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 #ifndef ANDROID_AUDIO_FLINGER_H 19 #define ANDROID_AUDIO_FLINGER_H 20 21 #include "Configuration.h" 22 #include <atomic> 23 #include <mutex> 24 #include <deque> 25 #include <map> 26 #include <vector> 27 #include <stdint.h> 28 #include <sys/types.h> 29 #include <limits.h> 30 31 #include <android-base/macros.h> 32 33 #include <cutils/atomic.h> 34 #include <cutils/compiler.h> 35 #include <cutils/properties.h> 36 37 #include <media/IAudioFlinger.h> 38 #include <media/IAudioFlingerClient.h> 39 #include <media/IAudioTrack.h> 40 #include <media/AudioSystem.h> 41 #include <media/AudioTrack.h> 42 #include <media/MmapStreamInterface.h> 43 #include <media/MmapStreamCallback.h> 44 45 #include <utils/Errors.h> 46 #include <utils/threads.h> 47 #include <utils/SortedVector.h> 48 #include <utils/TypeHelpers.h> 49 #include <utils/Vector.h> 50 51 #include <binder/AppOpsManager.h> 52 #include <binder/BinderService.h> 53 #include <binder/IAppOpsCallback.h> 54 #include <binder/MemoryDealer.h> 55 56 #include <system/audio.h> 57 #include <system/audio_policy.h> 58 59 #include <media/audiohal/EffectBufferHalInterface.h> 60 #include <media/audiohal/StreamHalInterface.h> 61 #include <media/AudioBufferProvider.h> 62 #include <media/AudioMixer.h> 63 #include <media/ExtendedAudioBufferProvider.h> 64 #include <media/LinearMap.h> 65 #include <media/VolumeShaper.h> 66 67 #include <audio_utils/SimpleLog.h> 68 69 #include "FastCapture.h" 70 #include "FastMixer.h" 71 #include <media/nbaio/NBAIO.h> 72 #include "AudioWatchdog.h" 73 #include "AudioStreamOut.h" 74 #include "SpdifStreamOut.h" 75 #include "AudioHwDevice.h" 76 77 #include <powermanager/IPowerManager.h> 78 79 #include <media/nblog/NBLog.h> 80 #include <private/media/AudioEffectShared.h> 81 #include <private/media/AudioTrackShared.h> 82 83 #include "android/media/BnAudioRecord.h" 84 85 namespace android { 86 87 class AudioMixer; 88 class AudioBuffer; 89 class AudioResampler; 90 class DeviceHalInterface; 91 class DevicesFactoryHalInterface; 92 class EffectsFactoryHalInterface; 93 class FastMixer; 94 class PassthruBufferProvider; 95 class RecordBufferConverter; 96 class ServerProxy; 97 98 // ---------------------------------------------------------------------------- 99 100 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 101 102 #define INCLUDING_FROM_AUDIOFLINGER_H 103 104 class AudioFlinger : 105 public BinderService<AudioFlinger>, 106 public BnAudioFlinger 107 { 108 friend class BinderService<AudioFlinger>; // for AudioFlinger() 109 110 public: getServiceName()111 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 112 113 virtual status_t dump(int fd, const Vector<String16>& args); 114 115 // IAudioFlinger interface, in binder opcode order 116 virtual sp<IAudioTrack> createTrack(const CreateTrackInput& input, 117 CreateTrackOutput& output, 118 status_t *status); 119 120 virtual sp<media::IAudioRecord> createRecord(const CreateRecordInput& input, 121 CreateRecordOutput& output, 122 status_t *status); 123 124 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 125 virtual audio_format_t format(audio_io_handle_t output) const; 126 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 127 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 128 virtual uint32_t latency(audio_io_handle_t output) const; 129 130 virtual status_t setMasterVolume(float value); 131 virtual status_t setMasterMute(bool muted); 132 133 virtual float masterVolume() const; 134 virtual bool masterMute() const; 135 136 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 137 audio_io_handle_t output); 138 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 139 140 virtual float streamVolume(audio_stream_type_t stream, 141 audio_io_handle_t output) const; 142 virtual bool streamMute(audio_stream_type_t stream) const; 143 144 virtual status_t setMode(audio_mode_t mode); 145 146 virtual status_t setMicMute(bool state); 147 virtual bool getMicMute() const; 148 149 virtual void setRecordSilenced(uid_t uid, bool silenced); 150 151 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 152 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 153 154 virtual void registerClient(const sp<IAudioFlingerClient>& client); 155 156 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 157 audio_channel_mask_t channelMask) const; 158 159 virtual