897{ 898 return mMasterVolume; 899} 900 901bool AudioFlinger::masterMute_l() const 902{ 903 return mMasterMute; 904} 905 906status_t AudioFlinger::checkStreamType(audio_stream_type_t stream) const 907{ 908 if (uint32_t(stream) >= AUDIO_STREAM_CNT) { 909 ALOGW("setStreamVolume() invalid stream %d", stream); 910 return BAD_VALUE; 911 } 912 pid_t caller = IPCThreadState::self()->getCallingPid(); 913 if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) { 914 ALOGW("setStreamVolume() pid %d cannot use internal stream type %d", caller, stream); 915 return PERMISSION_DENIED; 916 } 917 918 return NO_ERROR; 919} 920 921status_t AudioFlinger::setStreamVolume(audio_stream_type_t stream, float value, 922 audio_io_handle_t output) 923{ 924 // check calling permissions 925 if (!settingsAllowed()) { 926 return PERMISSION_DENIED; 927 } 928 929 status_t status = checkStreamType(stream); 930 if (status != NO_ERROR) { 931 return status; 932 } 933 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to change AUDIO_STREAM_PATCH volume"); 934 935 AutoMutex lock(mLock); 936 PlaybackThread *thread = NULL; 937 if (output != AUDIO_IO_HANDLE_NONE) { 938 thread = checkPlaybackThread_l(output); 939 if (thread == NULL) { 940 return BAD_VALUE; 941 } 942 } 943 944 mStreamTypes[stream].volume = value; 945 946 if (thread == NULL) { 947 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 948 mPlaybackThreads.valueAt(i)->setStreamVolume(stream, value); 949 } 950 } else { 951 thread->setStreamVolume(stream, value); 952 } 953 954 return NO_ERROR; 955} 956 957status_t AudioFlinger::setStreamMute(audio_stream_type_t stream, bool muted) 958{ 959 // check calling permissions 960 if (!settingsAllowed()) { 961 return PERMISSION_DENIED; 962 } 963 964 status_t status = checkStreamType(stream); 965 if (status != NO_ERROR) { 966 return status; 967 } 968 ALOG_ASSERT(stream != AUDIO_STREAM_PATCH, "attempt to mute AUDIO_STREAM_PATCH"); 969 970 if (uint32_t(stream) == AUDIO_STREAM_ENFORCED_AUDIBLE) { 971 ALOGE("setStreamMute() invalid stream %d", stream); 972 return BAD_VALUE; 973 } 974 975 AutoMutex lock(mLock); 976 mStreamTypes[stream].mute = muted; 977 for (size_t i = 0; i < mPlaybackThreads.size(); i++) 978 mPlaybackThreads.valueAt(i)->setStreamMute(stream, muted); 979 980 return NO_ERROR; 981} 982 983float AudioFlinger::streamVolume(audio_stream_type_t stream, audio_io_handle_t output) const 984{ 985 status_t status = checkStreamType(stream); 986 if (status != NO_ERROR) { 987 return 0.0f; 988 } 989 990 AutoMutex lock(mLock); 991 float volume; 992 if (output != AUDIO_IO_HANDLE_NONE) { 993 PlaybackThread *thread = checkPlaybackThread_l(output); 994 if (thread == NULL) { 995 return 0.0f; 996 } 997 volume = thread->streamVolume(stream); 998 } else { 999 volume = streamVolume_l(stream); 1000 } 1001 1002 return volume; 1003} 1004 1005bool AudioFlinger::streamMute(audio_stream_type_t stream) const 1006{ 1007 status_t status = checkStreamType(stream); 1008 if (status != NO_ERROR) { 1009 return true; 1010 } 1011 1012 AutoMutex lock(mLock); 1013 return streamMute_l(stream); 1014} 1015 1016status_t AudioFlinger::setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) 1017{ 1018 ALOGV("setParameters(): io %d, keyvalue %s, calling pid %d", 1019 ioHandle, keyValuePairs.string(), IPCThreadState::self()->getCallingPid()); 1020 1021 // check calling permissions 1022 if (!settingsAllowed()) { 1023 return PERMISSION_DENIED; 1024 } 1025 1026 // AUDIO_IO_HANDLE_NONE means the parameters are global to the audio hardware interface 1027 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1028 Mutex::Autolock _l(mLock); 1029 status_t final_result = NO_ERROR; 1030 { 1031 AutoMutex lock(mHardwareLock); 1032 mHardwareStatus = AUDIO_HW_SET_PARAMETER; 1033 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1034 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1035 status_t result = dev->set_parameters(dev, keyValuePairs.string()); 1036 final_result = result ?: final_result; 1037 } 1038 mHardwareStatus = AUDIO_HW_IDLE; 1039 } 1040 // disable AEC and NS if the device is a BT SCO headset supporting those pre processings 1041 AudioParameter param = AudioParameter(keyValuePairs); 1042 String8 value; 1043 if (param.get(String8(AUDIO_PARAMETER_KEY_BT_NREC), value) == NO_ERROR) { 1044 bool btNrecIsOff = (value == AUDIO_PARAMETER_VALUE_OFF); 1045 if (mBtNrecIsOff != btNrecIsOff) { 1046 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1047 sp<RecordThread> thread = mRecordThreads.valueAt(i); 1048 audio_devices_t device = thread->inDevice(); 1049 bool suspend = audio_is_bluetooth_sco_device(device) && btNrecIsOff; 1050 // collect all of the thread's session IDs 1051 KeyedVector<int, bool> ids = thread->sessionIds(); 1052 // suspend effects associated with those session IDs 1053 for (size_t j = 0; j < ids.size(); ++j) { 1054 int sessionId = ids.keyAt(j); 1055 thread->setEffectSuspended(FX_IID_AEC, 1056 suspend, 1057 sessionId); 1058 thread->setEffectSuspended(FX_IID_NS, 1059 suspend, 1060 sessionId); 1061 } 1062 } 1063 mBtNrecIsOff = btNrecIsOff; 1064 } 1065 } 1066 String8 screenState; 1067 if (param.get(String8(AudioParameter::keyScreenState), screenState) == NO_ERROR) { 1068 bool isOff = screenState == "off"; 1069 if (isOff != (AudioFlinger::mScreenState & 1)) { 1070 AudioFlinger::mScreenState = ((AudioFlinger::mScreenState & ~1) + 2) | isOff; 1071 } 1072 } 1073 return final_result; 1074 } 1075 1076 // hold a strong ref on thread in case closeOutput() or closeInput() is called 1077 // and the thread is exited once the lock is released 1078 sp<ThreadBase> thread; 1079 { 1080 Mutex::Autolock _l(mLock); 1081 thread = checkPlaybackThread_l(ioHandle); 1082 if (thread == 0) { 1083 thread = checkRecordThread_l(ioHandle); 1084 } else if (thread == primaryPlaybackThread_l()) { 1085 // indicate output device change to all input threads for pre processing 1086 AudioParameter param = AudioParameter(keyValuePairs); 1087 int value; 1088 if ((param.getInt(String8(AudioParameter::keyRouting), value) == NO_ERROR) && 1089 (value != 0)) { 1090 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1091 mRecordThreads.valueAt(i)->setParameters(keyValuePairs); 1092 } 1093 } 1094 } 1095 } 1096 if (thread != 0) { 1097 return thread->setParameters(keyValuePairs); 1098 } 1099 return BAD_VALUE; 1100} 1101 1102String8 AudioFlinger::getParameters(audio_io_handle_t ioHandle, const String8& keys) const 1103{ 1104 ALOGVV("getParameters() io %d, keys %s, calling pid %d", 1105 ioHandle, keys.string(), IPCThreadState::self()->getCallingPid()); 1106 1107 Mutex::Autolock _l(mLock); 1108 1109 if (ioHandle == AUDIO_IO_HANDLE_NONE) { 1110 String8 out_s8; 1111 1112 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1113 char *s; 1114 { 1115 AutoMutex lock(mHardwareLock); 1116 mHardwareStatus = AUDIO_HW_GET_PARAMETER; 1117 audio_hw_device_t *dev = mAudioHwDevs.valueAt(i)->hwDevice(); 1118 s = dev->get_parameters(dev, keys.string()); 1119 mHardwareStatus = AUDIO_HW_IDLE; 1120 } 1121 out_s8 += String8(s ? s : ""); 1122 free(s); 1123 } 1124 return out_s8; 1125 } 1126 1127 PlaybackThread *playbackThread = checkPlaybackThread_l(ioHandle); 1128 if (playbackThread != NULL) { 1129 return playbackThread->getParameters(keys); 1130 } 1131 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1132 if (recordThread != NULL) { 1133 return recordThread->getParameters(keys); 1134 } 1135 return String8(""); 1136} 1137 1138size_t AudioFlinger::getInputBufferSize(uint32_t sampleRate, audio_format_t format, 1139 audio_channel_mask_t channelMask) const 1140{ 1141 status_t ret = initCheck(); 1142 if (ret != NO_ERROR) { 1143 return 0; 1144 } 1145 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1146 return 0; 1147 } 1148 1149 AutoMutex lock(mHardwareLock); 1150 mHardwareStatus = AUDIO_HW_GET_INPUT_BUFFER_SIZE; 1151 audio_config_t config, proposed; 1152 memset(&proposed, 0, sizeof(proposed)); 1153 proposed.sample_rate = sampleRate; 1154 proposed.channel_mask = channelMask; 1155 proposed.format = format; 1156 1157 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1158 size_t frames; 1159 for (;;) { 1160 // Note: config is currently a const parameter for get_input_buffer_size() 1161 // but we use a copy from proposed in case config changes from the call. 1162 config = proposed; 1163 frames = dev->get_input_buffer_size(dev, &config); 1164 if (frames != 0) { 1165 break; // hal success, config is the result 1166 } 1167 // change one parameter of the configuration each iteration to a more "common" value 1168 // to see if the device will support it. 1169 if (proposed.format != AUDIO_FORMAT_PCM_16_BIT) { 1170 proposed.format = AUDIO_FORMAT_PCM_16_BIT; 1171 } else if (proposed.sample_rate != 44100) { // 44.1 is claimed as must in CDD as well as 1172 proposed.sample_rate = 44100; // legacy AudioRecord.java. TODO: Query hw? 1173 } else { 1174 ALOGW("getInputBufferSize failed with minimum buffer size sampleRate %u, " 1175 "format %#x, channelMask 0x%X", 1176 sampleRate, format, channelMask); 1177 break; // retries failed, break out of loop with frames == 0. 