status_t openOutput(audio_module_handle_t module, 160 audio_io_handle_t *output, 161 audio_config_t *config, 162 audio_devices_t *devices, 163 const String8& address, 164 uint32_t *latencyMs, 165 audio_output_flags_t flags); 166 167 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 168 audio_io_handle_t output2); 169 170 virtual status_t closeOutput(audio_io_handle_t output); 171 172 virtual status_t suspendOutput(audio_io_handle_t output); 173 174 virtual status_t restoreOutput(audio_io_handle_t output); 175 176 virtual status_t openInput(audio_module_handle_t module, 177 audio_io_handle_t *input, 178 audio_config_t *config, 179 audio_devices_t *device, 180 const String8& address, 181 audio_source_t source, 182 audio_input_flags_t flags); 183 184 virtual status_t closeInput(audio_io_handle_t input); 185 186 virtual status_t invalidateStream(audio_stream_type_t stream); 187 188 virtual status_t setVoiceVolume(float volume); 189 190 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 191 audio_io_handle_t output) const; 192 193 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 194 195 // This is the binder API. For the internal API see nextUniqueId(). 196 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 197 198 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 199 200 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 201 202 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 203 204 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 205 206 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 207 effect_descriptor_t *descriptor) const; 208 209 virtual sp<IEffect> createEffect( 210 effect_descriptor_t *pDesc, 211 const sp<IEffectClient>& effectClient, 212 int32_t priority, 213 audio_io_handle_t io, 214 audio_session_t sessionId, 215 const String16& opPackageName, 216 pid_t pid, 217 status_t *status /*non-NULL*/, 218 int *id, 219 int *enabled); 220 221 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 222 audio_io_handle_t dstOutput); 223 224 virtual audio_module_handle_t loadHwModule(const char *name); 225 226 virtual uint32_t getPrimaryOutputSamplingRate(); 227 virtual size_t getPrimaryOutputFrameCount(); 228 229 virtual status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory) override; 230 231 /* List available audio ports and their attributes */ 232 virtual status_t listAudioPorts(unsigned int *num_ports, 233 struct audio_port *ports); 234 235 /* Get attributes for a given audio port */ 236 virtual status_t getAudioPort(struct audio_port *port); 237 238 /* Create an audio patch between several source and sink ports */ 239 virtual status_t createAudioPatch(const struct audio_patch *patch, 240 audio_patch_handle_t *handle); 241 242 /* Release an audio patch */ 243 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 244 245 /* List existing audio patches */ 246 virtual status_t listAudioPatches(unsigned int *num_patches, 247 struct audio_patch *patches); 248 249 /* Set audio port configuration */ 250 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 251 252 /* Get the HW synchronization source used for an audio session */ 253 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 254 255 /* Indicate JAVA services are ready (scheduling, power management ...) */ 256 virtual status_t systemReady(); 257 258 virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones); 259 260 virtual status_t onTransact( 261 uint32_t code, 262 const Parcel& data, 263 Parcel* reply, 264 uint32_t flags); 265 266 // end of IAudioFlinger interface 267 268 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 269 void unregisterWriter(const sp<NBLog::Writer>& writer); 270 sp<EffectsFactoryHalInterface> getEffectsFactory(); 271 272 status_t openMmapStream(MmapStreamInterface::stream_direction_t direction, 273 const audio_attributes_t *attr, 274 audio_config_base_t *config, 275 const AudioClient& client, 276 audio_port_handle_t *deviceId, 277 audio_session_t *sessionId, 278 const sp<MmapStreamCallback>& callback, 279 sp<MmapStreamInterface>& interface, 280 audio_port_handle_t *handle); 281 private: 282 // FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed. 283 static const size_t kLogMemorySize = 400 * 1024; 284 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 285 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 286 // for as long as possible. The memory is only freed when it is needed for another log writer. 