1178 } 1179 } 1180 mHardwareStatus = AUDIO_HW_IDLE; 1181 if (frames > 0 && config.sample_rate != sampleRate) { 1182 frames = destinationFramesPossible(frames, sampleRate, config.sample_rate); 1183 } 1184 return frames; // may be converted to bytes at the Java level. 1185} 1186 1187uint32_t AudioFlinger::getInputFramesLost(audio_io_handle_t ioHandle) const 1188{ 1189 Mutex::Autolock _l(mLock); 1190 1191 RecordThread *recordThread = checkRecordThread_l(ioHandle); 1192 if (recordThread != NULL) { 1193 return recordThread->getInputFramesLost(); 1194 } 1195 return 0; 1196} 1197 1198status_t AudioFlinger::setVoiceVolume(float value) 1199{ 1200 status_t ret = initCheck(); 1201 if (ret != NO_ERROR) { 1202 return ret; 1203 } 1204 1205 // check calling permissions 1206 if (!settingsAllowed()) { 1207 return PERMISSION_DENIED; 1208 } 1209 1210 AutoMutex lock(mHardwareLock); 1211 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1212 mHardwareStatus = AUDIO_HW_SET_VOICE_VOLUME; 1213 ret = dev->set_voice_volume(dev, value); 1214 mHardwareStatus = AUDIO_HW_IDLE; 1215 1216 return ret; 1217} 1218 1219status_t AudioFlinger::getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 1220 audio_io_handle_t output) const 1221{ 1222 status_t status; 1223 1224 Mutex::Autolock _l(mLock); 1225 1226 PlaybackThread *playbackThread = checkPlaybackThread_l(output); 1227 if (playbackThread != NULL) { 1228 return playbackThread->getRenderPosition(halFrames, dspFrames); 1229 } 1230 1231 return BAD_VALUE; 1232} 1233 1234void AudioFlinger::registerClient(const sp<IAudioFlingerClient>& client) 1235{ 1236 Mutex::Autolock _l(mLock); 1237 if (client == 0) { 1238 return; 1239 } 1240 bool clientAdded = false; 1241 { 1242 Mutex::Autolock _cl(mClientLock); 1243 1244 pid_t pid = IPCThreadState::self()->getCallingPid(); 1245 if (mNotificationClients.indexOfKey(pid) < 0) { 1246 sp<NotificationClient> notificationClient = new NotificationClient(this, 1247 client, 1248 pid); 1249 ALOGV("registerClient() client %p, pid %d", notificationClient.get(), pid); 1250 1251 mNotificationClients.add(pid, notificationClient); 1252 1253 sp<IBinder> binder = IInterface::asBinder(client); 1254 binder->linkToDeath(notificationClient); 1255 clientAdded = true; 1256 } 1257 } 1258 1259 // mClientLock should not be held here because ThreadBase::sendIoConfigEvent() will lock the 1260 // ThreadBase mutex and the locking order is ThreadBase::mLock then AudioFlinger::mClientLock. 1261 if (clientAdded) { 1262 // the config change is always sent from playback or record threads to avoid deadlock 1263 // with AudioSystem::gLock 1264 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1265 mPlaybackThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::OUTPUT_OPENED); 1266 } 1267 1268 for (size_t i = 0; i < mRecordThreads.size(); i++) { 1269 mRecordThreads.valueAt(i)->sendIoConfigEvent(AudioSystem::INPUT_OPENED); 1270 } 1271 } 1272} 1273 1274void AudioFlinger::removeNotificationClient(pid_t pid) 1275{ 1276 Mutex::Autolock _l(mLock); 1277 { 1278 Mutex::Autolock _cl(mClientLock); 1279 mNotificationClients.removeItem(pid); 1280 } 1281 1282 ALOGV("%d died, releasing its sessions", pid); 1283 size_t num = mAudioSessionRefs.size(); 1284 bool removed = false; 1285 for (size_t i = 0; i< num; ) { 1286 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 1287 ALOGV(" pid %d @ %d", ref->mPid, i); 1288 if (ref->mPid == pid) { 1289 ALOGV(" removing entry for pid %d session %d", pid, ref->mSessionid); 1290 mAudioSessionRefs.removeAt(i); 1291 delete ref; 1292 removed = true; 1293 num--; 1294 } else { 1295 i++; 1296 } 1297 } 1298 if (removed) { 1299 purgeStaleEffects_l(); 1300 } 1301} 1302 1303void AudioFlinger::audioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2) 1304{ 1305 Mutex::Autolock _l(mClientLock); 1306 size_t size = mNotificationClients.size(); 1307 for (size_t i = 0; i < size; i++) { 1308 mNotificationClients.valueAt(i)->audioFlingerClient()->ioConfigChanged(event, 1309 ioHandle, 1310 param2); 1311 } 1312} 1313 1314// removeClient_l() must be called with AudioFlinger::mClientLock held 1315void AudioFlinger::removeClient_l(pid_t pid) 1316{ 1317 ALOGV("removeClient_l() pid %d, calling pid %d", pid, 1318 IPCThreadState::self()->getCallingPid()); 1319 mClients.removeItem(pid); 1320} 1321 1322// getEffectThread_l() must be called with AudioFlinger::mLock held 1323sp<AudioFlinger::PlaybackThread> AudioFlinger::getEffectThread_l(int sessionId, int EffectId) 1324{ 1325 sp<PlaybackThread> thread; 1326 1327 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1328 if (mPlaybackThreads.valueAt(i)->getEffect(sessionId, EffectId) != 0) { 1329 ALOG_ASSERT(thread == 0); 1330 thread = mPlaybackThreads.valueAt(i); 1331 } 1332 } 1333 1334 return thread; 1335} 1336 1337 1338 1339// ---------------------------------------------------------------------------- 1340 1341AudioFlinger::Client::Client(const sp<AudioFlinger>& audioFlinger, pid_t pid) 1342 : RefBase(), 1343 mAudioFlinger(audioFlinger), 1344 // FIXME should be a "k" constant not hard-coded, in .h or ro. property, see 4 lines below 1345 mMemoryDealer(new MemoryDealer(1024*1024, "AudioFlinger::Client")), 1346 mPid(pid), 1347 mTimedTrackCount(0) 1348{ 1349 // 1 MB of address space is good for 32 tracks, 8 buffers each, 4 KB/buffer 1350} 1351 1352// Client destructor must be called with AudioFlinger::mClientLock held 1353AudioFlinger::Client::~Client() 1354{ 1355 mAudioFlinger->removeClient_l(mPid); 1356} 1357 1358sp<MemoryDealer> AudioFlinger::Client::heap() const 1359{ 1360 return mMemoryDealer; 1361} 1362 1363// Reserve one of the limited slots for a timed audio track associated 1364// with this client 1365bool AudioFlinger::Client::reserveTimedTrack() 1366{ 1367 const int kMaxTimedTracksPerClient = 4; 1368 1369 Mutex::Autolock _l(mTimedTrackLock); 1370 1371 if (mTimedTrackCount >= kMaxTimedTracksPerClient) { 1372 ALOGW("can not create timed track - pid %d has exceeded the limit", 1373 mPid); 1374 return false; 1375 } 1376 1377 mTimedTrackCount++; 1378 return true; 1379} 1380 1381// Release a slot for a timed audio track 1382void AudioFlinger::Client::releaseTimedTrack() 1383{ 1384 Mutex::Autolock _l(mTimedTrackLock); 1385 mTimedTrackCount--; 1386} 1387 1388// ---------------------------------------------------------------------------- 1389 1390AudioFlinger::NotificationClient::NotificationClient(const sp<AudioFlinger>& audioFlinger, 1391 const sp<IAudioFlingerClient>& client, 1392 pid_t pid) 1393 : mAudioFlinger(audioFlinger), mPid(pid), mAudioFlingerClient(client) 1394{ 1395} 1396 1397AudioFlinger::NotificationClient::~NotificationClient() 1398{ 1399} 1400 1401void AudioFlinger::NotificationClient::binderDied(const wp<IBinder>& who __unused) 1402{ 1403 sp<NotificationClient> keep(this); 1404 mAudioFlinger->removeNotificationClient(mPid); 1405} 1406 1407 1408// ---------------------------------------------------------------------------- 1409 1410static bool deviceRequiresCaptureAudioOutputPermission(audio_devices_t inDevice) { 1411 return audio_is_remote_submix_device(inDevice); 1412} 1413 1414sp<IAudioRecord> AudioFlinger::openRecord( 1415 audio_io_handle_t input, 1416 uint32_t sampleRate, 1417 audio_format_t format, 1418 audio_channel_mask_t channelMask, 1419 size_t *frameCount, 1420 IAudioFlinger::track_flags_t *flags, 1421 pid_t tid, 1422 int *sessionId, 1423 size_t *notificationFrames, 1424 sp<IMemory>& cblk, 1425 sp<IMemory>& buffers, 1426 status_t *status) 1427{ 1428 sp<RecordThread::RecordTrack> recordTrack; 1429 sp<RecordHandle> recordHandle; 1430 sp<Client> client; 1431 status_t lStatus; 1432 int lSessionId; 1433 1434 cblk.clear(); 1435 buffers.clear(); 1436 1437 // check calling permissions 1438 if (!recordingAllowed()) { 1439 ALOGE("openRecord() permission denied: recording not allowed"); 1440 lStatus = PERMISSION_DENIED; 1441 goto Exit; 1442 } 1443 1444 // further sample rate checks are performed by createRecordTrack_l() 1445 if (sampleRate == 0) { 1446 ALOGE("openRecord() invalid sample