287 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 288 Mutex mUnregisteredWritersLock; 289 290 public: 291 292 class SyncEvent; 293 294 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 295 296 class SyncEvent : public RefBase { 297 public: SyncEvent(AudioSystem::sync_event_t type,audio_session_t triggerSession,audio_session_t listenerSession,sync_event_callback_t callBack,wp<RefBase> cookie)298 SyncEvent(AudioSystem::sync_event_t type, 299 audio_session_t triggerSession, 300 audio_session_t listenerSession, 301 sync_event_callback_t callBack, 302 wp<RefBase> cookie) 303 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 304 mCallback(callBack), mCookie(cookie) 305 {} 306 ~SyncEvent()307 virtual ~SyncEvent() {} 308 trigger()309 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } isCancelled()310 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } cancel()311 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } type()312 AudioSystem::sync_event_t type() const { return mType; } triggerSession()313 audio_session_t triggerSession() const { return mTriggerSession; } listenerSession()314 audio_session_t listenerSession() const { return mListenerSession; } cookie()315 wp<RefBase> cookie() const { return mCookie; } 316 317 private: 318 const AudioSystem::sync_event_t mType; 319 const audio_session_t mTriggerSession; 320 const audio_session_t mListenerSession; 321 sync_event_callback_t mCallback; 322 const wp<RefBase> mCookie; 323 mutable Mutex mLock; 324 }; 325 326 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 327 audio_session_t triggerSession, 328 audio_session_t listenerSession, 329 sync_event_callback_t callBack, 330 const wp<RefBase>& cookie); 331 btNrecIsOff()332 bool btNrecIsOff() const { return mBtNrecIsOff.load(); } 333 334 335 private: 336 getMode()337 audio_mode_t getMode() const { return mMode; } 338 339 AudioFlinger() ANDROID_API; 340 virtual ~AudioFlinger(); 341 342 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev initCheck()343 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 344 NO_INIT : NO_ERROR; } 345 346 // RefBase 347 virtual void onFirstRef(); 348 349 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 350 audio_devices_t devices); 351 void purgeStaleEffects_l(); 352 353 // Set kEnableExtendedChannels to true to enable greater than stereo output 354 // for the MixerThread and device sink. Number of channels allowed is 355 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 356 static const bool kEnableExtendedChannels = true; 357 358 // Returns true if channel mask is permitted for the PCM sink in the MixerThread isValidPcmSinkChannelMask(audio_channel_mask_t channelMask)359 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 360 switch (audio_channel_mask_get_representation(channelMask)) { 361 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 362 uint32_t channelCount = FCC_2; // stereo is default 363 if (kEnableExtendedChannels) { 364 channelCount = audio_channel_count_from_out_mask(channelMask); 365 if (channelCount < FCC_2 // mono is not supported at this time 366 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 367 return false; 368 } 369 } 370 // check that channelMask is the "canonical" one we expect for the channelCount. 371 return channelMask == audio_channel_out_mask_from_count(channelCount); 372 } 373 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 374 if (kEnableExtendedChannels) { 375 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 376 if (channelCount >= FCC_2 // mono is not supported at this time 377 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 378 return true; 379 } 380 } 381 return false; 382 default: 383 return false; 384 } 385 } 386 387 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 388 static const bool kEnableExtendedPrecision = true; 389 390 // Returns true if format is permitted for the PCM sink in the MixerThread isValidPcmSinkFormat(audio_format_t format)391 static inline bool isValidPcmSinkFormat(audio_format_t format) { 392 switch (format) { 393 case AUDIO_FORMAT_PCM_16_BIT: 394 return true; 395 case AUDIO_FORMAT_PCM_FLOAT: 396 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 397 case AUDIO_FORMAT_PCM_32_BIT: 398 case AUDIO_FORMAT_PCM_8_24_BIT: 399 return kEnableExtendedPrecision; 400 default: 401 return false; 402 } 403 } 404 405 // standby delay for MIXER and DUPLICATING playback threads is read from property 406 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 407 static nsecs_t mStandbyTimeInNsecs; 408 409 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 410 // AudioFlinger::setParameters() updates, other threads read w/o lock 411 static uint32_t mScreenState; 412 413 // Internal dump utilities. 