rate %u", sampleRate); 1447 lStatus = BAD_VALUE; 1448 goto Exit; 1449 } 1450 1451 // we don't yet support anything other than linear PCM 1452 if (!audio_is_valid_format(format) || !audio_is_linear_pcm(format)) { 1453 ALOGE("openRecord() invalid format %#x", format); 1454 lStatus = BAD_VALUE; 1455 goto Exit; 1456 } 1457 1458 // further channel mask checks are performed by createRecordTrack_l() 1459 if (!audio_is_input_channel(channelMask)) { 1460 ALOGE("openRecord() invalid channel mask %#x", channelMask); 1461 lStatus = BAD_VALUE; 1462 goto Exit; 1463 } 1464 1465 { 1466 Mutex::Autolock _l(mLock); 1467 RecordThread *thread = checkRecordThread_l(input); 1468 if (thread == NULL) { 1469 ALOGE("openRecord() checkRecordThread_l failed"); 1470 lStatus = BAD_VALUE; 1471 goto Exit; 1472 } 1473 1474 pid_t pid = IPCThreadState::self()->getCallingPid(); 1475 client = registerPid(pid); 1476 1477 if (sessionId != NULL && *sessionId != AUDIO_SESSION_ALLOCATE) { 1478 lSessionId = *sessionId; 1479 } else { 1480 // if no audio session id is provided, create one here 1481 lSessionId = nextUniqueId(); 1482 if (sessionId != NULL) { 1483 *sessionId = lSessionId; 1484 } 1485 } 1486 ALOGV("openRecord() lSessionId: %d input %d", lSessionId, input); 1487 1488 // TODO: the uid should be passed in as a parameter to openRecord 1489 recordTrack = thread->createRecordTrack_l(client, sampleRate, format, channelMask, 1490 frameCount, lSessionId, notificationFrames, 1491 IPCThreadState::self()->getCallingUid(), 1492 flags, tid, &lStatus); 1493 LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0)); 1494 1495 if (lStatus == NO_ERROR) { 1496 // Check if one effect chain was awaiting for an AudioRecord to be created on this 1497 // session and move it to this thread. 1498 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)lSessionId); 1499 if (chain != 0) { 1500 Mutex::Autolock _l(thread->mLock); 1501 thread->addEffectChain_l(chain); 1502 } 1503 } 1504 } 1505 1506 if (lStatus != NO_ERROR) { 1507 // remove local strong reference to Client before deleting the RecordTrack so that the 1508 // Client destructor is called by the TrackBase destructor with mClientLock held 1509 // Don't hold mClientLock when releasing the reference on the track as the 1510 // destructor will acquire it. 1511 { 1512 Mutex::Autolock _cl(mClientLock); 1513 client.clear(); 1514 } 1515 recordTrack.clear(); 1516 goto Exit; 1517 } 1518 1519 cblk = recordTrack->getCblk(); 1520 buffers = recordTrack->getBuffers(); 1521 1522 // return handle to client 1523 recordHandle = new RecordHandle(recordTrack); 1524 1525Exit: 1526 *status = lStatus; 1527 return recordHandle; 1528} 1529 1530 1531 1532// ---------------------------------------------------------------------------- 1533 1534audio_module_handle_t AudioFlinger::loadHwModule(const char *name) 1535{ 1536 if (name == NULL) { 1537 return 0; 1538 } 1539 if (!settingsAllowed()) { 1540 return 0; 1541 } 1542 Mutex::Autolock _l(mLock); 1543 return loadHwModule_l(name); 1544} 1545 1546// loadHwModule_l() must be called with AudioFlinger::mLock held 1547audio_module_handle_t AudioFlinger::loadHwModule_l(const char *name) 1548{ 1549 for (size_t i = 0; i < mAudioHwDevs.size(); i++) { 1550 if (strncmp(mAudioHwDevs.valueAt(i)->moduleName(), name, strlen(name)) == 0) { 1551 ALOGW("loadHwModule() module %s already loaded", name); 1552 return mAudioHwDevs.keyAt(i); 1553 } 1554 } 1555 1556 audio_hw_device_t *dev; 1557 1558 int rc = load_audio_interface(name, &dev); 1559 if (rc) { 1560 ALOGI("loadHwModule() error %d loading module %s ", rc, name); 1561 return 0; 1562 } 1563 1564 mHardwareStatus = AUDIO_HW_INIT; 1565 rc = dev->init_check(dev); 1566 mHardwareStatus = AUDIO_HW_IDLE; 1567 if (rc) { 1568 ALOGI("loadHwModule() init check error %d for module %s ", rc, name); 1569 return 0; 1570 } 1571 1572 // Check and cache this HAL's level of support for master mute and master 1573 // volume. If this is the first HAL opened, and it supports the get 1574 // methods, use the initial values provided by the HAL as the current 1575 // master mute and volume settings. 1576 1577 AudioHwDevice::Flags flags = static_cast<AudioHwDevice::Flags>(0); 1578 { // scope for auto-lock pattern 1579 AutoMutex lock(mHardwareLock); 1580 1581 if (0 == mAudioHwDevs.size()) { 1582 mHardwareStatus = AUDIO_HW_GET_MASTER_VOLUME; 1583 if (NULL != dev->get_master_volume) { 1584 float mv; 1585 if (OK == dev->get_master_volume(dev, &mv)) { 1586 mMasterVolume = mv; 1587 } 1588 } 1589 1590 mHardwareStatus = AUDIO_HW_GET_MASTER_MUTE; 1591 if (NULL != dev->get_master_mute) { 1592 bool mm; 1593 if (OK == dev->get_master_mute(dev, &mm)) { 1594 mMasterMute = mm; 1595 } 1596 } 1597 } 1598 1599 mHardwareStatus = AUDIO_HW_SET_MASTER_VOLUME; 1600 if ((NULL != dev->set_master_volume) && 1601 (OK == dev->set_master_volume(dev, mMasterVolume))) { 1602 flags = static_cast<AudioHwDevice::Flags>(flags | 1603 AudioHwDevice::AHWD_CAN_SET_MASTER_VOLUME); 1604 } 1605 1606 mHardwareStatus = AUDIO_HW_SET_MASTER_MUTE; 1607 if ((NULL != dev->set_master_mute) && 1608 (OK == dev->set_master_mute(dev, mMasterMute))) { 1609 flags = static_cast<AudioHwDevice::Flags>(flags | 1610 AudioHwDevice::AHWD_CAN_SET_MASTER_MUTE); 1611 } 1612 1613 mHardwareStatus = AUDIO_HW_IDLE; 1614 } 1615 1616 audio_module_handle_t handle = nextUniqueId(); 1617 mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags)); 1618 1619 ALOGI("loadHwModule() Loaded %s audio interface from %s (%s) handle %d", 1620 name, dev->common.module->name, dev->common.module->id, handle); 1621 1622 return handle; 1623 1624} 1625 1626// ---------------------------------------------------------------------------- 1627 1628uint32_t AudioFlinger::getPrimaryOutputSamplingRate() 1629{ 1630 Mutex::Autolock _l(mLock); 1631 PlaybackThread *thread = primaryPlaybackThread_l(); 1632 return thread != NULL ? thread->sampleRate() : 0; 1633} 1634 1635size_t AudioFlinger::getPrimaryOutputFrameCount() 1636{ 1637 Mutex::Autolock _l(mLock); 1638 PlaybackThread *thread = primaryPlaybackThread_l(); 1639 return thread != NULL ? thread->frameCountHAL() : 0; 1640} 1641 1642// ---------------------------------------------------------------------------- 1643 1644status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice) 1645{ 1646 uid_t uid = IPCThreadState::self()->getCallingUid(); 1647 if (uid != AID_SYSTEM) { 1648 return PERMISSION_DENIED; 1649 } 1650 Mutex::Autolock _l(mLock); 1651 if (mIsDeviceTypeKnown) { 1652 return INVALID_OPERATION; 1653 } 1654 mIsLowRamDevice = isLowRamDevice; 1655 mIsDeviceTypeKnown = true; 1656 return NO_ERROR; 1657} 1658 1659audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId) 1660{ 1661 Mutex::Autolock _l(mLock); 1662 1663 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1664 if (index >= 0) { 1665 ALOGV("getAudioHwSyncForSession found ID %d for session %d", 1666 mHwAvSyncIds.valueAt(index), sessionId); 1667 return mHwAvSyncIds.valueAt(index); 1668 } 1669 1670 audio_hw_device_t *dev = mPrimaryHardwareDev->hwDevice(); 1671 if (dev == NULL) { 1672 return AUDIO_HW_SYNC_INVALID; 1673 } 1674 char *reply = dev->get_parameters(dev, AUDIO_PARAMETER_HW_AV_SYNC); 1675 AudioParameter param = AudioParameter(String8(reply)); 1676 free(reply); 1677 1678 int value; 1679 if (param.getInt(String8(AUDIO_PARAMETER_HW_AV_SYNC), value) != NO_ERROR) { 1680 ALOGW("getAudioHwSyncForSession error getting sync for session %d", sessionId); 1681 return AUDIO_HW_SYNC_INVALID; 1682 } 1683 1684 // allow only one session for a given HW A/V sync ID. 1685 for (size_t i = 0; i < mHwAvSyncIds.size(); i++) { 1686 if (mHwAvSyncIds.valueAt(i) == (audio_hw_sync_t)value) { 1687 ALOGV("getAudioHwSyncForSession removing ID %d for session %d", 1688 value, mHwAvSyncIds.keyAt(i)); 1689 mHwAvSyncIds.removeItemsAt(i); 1690 break; 1691 } 1692 } 1693 1694 mHwAvSyncIds.add(sessionId, value); 1695 1696 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1697 sp<PlaybackThread> thread = mPlaybackThreads.valueAt(i); 1698 uint32_t sessions = thread->hasAudioSession(sessionId); 1699 if (sessions & PlaybackThread::TRACK_SESSION) { 1700 AudioParameter param = AudioParameter(); 1701 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), value); 1702 thread->setParameters(param.toString()); 1703 break; 1704 } 1705 } 1706 1707 ALOGV("getAudioHwSyncForSession adding ID %d for session %d", value, sessionId); 1708 return (audio_hw_sync_t)value; 1709} 1710 1711// setAudioHwSyncForSession_l() must be called with AudioFlinger::mLock held 1712void AudioFlinger::setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId) 1713{ 1714 ssize_t index = mHwAvSyncIds.indexOfKey(sessionId); 1715 if (index >= 0) { 1716 audio_hw_sync_t syncId = mHwAvSyncIds.valueAt(index); 1717 ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId); 1718 AudioParameter param = AudioParameter(); 1719 param.addInt(String8(AUDIO_PARAMETER_STREAM_HW_AV_SYNC), syncId); 1720 thread->setParameters(param.toString()); 1721 } 1722} 1723 1724 1725// ---------------------------------------------------------------------------- 1726 1727 1728sp<AudioFlinger::PlaybackThread> AudioFlinger::openOutput_l(audio_module_handle_t module, 1729 audio_io_handle_t *output, 1730 audio_config_t *config, 1731 audio_devices_t devices, 1732 const String8& address, 1733 audio_output_flags_t flags) 1734{ 1735 AudioHwDevice *outHwDev = findSuitableHwDev_l(module, devices); 1736 if (outHwDev == NULL) { 1737 return 0; 1738 } 1739 1740 audio_hw_device_t *hwDevHal = outHwDev->hwDevice(); 1741 if (*output == AUDIO_IO_HANDLE_NONE) { 1742 *output = nextUniqueId(); 1743 } 1744 1745 mHardwareStatus = AUDIO_HW_OUTPUT_OPEN; 1746 1747 // FOR TESTING ONLY: 1748 // This if statement allows overriding the audio policy settings 1749 // and forcing a specific format or channel mask to the HAL/Sink device for testing. 1750 if (!(flags & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD | AUDIO_OUTPUT_FLAG_DIRECT))) { 1751 // Check only for Normal Mixing mode 1752 if (kEnableExtendedPrecision) { 1753 // Specify format (uncomment one below to choose) 1754 //config->format = AUDIO_FORMAT_PCM_FLOAT; 1755 //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; 1756 //config->format = AUDIO_FORMAT_PCM_32_BIT; 1757 //config->format = AUDIO_FORMAT_PCM_8_24_BIT; 1758 // ALOGV("openOutput_l() upgrading format to %#08x", config->format); 1759 } 1760 if (kEnableExtendedChannels) { 1761 // Specify channel mask (uncomment one below to choose) 1762 //config->channel_mask = audio_channel_out_mask_from_count(4); // for USB 4ch 1763 //config->channel_mask = audio_channel_mask_from_representation_and_bits( 1764 // AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1); // another 4ch example 1765 } 1766 } 1767 1768 AudioStreamOut *outputStream = NULL; 1769 status_t status = outHwDev->openOutputStream( 1770 &outputStream, 1771 *output, 1772 devices, 1773 flags, 1774 config, 1775 address.string()); 1776 1777 mHardwareStatus = AUDIO_HW_IDLE; 1778 1779 if (status == NO_ERROR) { 1780 1781 PlaybackThread *thread; 1782 if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) { 1783 thread = new OffloadThread(this, outputStream, *output, devices); 1784 ALOGV("openOutput_l() created offload output: ID %d thread %p", *output, thread); 1785 } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT) 1786 || !isValidPcmSinkFormat(config->format) 1787 || !isValidPcmSinkChannelMask(config->channel_mask)) { 1788 thread = new DirectOutputThread(this, outputStream, *output, devices); 1789 ALOGV("openOutput_l() created direct output: ID %d thread %p", *output, thread); 1790 } else { 1791 thread = new MixerThread(this, outputStream, *output, devices); 1792 ALOGV("openOutput_l() created mixer output: ID %d thread %p", *output, thread); 1793 } 1794 mPlaybackThreads.add(*output, thread); 1795 return thread; 1796 } 1797 1798 return 0; 1799} 1800 1801status_t AudioFlinger::openOutput(audio_module_handle_t module, 1802 audio_io_handle_t *output, 1803 audio_config_t *config, 1804 audio_devices_t *devices, 1805 const String8& address, 1806 uint32_t *latencyMs, 1807 audio_output_flags_t flags) 1808{ 1809 ALOGI("openOutput(), module %d Device %x, SamplingRate %d, Format %#08x, Channels %x, flags %x", 1810 module, 1811 (devices != NULL) ? *devices : 0, 1812 config->sample_rate, 1813 config->format, 1814 config->channel_mask, 1815 flags); 1816 1817 if (*devices == AUDIO_DEVICE_NONE) { 1818 return BAD_VALUE; 1819 } 1820 1821 Mutex::Autolock _l(mLock); 1822 1823 sp<PlaybackThread> thread = openOutput_l(module, output, config, *devices, address, flags); 1824 if (thread != 0) { 1825 *latencyMs = thread->latency(); 1826 1827 // notify client processes of the new output creation 1828 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1829 1830 // the first primary output opened designates the primary hw device 1831 if ((mPrimaryHardwareDev == NULL) && (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { 1832 ALOGI("Using module %d has the primary audio interface", module); 1833 mPrimaryHardwareDev = thread->getOutput()->audioHwDev; 1834 1835 AutoMutex lock(mHardwareLock); 1836 mHardwareStatus = AUDIO_HW_SET_MODE; 1837 mPrimaryHardwareDev->hwDevice()->set_mode(mPrimaryHardwareDev->hwDevice(), mMode); 1838 mHardwareStatus = AUDIO_HW_IDLE; 1839 1840 mPrimaryOutputSampleRate = config->sample_rate; 1841 } 1842 return NO_ERROR; 1843 } 1844 1845 return NO_INIT; 1846} 1847 1848audio_io_handle_t AudioFlinger::openDuplicateOutput(audio_io_handle_t output1, 1849 audio_io_handle_t output2) 1850{ 1851 Mutex::Autolock _l(mLock); 1852 MixerThread *thread1 = checkMixerThread_l(output1); 1853 MixerThread *thread2 = checkMixerThread_l(output2); 1854 1855 if (thread1 == NULL || thread2 == NULL) { 1856 ALOGW("openDuplicateOutput() wrong output mixer type for output %d or %d", output1, 1857 output2); 1858 return AUDIO_IO_HANDLE_NONE; 1859 } 1860 1861 audio_io_handle_t id = nextUniqueId(); 1862 DuplicatingThread *thread = new DuplicatingThread(this, thread1, id); 1863 thread->addOutputTrack(thread2); 1864 mPlaybackThreads.add(id, thread); 1865 // notify client processes of the new output creation 1866 thread->audioConfigChanged(AudioSystem::OUTPUT_OPENED); 1867 return id; 1868} 1869 1870status_t AudioFlinger::closeOutput(audio_io_handle_t output) 1871{ 1872 return closeOutput_nonvirtual(output); 1873} 1874 1875status_t AudioFlinger::closeOutput_nonvirtual(audio_io_handle_t output) 1876{ 1877 // keep strong reference on the playback thread so that 1878 // it is not destroyed while exit() is executed 1879 sp<PlaybackThread> thread; 1880 { 1881 Mutex::Autolock _l(mLock); 1882 thread = checkPlaybackThread_l(output); 1883 if (thread == NULL) { 1884 return BAD_VALUE; 1885 } 1886 1887 ALOGV("closeOutput() %d", output); 1888 1889 if (thread->type() == ThreadBase::MIXER) { 1890 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 1891 if (mPlaybackThreads.valueAt(i)->type() == ThreadBase::DUPLICATING) { 1892 DuplicatingThread *dupThread = 1893 (DuplicatingThread *)mPlaybackThreads.valueAt(i).get(); 1894 dupThread->removeOutputTrack((MixerThread *)thread.get()); 1895 1896 } 1897 } 1898 } 1899 1900 1901 mPlaybackThreads.removeItem(output); 1902 // save all effects to the default thread 1903 if (mPlaybackThreads.size()) { 1904 PlaybackThread *dstThread = checkPlaybackThread_l(mPlaybackThreads.keyAt(0)); 1905 if (dstThread != NULL) { 1906 // audioflinger lock is held here so the acquisition order of thread locks does not 1907 // matter 1908 Mutex::Autolock _dl(dstThread->mLock); 1909 Mutex::Autolock _sl(thread->mLock); 1910 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 1911 for (size_t i = 0; i < effectChains.size(); i ++) { 1912 moveEffectChain_l(effectChains[i]->sessionId(), thread.get(), dstThread, true); 1913 } 1914 } 1915 } 1916 audioConfigChanged(AudioSystem::OUTPUT_CLOSED, output, NULL); 1917 } 1918 thread->exit(); 1919 // The thread entity (active unit of execution) is no longer running here, 1920 // but the ThreadBase container still exists. 