414 static const int kDumpLockRetries = 50; 415 static const int kDumpLockSleepUs = 20000; 416 static bool dumpTryLock(Mutex& mutex); 417 void dumpPermissionDenial(int fd, const Vector<String16>& args); 418 void dumpClients(int fd, const Vector<String16>& args); 419 void dumpInternals(int fd, const Vector<String16>& args); 420 421 // --- Client --- 422 class Client : public RefBase { 423 public: 424 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 425 virtual ~Client(); 426 sp<MemoryDealer> heap() const; pid()427 pid_t pid() const { return mPid; } audioFlinger()428 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 429 430 private: 431 DISALLOW_COPY_AND_ASSIGN(Client); 432 433 const sp<AudioFlinger> mAudioFlinger; 434 sp<MemoryDealer> mMemoryDealer; 435 const pid_t mPid; 436 }; 437 438 // --- Notification Client --- 439 class NotificationClient : public IBinder::DeathRecipient { 440 public: 441 NotificationClient(const sp<AudioFlinger>& audioFlinger, 442 const sp<IAudioFlingerClient>& client, 443 pid_t pid); 444 virtual ~NotificationClient(); 445 audioFlingerClient()446 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 447 448 // IBinder::DeathRecipient 449 virtual void binderDied(const wp<IBinder>& who); 450 451 private: 452 DISALLOW_COPY_AND_ASSIGN(NotificationClient); 453 454 const sp<AudioFlinger> mAudioFlinger; 455 const pid_t mPid; 456 const sp<IAudioFlingerClient> mAudioFlingerClient; 457 }; 458 459 // --- MediaLogNotifier --- 460 // Thread in charge of notifying MediaLogService to start merging. 461 // Receives requests from AudioFlinger's binder activity. It is used to reduce the amount of 462 // binder calls to MediaLogService in case of bursts of AudioFlinger binder calls. 463 class MediaLogNotifier : public Thread { 464 public: 465 MediaLogNotifier(); 466 467 // Requests a MediaLogService notification. It's ignored if there has recently been another 468 void requestMerge(); 469 private: 470 // Every iteration blocks waiting for a request, then interacts with MediaLogService to 471 // start merging. 472 // As every MediaLogService binder call is expensive, once it gets a request it ignores the 473 // following ones for a period of time. 474 virtual bool threadLoop() override; 475 476 bool mPendingRequests; 477 478 // Mutex and condition variable around mPendingRequests' value 479 Mutex mMutex; 480 Condition mCond; 481 482 // Duration of the sleep period after a processed request 483 static const int kPostTriggerSleepPeriod = 1000000; 484 }; 485 486 const sp<MediaLogNotifier> mMediaLogNotifier; 487 488 // This is a helper that is called during incoming binder calls. 489 void requestLogMerge(); 490 491 class TrackHandle; 492 class RecordHandle; 493 class RecordThread; 494 class PlaybackThread; 495 class MixerThread; 496 class DirectOutputThread; 497 class OffloadThread; 498 class DuplicatingThread; 499 class AsyncCallbackThread; 500 class Track; 501 class RecordTrack; 502 class EffectModule; 503 class EffectHandle; 504 class EffectChain; 505 506 struct AudioStreamIn; 507 508 struct stream_type_t { stream_type_tstream_type_t509 stream_type_t() 510 : volume(1.0f), 511 mute(false) 512 { 513 } 514 float volume; 515 bool mute; 516 }; 517 518 // --- PlaybackThread --- 519 #ifdef FLOAT_EFFECT_CHAIN 520 #define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_FLOAT 521 using effect_buffer_t = float; 522 #else 523 #define EFFECT_BUFFER_FORMAT AUDIO_FORMAT_PCM_16_BIT 524 using effect_buffer_t = int16_t; 525 #endif 526 527 #include "Threads.h" 528 529 #include "Effects.h" 530 531 #include "PatchPanel.h" 532 533 // server side of the client's IAudioTrack 534 class TrackHandle : public android::BnAudioTrack { 535 public: 536 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 537 virtual ~TrackHandle(); 538 virtual sp<IMemory> getCblk() const; 539 virtual status_t start(); 540 virtual void stop(); 541 virtual void flush(); 542 virtual void pause(); 543 virtual status_t