1921 1922 if (thread->type() != ThreadBase::DUPLICATING) { 1923 closeOutputFinish(thread); 1924 } 1925 1926 return NO_ERROR; 1927} 1928 1929void AudioFlinger::closeOutputFinish(sp<PlaybackThread> thread) 1930{ 1931 AudioStreamOut *out = thread->clearOutput(); 1932 ALOG_ASSERT(out != NULL, "out shouldn't be NULL"); 1933 // from now on thread->mOutput is NULL 1934 out->hwDev()->close_output_stream(out->hwDev(), out->stream); 1935 delete out; 1936} 1937 1938void AudioFlinger::closeOutputInternal_l(sp<PlaybackThread> thread) 1939{ 1940 mPlaybackThreads.removeItem(thread->mId); 1941 thread->exit(); 1942 closeOutputFinish(thread); 1943} 1944 1945status_t AudioFlinger::suspendOutput(audio_io_handle_t output) 1946{ 1947 Mutex::Autolock _l(mLock); 1948 PlaybackThread *thread = checkPlaybackThread_l(output); 1949 1950 if (thread == NULL) { 1951 return BAD_VALUE; 1952 } 1953 1954 ALOGV("suspendOutput() %d", output); 1955 thread->suspend(); 1956 1957 return NO_ERROR; 1958} 1959 1960status_t AudioFlinger::restoreOutput(audio_io_handle_t output) 1961{ 1962 Mutex::Autolock _l(mLock); 1963 PlaybackThread *thread = checkPlaybackThread_l(output); 1964 1965 if (thread == NULL) { 1966 return BAD_VALUE; 1967 } 1968 1969 ALOGV("restoreOutput() %d", output); 1970 1971 thread->restore(); 1972 1973 return NO_ERROR; 1974} 1975 1976status_t AudioFlinger::openInput(audio_module_handle_t module, 1977 audio_io_handle_t *input, 1978 audio_config_t *config, 1979 audio_devices_t *devices, 1980 const String8& address, 1981 audio_source_t source, 1982 audio_input_flags_t flags) 1983{ 1984 Mutex::Autolock _l(mLock); 1985 1986 if (*devices == AUDIO_DEVICE_NONE) { 1987 return BAD_VALUE; 1988 } 1989 1990 sp<RecordThread> thread = openInput_l(module, input, config, *devices, address, source, flags); 1991 1992 if (thread != 0) { 1993 // notify client processes of the new input creation 1994 thread->audioConfigChanged(AudioSystem::INPUT_OPENED); 1995 return NO_ERROR; 1996 } 1997 return NO_INIT; 1998} 1999 2000sp<AudioFlinger::RecordThread> AudioFlinger::openInput_l(audio_module_handle_t module, 2001 audio_io_handle_t *input, 2002 audio_config_t *config, 2003 audio_devices_t devices, 2004 const String8& address, 2005 audio_source_t source, 2006 audio_input_flags_t flags) 2007{ 2008 AudioHwDevice *inHwDev = findSuitableHwDev_l(module, devices); 2009 if (inHwDev == NULL) { 2010 *input = AUDIO_IO_HANDLE_NONE; 2011 return 0; 2012 } 2013 2014 if (*input == AUDIO_IO_HANDLE_NONE) { 2015 *input = nextUniqueId(); 2016 } 2017 2018 audio_config_t halconfig = *config; 2019 audio_hw_device_t *inHwHal = inHwDev->hwDevice(); 2020 audio_stream_in_t *inStream = NULL; 2021 status_t status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2022 &inStream, flags, address.string(), source); 2023 ALOGV("openInput_l() openInputStream returned input %p, SamplingRate %d" 2024 ", Format %#x, Channels %x, flags %#x, status %d addr %s", 2025 inStream, 2026 halconfig.sample_rate, 2027 halconfig.format, 2028 halconfig.channel_mask, 2029 flags, 2030 status, address.string()); 2031 2032 // If the input could not be opened with the requested parameters and we can handle the 2033 // conversion internally, try to open again with the proposed parameters. 2034 if (status == BAD_VALUE && 2035 audio_is_linear_pcm(config->format) && 2036 audio_is_linear_pcm(halconfig.format) && 2037 (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) && 2038 (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_2) && 2039 (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_2)) { 2040 // FIXME describe the change proposed by HAL (save old values so we can log them here) 2041 ALOGV("openInput_l() reopening with proposed sampling rate and channel mask"); 2042 inStream = NULL; 2043 status = inHwHal->open_input_stream(inHwHal, *input, devices, &halconfig, 2044 &inStream, flags, address.string(), source); 2045 // FIXME log this new status; HAL should not propose any further changes 2046 } 2047 2048 if (status == NO_ERROR && inStream != NULL) { 2049 2050#ifdef TEE_SINK 2051 // Try to re-use most recently used Pipe to archive a copy of input for dumpsys, 2052 // or (re-)create if current Pipe is idle and does not match the new format 2053 sp<NBAIO_Sink> teeSink; 2054 enum { 2055 TEE_SINK_NO, // don't copy input 2056 TEE_SINK_NEW, // copy input using a new pipe 2057 TEE_SINK_OLD, // copy input using an existing pipe 2058 } kind; 2059 NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate, 2060 audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format); 2061 if (!mTeeSinkInputEnabled) { 2062 kind = TEE_SINK_NO; 2063 } else if (!Format_isValid(format)) { 2064 kind = TEE_SINK_NO; 2065 } else if (mRecordTeeSink == 0) { 2066 kind = TEE_SINK_NEW; 2067 } else if (mRecordTeeSink->getStrongCount() != 1) { 2068 kind = TEE_SINK_NO; 2069 } else if (Format_isEqual(format, mRecordTeeSink->format())) { 2070 kind = TEE_SINK_OLD; 2071 } else { 2072 kind = TEE_SINK_NEW; 2073 } 2074 switch (kind) { 2075 case TEE_SINK_NEW: { 2076 Pipe *pipe = new Pipe(mTeeSinkInputFrames, format); 2077 size_t numCounterOffers = 0; 2078 const NBAIO_Format offers[1] = {format}; 2079 ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers); 2080 ALOG_ASSERT(index == 0); 2081 PipeReader *pipeReader = new PipeReader(*pipe); 2082 numCounterOffers = 0; 2083 index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers); 2084 ALOG_ASSERT(index == 0); 2085 mRecordTeeSink = pipe; 2086 mRecordTeeSource = pipeReader; 2087 teeSink = pipe; 2088 } 2089 break; 2090 case TEE_SINK_OLD: 2091 teeSink = mRecordTeeSink; 2092 break; 2093 case TEE_SINK_NO: 2094 default: 2095 break; 2096 } 2097#endif 2098 2099 AudioStreamIn *inputStream = new AudioStreamIn(inHwDev, inStream); 2100 2101 // Start record thread 2102 // RecordThread requires both input and output device indication to forward to audio 2103 // pre processing modules 2104 sp<RecordThread> thread = new RecordThread(this, 2105 inputStream, 2106 *input, 2107 primaryOutputDevice_l(), 2108 devices 2109#ifdef TEE_SINK 2110 , teeSink 2111#endif 2112 ); 2113 mRecordThreads.add(*input, thread); 2114 ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get()); 2115 return thread; 2116 } 2117 2118 *input = AUDIO_IO_HANDLE_NONE; 2119 return 0; 2120} 2121 2122status_t AudioFlinger::closeInput(audio_io_handle_t input) 2123{ 2124 return closeInput_nonvirtual(input); 2125} 2126 2127status_t AudioFlinger::closeInput_nonvirtual(audio_io_handle_t input) 2128{ 2129 // keep strong reference on the record thread so that 2130 // it is not destroyed while exit() is executed 2131 sp<RecordThread> thread; 2132 { 2133 Mutex::Autolock _l(mLock); 2134 thread = checkRecordThread_l(input); 2135 if (thread == 0) { 2136 return BAD_VALUE; 2137 } 2138 2139 ALOGV("closeInput() %d", input); 2140 2141 // If we still have effect chains, it means that a client still holds a handle 2142 // on at least one effect. We must either move the chain to an existing thread with the 2143 // same session ID or put it aside in case a new record thread is opened for a 2144 // new capture on the same session 2145 sp<EffectChain> chain; 2146 { 2147 Mutex::Autolock _sl(thread->mLock); 2148 Vector< sp<EffectChain> > effectChains = thread->getEffectChains_l(); 2149 // Note: maximum one chain per record thread 2150 if (effectChains.size() != 0) { 2151 chain = effectChains[0]; 2152 } 2153 } 2154 if (chain != 0) { 2155 // first check if a record thread is already opened with a client on the same session. 2156 // This should only happen in case of overlap between one thread tear down and the 2157 // creation of its replacement 2158 size_t i; 2159 for (i = 0; i < mRecordThreads.size(); i++) { 2160 sp<RecordThread> t = mRecordThreads.valueAt(i); 2161 if (t == thread) { 2162 continue; 2163 } 2164 if (t->hasAudioSession(chain->sessionId()) != 0) { 2165 Mutex::Autolock _l(t->mLock); 2166 ALOGV("closeInput() found thread %d for effect session %d", 2167 t->id(), chain->sessionId()); 2168 t->addEffectChain_l(chain); 2169 break; 2170 } 2171 } 2172 // put the chain aside if we could not find a record thread with the same session id. 2173 if (i == mRecordThreads.size()) { 2174 putOrphanEffectChain_l(chain); 2175 } 2176 } 2177 audioConfigChanged(AudioSystem::INPUT_CLOSED, input, NULL); 2178 mRecordThreads.removeItem(input); 2179 } 2180 // FIXME: calling thread->exit() without mLock held should not be needed anymore now that 2181 // we have a different lock for notification client 2182 closeInputFinish(thread); 2183 return NO_ERROR; 2184} 2185 2186void AudioFlinger::closeInputFinish(sp<RecordThread> thread) 2187{ 2188 thread->exit(); 2189 AudioStreamIn *in = thread->clearInput(); 2190 ALOG_ASSERT(in != NULL, "in shouldn't be NULL"); 2191 // from now on thread->mInput is NULL 2192 in->hwDev()->close_input_stream(in->hwDev(), in->stream); 2193 delete in; 2194} 2195 2196void AudioFlinger::closeInputInternal_l(sp<RecordThread> thread) 2197{ 2198 mRecordThreads.removeItem(thread->mId); 2199 closeInputFinish(thread); 2200} 2201 2202status_t AudioFlinger::invalidateStream(audio_stream_type_t stream) 2203{ 2204 Mutex::Autolock _l(mLock); 2205 ALOGV("invalidateStream() stream %d", stream); 2206 2207 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2208 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2209 thread->invalidateTracks(stream); 2210 } 2211 2212 return NO_ERROR; 2213} 2214 2215 2216audio_unique_id_t AudioFlinger::newAudioUniqueId() 2217{ 2218 return nextUniqueId(); 2219} 2220 2221void AudioFlinger::acquireAudioSessionId(int audioSession, pid_t pid) 2222{ 2223 Mutex::Autolock _l(mLock); 2224 pid_t caller = IPCThreadState::self()->getCallingPid(); 2225 ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid); 2226 if (pid != -1 && (caller == getpid_cached)) { 2227 caller = pid; 2228 } 2229 2230 { 2231 Mutex::Autolock _cl(mClientLock); 2232 // Ignore requests received from processes not known as notification client. The request 2233 // is likely proxied by mediaserver (e.g CameraService) and releaseAudioSessionId() can be 2234 // called from a different pid leaving a stale session reference. Also we don't know how 2235 // to clear this reference if the client process dies. 2236 if (mNotificationClients.indexOfKey(caller) < 0) { 2237 ALOGW("acquireAudioSessionId() unknown client %d for session %d", caller, audioSession); 2238 return; 2239 } 2240 } 2241 2242 size_t num = mAudioSessionRefs.size(); 2243 for (size_t i = 0; i< num; i++) { 2244 AudioSessionRef *ref = mAudioSessionRefs.editItemAt(i); 2245 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2246 ref->mCnt++; 2247 ALOGV(" incremented refcount to %d", ref->mCnt); 2248 return; 2249 } 2250 } 2251 mAudioSessionRefs.push(new AudioSessionRef(audioSession, caller)); 2252 ALOGV(" added new entry for %d", audioSession); 2253} 2254 2255void AudioFlinger::releaseAudioSessionId(int audioSession, pid_t pid) 2256{ 2257 Mutex::Autolock _l(mLock); 2258 pid_t caller = IPCThreadState::self()->getCallingPid(); 2259 ALOGV("releasing %d from %d for %d", audioSession, caller, pid); 2260 if (pid != -1 && (caller == getpid_cached)) { 2261 caller = pid; 2262 } 2263 size_t num = mAudioSessionRefs.size(); 2264 for (size_t i = 0; i< num; i++) { 2265 AudioSessionRef *ref = mAudioSessionRefs.itemAt(i); 2266 if (ref->mSessionid == audioSession && ref->mPid == caller) { 2267 ref->mCnt--; 2268 ALOGV(" decremented refcount to %d", ref->mCnt); 2269 if (ref->mCnt == 0) { 2270 mAudioSessionRefs.removeAt(i); 2271 delete ref; 2272 purgeStaleEffects_l(); 2273 } 2274 return; 2275 } 2276 } 2277 // If the caller is mediaserver it is likely that the session being released was acquired 2278 // on behalf of a process not in notification clients and we ignore the warning. 2279 ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller); 2280} 2281 2282void AudioFlinger::purgeStaleEffects_l() { 2283 2284 ALOGV("purging stale effects"); 2285 2286 Vector< sp<EffectChain> > chains; 2287 2288 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2289 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2290 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2291 sp<EffectChain> ec = t->mEffectChains[j]; 2292 if (ec->sessionId() > AUDIO_SESSION_OUTPUT_MIX) { 2293 chains.push(ec); 2294 } 2295 } 2296 } 2297 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2298 sp<RecordThread> t = mRecordThreads.valueAt(i); 2299 for (size_t j = 0; j < t->mEffectChains.size(); j++) { 2300 sp<EffectChain> ec = t->mEffectChains[j]; 2301 chains.push(ec); 2302 } 2303 } 2304 2305 for (size_t i = 0; i < chains.size(); i++) { 2306 sp<EffectChain> ec = chains[i]; 2307 int sessionid = ec->sessionId(); 2308 sp<ThreadBase> t = ec->mThread.promote(); 2309 if (t == 0) { 2310 continue; 2311 } 2312 size_t numsessionrefs = mAudioSessionRefs.size(); 2313 bool found = false; 2314 for (size_t k = 0; k < numsessionrefs; k++) { 2315 AudioSessionRef *ref = mAudioSessionRefs.itemAt(k); 2316 if (ref->mSessionid == sessionid) { 2317 ALOGV(" session %d still exists for %d with %d refs", 2318 sessionid, ref->mPid, ref->mCnt); 2319 found = true; 2320 break; 2321 } 2322 } 2323 if (!found) { 2324 Mutex::Autolock _l(t->mLock); 2325 // remove all effects from the chain 2326 while (ec->mEffects.size()) { 2327 sp<EffectModule> effect = ec->mEffects[0]; 2328 effect->unPin(); 2329 t->removeEffect_l(effect); 2330 if (effect->purgeHandles()) { 2331 t->checkSuspendOnEffectEnabled_l(effect, false, effect->sessionId()); 2332 } 2333 AudioSystem::unregisterEffect(effect->id()); 2334 } 2335 } 2336 } 2337 return; 2338} 2339 2340// checkPlaybackThread_l() must be called with AudioFlinger::mLock held 2341AudioFlinger::PlaybackThread *AudioFlinger::checkPlaybackThread_l(audio_io_handle_t output) const 2342{ 2343 return mPlaybackThreads.valueFor(output).get(); 2344} 2345 2346// checkMixerThread_l() must be called with AudioFlinger::mLock held 2347AudioFlinger::MixerThread *AudioFlinger::checkMixerThread_l(audio_io_handle_t output) const 2348{ 2349 PlaybackThread *thread = checkPlaybackThread_l(output); 2350 return thread != NULL && thread->type() != ThreadBase::DIRECT ? (MixerThread *) thread : NULL; 2351} 2352 2353// checkRecordThread_l() must be called with AudioFlinger::mLock held 2354AudioFlinger::RecordThread *AudioFlinger::checkRecordThread_l(audio_io_handle_t input) const 2355{ 2356 return mRecordThreads.valueFor(input).get(); 2357} 2358 2359uint32_t AudioFlinger::nextUniqueId() 2360{ 2361 return (uint32_t) android_atomic_inc(&mNextUniqueId); 2362} 2363 2364AudioFlinger::PlaybackThread *AudioFlinger::primaryPlaybackThread_l() const 2365{ 2366 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2367 PlaybackThread *thread = mPlaybackThreads.valueAt(i).get(); 2368 AudioStreamOut *output = thread->getOutput(); 2369 if (output != NULL && output->audioHwDev == mPrimaryHardwareDev) { 2370 return thread; 2371 } 2372 } 2373 return NULL; 2374} 2375 2376audio_devices_t AudioFlinger::primaryOutputDevice_l() const 2377{ 2378 PlaybackThread *thread = primaryPlaybackThread_l(); 2379 2380 if (thread == NULL) { 2381 return 0; 2382 } 2383 2384 return thread->outDevice(); 2385} 2386 2387sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type, 2388 int triggerSession, 2389 int listenerSession, 2390 sync_event_callback_t callBack, 2391 wp<RefBase> cookie) 2392{ 2393 Mutex::Autolock _l(mLock); 2394 2395 sp<SyncEvent> event = new SyncEvent(type, triggerSession, listenerSession, callBack, cookie); 2396 status_t playStatus = NAME_NOT_FOUND; 2397 status_t recStatus = NAME_NOT_FOUND; 2398 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2399 playStatus = mPlaybackThreads.valueAt(i)->setSyncEvent(event); 2400 if (playStatus == NO_ERROR) { 2401 return event; 2402 } 2403 } 2404 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2405 recStatus = mRecordThreads.valueAt(i)->setSyncEvent(event); 2406 if (recStatus == NO_ERROR) { 2407 return event; 2408 } 2409 } 2410 if (playStatus == NAME_NOT_FOUND || recStatus == NAME_NOT_FOUND) { 2411 mPendingSyncEvents.add(event); 2412 } else { 2413 ALOGV("createSyncEvent() invalid event %d", event->type()); 2414 event.clear(); 2415 } 2416 return event; 2417} 2418 2419// ---------------------------------------------------------------------------- 2420// Effect management 2421// ---------------------------------------------------------------------------- 2422 2423 2424status_t AudioFlinger::queryNumberEffects(uint32_t *numEffects) const 2425{ 2426 Mutex::Autolock _l(mLock); 2427 return EffectQueryNumberEffects(numEffects); 2428} 2429 2430status_t AudioFlinger::queryEffect(uint32_t index, effect_descriptor_t *descriptor) const 2431{ 2432 Mutex::Autolock _l(mLock); 2433 return EffectQueryEffect(index, descriptor); 2434} 2435 2436status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid, 2437 effect_descriptor_t *descriptor) const 2438{ 2439 Mutex::Autolock _l(mLock); 2440 return EffectGetDescriptor(pUuid, descriptor); 2441} 2442 2443 2444sp<IEffect> AudioFlinger::createEffect( 2445 effect_descriptor_t *pDesc, 2446 const sp<IEffectClient>& effectClient, 2447 int32_t priority, 2448 audio_io_handle_t io, 2449 int sessionId, 2450 status_t *status, 2451 int *id, 2452 int *enabled) 2453{ 2454 status_t lStatus = NO_ERROR; 2455 sp<EffectHandle> handle; 2456 effect_descriptor_t desc; 2457 2458 pid_t pid = IPCThreadState::self()->getCallingPid(); 2459 ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d", 2460 pid, effectClient.get(), priority, sessionId, io); 2461 2462 if (pDesc == NULL) { 2463 lStatus = BAD_VALUE; 2464 goto Exit; 2465 } 2466 2467 // check audio settings permission for global effects 2468 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && !settingsAllowed()) { 2469 lStatus = PERMISSION_DENIED; 2470 goto Exit; 2471 } 2472 2473 // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects 2474 // that can only be created by audio policy manager (running in same process) 2475 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) { 2476 lStatus = PERMISSION_DENIED; 2477 goto Exit; 2478 } 2479 2480 { 2481 if (!EffectIsNullUuid(&pDesc->uuid)) { 2482 // if uuid is specified, request effect descriptor 2483 lStatus = EffectGetDescriptor(&pDesc->uuid, &desc); 