attachAuxEffect(int effectId); 544 virtual status_t setParameters(const String8& keyValuePairs); 545 virtual media::VolumeShaper::Status applyVolumeShaper( 546 const sp<media::VolumeShaper::Configuration>& configuration, 547 const sp<media::VolumeShaper::Operation>& operation) override; 548 virtual sp<media::VolumeShaper::State> getVolumeShaperState(int id) override; 549 virtual status_t getTimestamp(AudioTimestamp& timestamp); 550 virtual void signal(); // signal playback thread for a change in control block 551 552 virtual status_t onTransact( 553 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 554 555 private: 556 const sp<PlaybackThread::Track> mTrack; 557 }; 558 559 // server side of the client's IAudioRecord 560 class RecordHandle : public android::media::BnAudioRecord { 561 public: 562 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 563 virtual ~RecordHandle(); 564 virtual binder::Status start(int /*AudioSystem::sync_event_t*/ event, 565 int /*audio_session_t*/ triggerSession); 566 virtual binder::Status stop(); 567 virtual binder::Status getActiveMicrophones( 568 std::vector<media::MicrophoneInfo>* activeMicrophones); 569 private: 570 const sp<RecordThread::RecordTrack> mRecordTrack; 571 572 // for use from destructor 573 void stop_nonvirtual(); 574 }; 575 576 // Mmap stream control interface implementation. Each MmapThreadHandle controls one 577 // MmapPlaybackThread or MmapCaptureThread instance. 578 class MmapThreadHandle : public MmapStreamInterface { 579 public: 580 explicit MmapThreadHandle(const sp<MmapThread>& thread); 581 virtual ~MmapThreadHandle(); 582 583 // MmapStreamInterface virtuals 584 virtual status_t createMmapBuffer(int32_t minSizeFrames, 585 struct audio_mmap_buffer_info *info); 586 virtual status_t getMmapPosition(struct audio_mmap_position *position); 587 virtual status_t start(const AudioClient& client, 588 audio_port_handle_t *handle); 589 virtual status_t stop(audio_port_handle_t handle); 590 virtual status_t standby(); 591 592 private: 593 const sp<MmapThread> mThread; 594 }; 595 596 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 597 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 598 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 599 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 600 MmapThread *checkMmapThread_l(audio_io_handle_t io) const; 601 VolumeInterface *getVolumeInterface_l(audio_io_handle_t output) const; 602 Vector <VolumeInterface *> getAllVolumeInterfaces_l() const; 603 604 sp<ThreadBase> openInput_l(audio_module_handle_t module, 605 audio_io_handle_t *input, 606 audio_config_t *config, 607 audio_devices_t device, 608 const String8& address, 609 audio_source_t source, 610 audio_input_flags_t flags); 611 sp<ThreadBase> openOutput_l(audio_module_handle_t module, 612 audio_io_handle_t *output, 613 audio_config_t *config, 614 audio_devices_t devices, 615 const String8& address, 616 audio_output_flags_t flags); 617 618 void closeOutputFinish(const sp<PlaybackThread>& thread); 619 void closeInputFinish(const sp<RecordThread>& thread); 620 621 // no range check, AudioFlinger::mLock held streamMute_l(audio_stream_type_t stream)622 bool streamMute_l(audio_stream_type_t stream) const 623 { return mStreamTypes[stream].mute; } 624 void ioConfigChanged(audio_io_config_event event, 625 const sp<AudioIoDescriptor>& ioDesc, 626 pid_t pid = 0); 627 628 // Allocate an audio_unique_id_t. 629 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 630 // audio_module_handle_t, and audio_patch_handle_t. 631 // They all share the same ID space, but the namespaces are actually independent 632 // because there are separate KeyedVectors for each kind of ID. 633 // The return value is cast to the specific type depending on how the ID will be used. 634 // FIXME This API does not handle rollover to zero (for unsigned IDs), 635 // or from positive to negative (for signed IDs). 636 // Thus it may fail by returning an ID of the wrong sign, 637 // or by returning a non-unique ID. 638 // This is the internal API. For the binder API see newAudioUniqueId(). 639 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 640 641 status_t moveEffectChain_l(audio_session_t sessionId, 642 PlaybackThread *srcThread, 643 PlaybackThread *dstThread, 644 bool reRegister); 645 646 // return thread associated with primary hardware device, or NULL 647 PlaybackThread *primaryPlaybackThread_l() const; 648 audio_devices_t primaryOutputDevice_l() const; 649 650 // return the playback thread with smallest HAL buffer size, and prefer fast 651 PlaybackThread *fastPlaybackThread_l() const; 652 653 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 654 655 656 void removeClient_l(pid_t pid); 657 void removeNotificationClient(pid_t pid); 658 bool isNonOffloadableGlobalEffectEnabled_l(); 659 void onNonOffloadableGlobalEffectEnable(); 660 bool isSessionAcquired_l(audio_session_t audioSession); 661 662 // Store an effect chain to mOrphanEffectChains keyed vector. 663 // Called when a thread exits and effects are still attached to it. 664 // If effects are later created on the same session, they will reuse the same 665 // effect chain and same instances in the effect library. 666 // return ALREADY_EXISTS if a chain with the same session already exists in 667 // mOrphanEffectChains. Note that this should never happen as there is only one 668 // chain for a given session and it is attached to only one thread at a time. 669 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 670 // Get an effect chain for the specified session in mOrphanEffectChains and remove 671 // it if found. Returns 0 if not found (this is the most common case). 672 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 673 // Called when the last effect handle on an effect instance is removed. If this 674 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 675 // and removed from mOrphanEffectChains if it does not contain any effect. 676 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 677 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 678 679 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 680 681 // AudioStreamIn is immutable, so their fields are const. 682 // For emphasis, we could also make all pointers to them be "const *", 683 // but that would clutter the code unnecessarily. 684 685 struct AudioStreamIn { 686 AudioHwDevice* const audioHwDev; 687 sp<StreamInHalInterface> stream; 688 audio_input_flags_t flags; 689 hwDevAudioStreamIn690 sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); } 691 AudioStreamInAudioStreamIn692 AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) : 693 audioHwDev(dev), stream(in), flags(flags) {} 694 }; 695 696 // for mAudioSessionRefs only 697 struct AudioSessionRef { AudioSessionRefAudioSessionRef698 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 699 mSessionid(sessionid), mPid(pid), mCnt(1) {} 700 const audio_session_t mSessionid; 701 const pid_t mPid; 702 int mCnt; 703 }; 704 705 mutable Mutex mLock; 706 // protects mClients and mNotificationClients. 707 // must be locked after mLock and ThreadBase::mLock if both must be locked 708 // avoids acquiring AudioFlinger::mLock from inside thread loop. 709 mutable Mutex mClientLock; 710 // protected by mClientLock 711 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 712 713 mutable Mutex mHardwareLock; 714 // NOTE: If both mLock and mHardwareLock mutexes must be held, 715 // always take mLock before mHardwareLock 716 717 // These two fields are immutable after onFirstRef(), so no lock needed to access 718 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 719 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 720 721 sp<DevicesFactoryHalInterface> mDevicesFactoryHal; 722 723 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 724 enum hardware_call_state { 725 AUDIO_HW_IDLE = 0, // no operation in progress 726 AUDIO_HW_INIT, // init_check 727 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 728 AUDIO_HW_OUTPUT_CLOSE, // unused 729 AUDIO_HW_INPUT_OPEN, // unused 730 AUDIO_HW_INPUT_CLOSE, // unused 731 AUDIO_HW_STANDBY, // unused 732 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 733 AUDIO_HW_GET_ROUTING, // unused 734 AUDIO_HW_SET_ROUTING, // unused 735 AUDIO_HW_GET_MODE, // unused 736 AUDIO_HW_SET_MODE, // set_mode 737 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 738 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 739 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 740 AUDIO_HW_SET_PARAMETER, // set_parameters 741 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 742 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 743 AUDIO_HW_GET_PARAMETER, // get_parameters 744 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 745 