2484 if (lStatus < 0) { 2485 ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus); 2486 goto Exit; 2487 } 2488 } else { 2489 // if uuid is not specified, look for an available implementation 2490 // of the required type in effect factory 2491 if (EffectIsNullUuid(&pDesc->type)) { 2492 ALOGW("createEffect() no effect type"); 2493 lStatus = BAD_VALUE; 2494 goto Exit; 2495 } 2496 uint32_t numEffects = 0; 2497 effect_descriptor_t d; 2498 d.flags = 0; // prevent compiler warning 2499 bool found = false; 2500 2501 lStatus = EffectQueryNumberEffects(&numEffects); 2502 if (lStatus < 0) { 2503 ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus); 2504 goto Exit; 2505 } 2506 for (uint32_t i = 0; i < numEffects; i++) { 2507 lStatus = EffectQueryEffect(i, &desc); 2508 if (lStatus < 0) { 2509 ALOGW("createEffect() error %d from EffectQueryEffect", lStatus); 2510 continue; 2511 } 2512 if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) { 2513 // If matching type found save effect descriptor. If the session is 2514 // 0 and the effect is not auxiliary, continue enumeration in case 2515 // an auxiliary version of this effect type is available 2516 found = true; 2517 d = desc; 2518 if (sessionId != AUDIO_SESSION_OUTPUT_MIX || 2519 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2520 break; 2521 } 2522 } 2523 } 2524 if (!found) { 2525 lStatus = BAD_VALUE; 2526 ALOGW("createEffect() effect not found"); 2527 goto Exit; 2528 } 2529 // For same effect type, chose auxiliary version over insert version if 2530 // connect to output mix (Compliance to OpenSL ES) 2531 if (sessionId == AUDIO_SESSION_OUTPUT_MIX && 2532 (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) { 2533 desc = d; 2534 } 2535 } 2536 2537 // Do not allow auxiliary effects on a session different from 0 (output mix) 2538 if (sessionId != AUDIO_SESSION_OUTPUT_MIX && 2539 (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) { 2540 lStatus = INVALID_OPERATION; 2541 goto Exit; 2542 } 2543 2544 // check recording permission for visualizer 2545 if ((memcmp(&desc.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) && 2546 !recordingAllowed()) { 2547 lStatus = PERMISSION_DENIED; 2548 goto Exit; 2549 } 2550 2551 // return effect descriptor 2552 *pDesc = desc; 2553 if (io == AUDIO_IO_HANDLE_NONE && sessionId == AUDIO_SESSION_OUTPUT_MIX) { 2554 // if the output returned by getOutputForEffect() is removed before we lock the 2555 // mutex below, the call to checkPlaybackThread_l(io) below will detect it 2556 // and we will exit safely 2557 io = AudioSystem::getOutputForEffect(&desc); 2558 ALOGV("createEffect got output %d", io); 2559 } 2560 2561 Mutex::Autolock _l(mLock); 2562 2563 // If output is not specified try to find a matching audio session ID in one of the 2564 // output threads. 2565 // If output is 0 here, sessionId is neither SESSION_OUTPUT_STAGE nor SESSION_OUTPUT_MIX 2566 // because of code checking output when entering the function. 2567 // Note: io is never 0 when creating an effect on an input 2568 if (io == AUDIO_IO_HANDLE_NONE) { 2569 if (sessionId == AUDIO_SESSION_OUTPUT_STAGE) { 2570 // output must be specified by AudioPolicyManager when using session 2571 // AUDIO_SESSION_OUTPUT_STAGE 2572 lStatus = BAD_VALUE; 2573 goto Exit; 2574 } 2575 // look for the thread where the specified audio session is present 2576 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2577 if (mPlaybackThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2578 io = mPlaybackThreads.keyAt(i); 2579 break; 2580 } 2581 } 2582 if (io == 0) { 2583 for (size_t i = 0; i < mRecordThreads.size(); i++) { 2584 if (mRecordThreads.valueAt(i)->hasAudioSession(sessionId) != 0) { 2585 io = mRecordThreads.keyAt(i); 2586 break; 2587 } 2588 } 2589 } 2590 // If no output thread contains the requested session ID, default to 2591 // first output. The effect chain will be moved to the correct output 2592 // thread when a track with the same session ID is created 2593 if (io == AUDIO_IO_HANDLE_NONE && mPlaybackThreads.size() > 0) { 2594 io = mPlaybackThreads.keyAt(0); 2595 } 2596 ALOGV("createEffect() got io %d for effect %s", io, desc.name); 2597 } 2598 ThreadBase *thread = checkRecordThread_l(io); 2599 if (thread == NULL) { 2600 thread = checkPlaybackThread_l(io); 2601 if (thread == NULL) { 2602 ALOGE("createEffect() unknown output thread"); 2603 lStatus = BAD_VALUE; 2604 goto Exit; 2605 } 2606 } else { 2607 // Check if one effect chain was awaiting for an effect to be created on this 2608 // session and used it instead of creating a new one. 2609 sp<EffectChain> chain = getOrphanEffectChain_l((audio_session_t)sessionId); 2610 if (chain != 0) { 2611 Mutex::Autolock _l(thread->mLock); 2612 thread->addEffectChain_l(chain); 2613 } 2614 } 2615 2616 sp<Client> client = registerPid(pid); 2617 2618 // create effect on selected output thread 2619 handle = thread->createEffect_l(client, effectClient, priority, sessionId, 2620 &desc, enabled, &lStatus); 2621 if (handle != 0 && id != NULL) { 2622 *id = handle->id(); 2623 } 2624 if (handle == 0) { 2625 // remove local strong reference to Client with mClientLock held 2626 Mutex::Autolock _cl(mClientLock); 2627 client.clear(); 2628 } 2629 } 2630 2631Exit: 2632 *status = lStatus; 2633 return handle; 2634} 2635 2636status_t AudioFlinger::moveEffects(int sessionId, audio_io_handle_t srcOutput, 2637 audio_io_handle_t dstOutput) 2638{ 2639 ALOGV("moveEffects() session %d, srcOutput %d, dstOutput %d", 2640 sessionId, srcOutput, dstOutput); 2641 Mutex::Autolock _l(mLock); 2642 if (srcOutput == dstOutput) { 2643 ALOGW("moveEffects() same dst and src outputs %d", dstOutput); 2644 return NO_ERROR; 2645 } 2646 PlaybackThread *srcThread = checkPlaybackThread_l(srcOutput); 2647 if (srcThread == NULL) { 2648 ALOGW("moveEffects() bad srcOutput %d", srcOutput); 2649 return BAD_VALUE; 2650 } 2651 PlaybackThread *dstThread = checkPlaybackThread_l(dstOutput); 2652 if (dstThread == NULL) { 2653 ALOGW("moveEffects() bad dstOutput %d", dstOutput); 2654 return BAD_VALUE; 2655 } 2656 2657 Mutex::Autolock _dl(dstThread->mLock); 2658 Mutex::Autolock _sl(srcThread->mLock); 2659 return moveEffectChain_l(sessionId, srcThread, dstThread, false); 2660} 2661 2662// moveEffectChain_l must be called with both srcThread and dstThread mLocks held 2663status_t AudioFlinger::moveEffectChain_l(int sessionId, 2664 AudioFlinger::PlaybackThread *srcThread, 2665 AudioFlinger::PlaybackThread *dstThread, 2666 bool reRegister) 2667{ 2668 ALOGV("moveEffectChain_l() session %d from thread %p to thread %p", 2669 sessionId, srcThread, dstThread); 2670 2671 sp<EffectChain> chain = srcThread->getEffectChain_l(sessionId); 2672 if (chain == 0) { 2673 ALOGW("moveEffectChain_l() effect chain for session %d not on source thread %p", 2674 sessionId, srcThread); 2675 return INVALID_OPERATION; 2676 } 2677 2678 // Check whether the destination thread has a channel count of FCC_2, which is 2679 // currently required for (most) effects. Prevent moving the effect chain here rather 2680 // than disabling the addEffect_l() call in dstThread below. 2681 if ((dstThread->type() == ThreadBase::MIXER || dstThread->type() == ThreadBase::DUPLICATING) && 2682 dstThread->mChannelCount != FCC_2) { 2683 ALOGW("moveEffectChain_l() effect chain failed because" 2684 " destination thread %p channel count(%u) != %u", 2685 dstThread, dstThread->mChannelCount, FCC_2); 2686 return INVALID_OPERATION; 2687 } 2688 2689 // remove chain first. This is useful only if reconfiguring effect chain on same output thread, 2690 // so that a new chain is created with correct parameters when first effect is added. This is 2691 // otherwise unnecessary as removeEffect_l() will remove the chain when last effect is 2692 // removed. 2693 srcThread->removeEffectChain_l(chain); 2694 2695 // transfer all effects one by one so that new effect chain is created on new thread with 2696 // correct buffer sizes and audio parameters and effect engines reconfigured accordingly 2697 sp<EffectChain> dstChain; 2698 uint32_t strategy = 0; // prevent compiler warning 2699 sp<EffectModule> effect = chain->getEffectFromId_l(0); 2700 Vector< sp<EffectModule> > removed; 2701 status_t status = NO_ERROR; 2702 while (effect != 0) { 2703 srcThread->removeEffect_l(effect); 2704 removed.add(effect); 2705 status = dstThread->addEffect_l(effect); 2706 if (status != NO_ERROR) { 2707 break; 2708 } 2709 // removeEffect_l() has stopped the effect if it was active so it must be restarted 2710 if (effect->state() == EffectModule::ACTIVE || 2711 effect->state() == EffectModule::STOPPING) { 2712 effect->start(); 2713 } 2714 // if the move request is not received from audio policy manager, the effect must be 2715 // re-registered with the new strategy and output 2716 if (dstChain == 0) { 2717 dstChain = effect->chain().promote(); 2718 if (dstChain == 0) { 2719 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get()); 2720 status = NO_INIT; 2721 break; 2722 } 2723 strategy = dstChain->strategy(); 2724 } 2725 if (reRegister) { 2726 AudioSystem::unregisterEffect(effect->id()); 2727 AudioSystem::registerEffect(&effect->desc(), 2728 dstThread->id(), 2729 strategy, 2730 sessionId, 2731 effect->id()); 2732 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2733 } 2734 effect = chain->getEffectFromId_l(0); 2735 } 2736 2737 if (status != NO_ERROR) { 2738 for (size_t i = 0; i < removed.size(); i++) { 2739 srcThread->addEffect_l(removed[i]); 2740 if (dstChain != 0 && reRegister) { 2741 AudioSystem::unregisterEffect(removed[i]->id()); 2742 AudioSystem::registerEffect(&removed[i]->desc(), 2743 srcThread->id(), 2744 strategy, 2745 sessionId, 2746 removed[i]->id()); 2747 AudioSystem::setEffectEnabled(effect->id(), effect->isEnabled()); 2748 } 2749 } 2750 } 2751 2752 return status; 2753} 2754 2755bool AudioFlinger::isNonOffloadableGlobalEffectEnabled_l() 2756{ 2757 if (mGlobalEffectEnableTime != 0 && 2758 ((systemTime() - mGlobalEffectEnableTime) < kMinGlobalEffectEnabletimeNs)) { 2759 return true; 2760 } 2761 2762 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2763 sp<EffectChain> ec = 2764 mPlaybackThreads.valueAt(i)->getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX); 2765 if (ec != 0 && ec->isNonOffloadableEnabled()) { 2766 return true; 2767 } 2768 } 2769 return false; 2770} 2771 2772void AudioFlinger::onNonOffloadableGlobalEffectEnable() 2773{ 2774 Mutex::Autolock _l(mLock); 2775 2776 mGlobalEffectEnableTime = systemTime(); 2777 2778 for (size_t i = 0; i < mPlaybackThreads.size(); i++) { 2779 sp<PlaybackThread> t = mPlaybackThreads.valueAt(i); 2780 if (t->mType == ThreadBase::OFFLOAD) { 2781 t->invalidateTracks(AUDIO_STREAM_MUSIC); 2782 } 2783 } 2784 2785} 2786 2787status_t AudioFlinger::putOrphanEffectChain_l(const sp<AudioFlinger::EffectChain>& chain) 2788{ 2789 audio_session_t session = (audio_session_t)chain->sessionId(); 2790 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2791 ALOGV("putOrphanEffectChain_l session %d index %d", session, index); 2792 if (index >= 0) { 2793 ALOGW("putOrphanEffectChain_l chain for session %d already present", session); 2794 return ALREADY_EXISTS; 2795 } 2796 mOrphanEffectChains.add(session, chain); 2797 return NO_ERROR; 2798} 2799 2800sp<AudioFlinger::EffectChain> AudioFlinger::getOrphanEffectChain_l(audio_session_t session) 2801{ 2802 sp<EffectChain> chain; 2803 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2804 ALOGV("getOrphanEffectChain_l session %d index %d", session, index); 2805 if (index >= 0) { 2806 chain = mOrphanEffectChains.valueAt(index); 2807 mOrphanEffectChains.removeItemsAt(index); 2808 } 2809 return chain; 2810} 2811 2812bool AudioFlinger::updateOrphanEffectChains(const sp<AudioFlinger::EffectModule>& effect) 2813{ 2814 Mutex::Autolock _l(mLock); 2815 audio_session_t session = (audio_session_t)effect->sessionId(); 2816 ssize_t index = mOrphanEffectChains.indexOfKey(session); 2817 ALOGV("updateOrphanEffectChains session %d index %d", session, index); 2818 if (index >= 0) { 2819 sp<EffectChain> chain = mOrphanEffectChains.valueAt(index); 2820 if (chain->removeEffect_l(effect) == 0) { 2821 ALOGV("updateOrphanEffectChains removing effect chain at index %d", index); 2822 mOrphanEffectChains.removeItemsAt(index); 2823 } 2824 return true; 2825 } 2826 return false; 2827} 2828 2829 2830struct Entry { 2831#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21 2832 char mFileName[TEE_MAX_FILENAME]; 2833}; 2834 2835int comparEntry(const void *p1, const void *p2) 2836{ 2837 return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName); 2838} 2839 2840#ifdef TEE_SINK 2841void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id) 2842{ 2843 NBAIO_Source *teeSource = source.get(); 2844 if (teeSource != NULL) { 2845 // .wav rotation 2846 // There is a benign race condition if 2 threads call this simultaneously. 2847 // They would both traverse the directory, but the result would simply be 2848 // failures at unlink() which are ignored. It's also unlikely since 2849 // normally dumpsys is only done by bugreport or from the command line. 2850 char teePath[32+256]; 2851 strcpy(teePath, "/data/misc/media"); 2852 size_t teePathLen = strlen(teePath); 2853 DIR *dir = opendir(teePath); 2854 teePath[teePathLen++] = '/'; 2855 if (dir != NULL) { 2856#define TEE_MAX_SORT 20 // number of entries to sort 2857#define TEE_MAX_KEEP 10 // number of entries to keep 2858 struct Entry entries[TEE_MAX_SORT]; 2859 size_t entryCount = 0; 2860 while (entryCount < TEE_MAX_SORT) { 2861 struct dirent de; 2862 struct dirent *result = NULL; 2863 int rc = readdir_r(dir, &de, &result); 2864 if (rc != 0) { 2865 ALOGW("readdir_r failed %d", rc); 2866 break; 2867 } 2868 if (result == NULL) { 2869 break; 2870 } 2871 if (result != &de) { 2872 ALOGW("readdir_r returned unexpected result %p != %p", result, &de); 2873 break; 2874 } 2875 // ignore non .wav file entries 2876 size_t nameLen = strlen(de.d_name); 2877 if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME || 2878 strcmp(&de.d_name[nameLen - 4], ".wav")) { 2879 continue; 2880 } 2881 strcpy(entries[entryCount++].mFileName, de.d_name); 2882 } 2883 (void) closedir(dir); 2884 if (entryCount > TEE_MAX_KEEP) { 2885 qsort(entries, entryCount, sizeof(Entry), comparEntry); 2886 for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) { 2887 strcpy(&teePath[teePathLen], entries[i].mFileName); 2888 (void) unlink(teePath); 2889 } 2890 } 2891 } else { 2892 if (fd >= 0) { 2893 dprintf(fd, "unable to rotate tees in %s: %s\n", teePath, strerror(errno)); 2894 } 2895 } 2896 char teeTime[16]; 2897 struct timeval tv; 2898 gettimeofday(&tv, NULL); 2899 struct tm tm; 2900 localtime_r(&tv.tv_sec, &tm); 2901 strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm); 2902 snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d.wav", teeTime, id); 2903 // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd 2904 int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR); 2905 if (teeFd >= 0) { 2906 // FIXME use libsndfile 2907 char wavHeader[44]; 2908 memcpy(wavHeader, 2909 "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", 2910 sizeof(wavHeader)); 2911 NBAIO_Format format = teeSource->format(); 2912 unsigned channelCount = Format_channelCount(format); 2913 uint32_t sampleRate = Format_sampleRate(format); 2914 size_t frameSize = Format_frameSize(format); 2915 wavHeader[22] = channelCount; // number of channels 2916 wavHeader[24] = sampleRate; // sample rate 2917 wavHeader[25] = sampleRate >> 8; 2918 wavHeader[32] = frameSize; // block alignment 2919 wavHeader[33] = frameSize >> 8; 2920 write(teeFd, wavHeader, sizeof(wavHeader)); 2921 size_t total = 0; 2922 bool firstRead = true; 2923#define TEE_SINK_READ 1024 // frames per I/O operation 2924 void *buffer = malloc(TEE_SINK_READ * frameSize); 2925 for (;;) { 2926 size_t count = TEE_SINK_READ; 2927 ssize_t actual = teeSource->read(buffer, count, 2928 AudioBufferProvider::kInvalidPTS); 2929 bool wasFirstRead = firstRead; 2930 firstRead = false; 2931 if (actual <= 0) { 2932 if (actual == (ssize_t) OVERRUN && wasFirstRead) { 2933 continue; 2934 } 2935 break; 2936 } 2937 ALOG_ASSERT(actual <= (ssize_t)count); 2938 write(teeFd, buffer, actual * frameSize); 2939 total += actual; 2940 } 2941 free(buffer); 2942 lseek(teeFd, (off_t) 4, SEEK_SET); 2943 uint32_t temp = 44 + total * frameSize - 8; 2944 // FIXME not big-endian safe 2945 write(teeFd, &temp, sizeof(temp)); 2946 lseek(teeFd, (off_t) 40, SEEK_SET); 2947 temp = total * frameSize; 2948 // FIXME not big-endian safe 2949 write(teeFd, &temp, sizeof(temp)); 2950 close(teeFd); 2951 if (fd >= 0) { 2952 dprintf(fd, "tee copied to %s\n", teePath); 2953 } 2954 } else { 2955 if (fd >= 0) { 2956 dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno)); 2957 } 2958 } 2959 } 2960} 2961#endif 2962 2963// ---------------------------------------------------------------------------- 2964 2965status_t AudioFlinger::onTransact( 2966 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags) 2967{ 2968 return BnAudioFlinger::onTransact(code, data, reply, flags); 2969} 2970 2971} // namespace android 2972