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 746 }; 747 748 mutable hardware_call_state mHardwareStatus; // for dump only 749 750 751 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 752 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 753 754 // member variables below are protected by mLock 755 float mMasterVolume; 756 bool mMasterMute; 757 // end of variables protected by mLock 758 759 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 760 761 // protected by mClientLock 762 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 763 764 // updated by atomic_fetch_add_explicit 765 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 766 767 audio_mode_t mMode; 768 std::atomic_bool mBtNrecIsOff; 769 770 // protected by mLock 771 Vector<AudioSessionRef*> mAudioSessionRefs; 772 773 float masterVolume_l() const; 774 bool masterMute_l() const; 775 audio_module_handle_t loadHwModule_l(const char *name); 776 777 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 778 // to be created 779 780 // Effect chains without a valid thread 781 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 782 783 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 784 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 785 786 // list of MMAP stream control threads. Those threads allow for wake lock, routing 787 // and volume control for activity on the associated MMAP stream at the HAL. 788 // Audio data transfer is directly handled by the client creating the MMAP stream 789 DefaultKeyedVector< audio_io_handle_t, sp<MmapThread> > mMmapThreads; 790 791 private: 792 sp<Client> registerPid(pid_t pid); // always returns non-0 793 794 // for use from destructor 795 status_t closeOutput_nonvirtual(audio_io_handle_t output); 796 void closeOutputInternal_l(const sp<PlaybackThread>& thread); 797 status_t closeInput_nonvirtual(audio_io_handle_t input); 798 void closeInputInternal_l(const sp<RecordThread>& thread); 799 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 800 801 status_t checkStreamType(audio_stream_type_t stream) const; 802 803 void filterReservedParameters(String8& keyValuePairs, uid_t callingUid); 804 805 #ifdef TEE_SINK 806 // all record threads serially share a common tee sink, which is re-created on format change 807 sp<NBAIO_Sink> mRecordTeeSink; 808 sp<NBAIO_Source> mRecordTeeSource; 809 #endif 810 811 public: 812 813 #ifdef TEE_SINK 814 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 815 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id, char suffix); 816 817 // whether tee sink is enabled by property 818 static bool mTeeSinkInputEnabled; 819 static bool mTeeSinkOutputEnabled; 820 static bool mTeeSinkTrackEnabled; 821 822 // runtime configured size of each tee sink pipe, in frames 823 static size_t mTeeSinkInputFrames; 824 static size_t mTeeSinkOutputFrames; 825 static size_t mTeeSinkTrackFrames; 826 827 // compile-time default size of tee sink pipes, in frames 828 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 829 static const size_t kTeeSinkInputFramesDefault = 0x200000; 830 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 831 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 832 #endif 833 834 // These methods read variables atomically without mLock, 835 // though the variables are updated with mLock. isLowRamDevice()836 bool isLowRamDevice() const { return mIsLowRamDevice; } 837 size_t getClientSharedHeapSize() const; 838 839 private: 840 std::atomic<bool> mIsLowRamDevice; 841 bool mIsDeviceTypeKnown; 842 int64_t mTotalMemory; 843 std::atomic<size_t> mClientSharedHeapSize; 844 static constexpr size_t kMinimumClientSharedHeapSizeBytes = 1024 * 1024; // 1MB 845 846 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 847 848 sp<PatchPanel> mPatchPanel; 849 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 850 851 bool mSystemReady; 852 }; 853 854 #undef INCLUDING_FROM_AUDIOFLINGER_H 855 856 std::string formatToString(audio_format_t format); 857 std::string inputFlagsToString(audio_input_flags_t flags); 858 std::string outputFlagsToString(audio_output_flags_t flags); 859 std::string devicesToString(audio_devices_t devices); 860 const char *sourceToString(audio_source_t source); 861 862 // ---------------------------------------------------------------------------- 863 864 } // namespace android 865 866 #endif // ANDROID_AUDIO_FLINGER_H 867