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 <deque> 23 #include <map> 24 #include <stdint.h> 25 #include <sys/types.h> 26 #include <limits.h> 27 28 #include <cutils/compiler.h> 29 #include <cutils/properties.h> 30 31 #include <media/IAudioFlinger.h> 32 #include <media/IAudioFlingerClient.h> 33 #include <media/IAudioTrack.h> 34 #include <media/IAudioRecord.h> 35 #include <media/AudioSystem.h> 36 #include <media/AudioTrack.h> 37 #include <media/MmapStreamInterface.h> 38 #include <media/MmapStreamCallback.h> 39 40 #include <utils/Atomic.h> 41 #include <utils/Errors.h> 42 #include <utils/threads.h> 43 #include <utils/SortedVector.h> 44 #include <utils/TypeHelpers.h> 45 #include <utils/Vector.h> 46 47 #include <binder/BinderService.h> 48 #include <binder/MemoryDealer.h> 49 50 #include <system/audio.h> 51 #include <system/audio_policy.h> 52 53 #include <media/audiohal/EffectBufferHalInterface.h> 54 #include <media/audiohal/StreamHalInterface.h> 55 #include <media/AudioBufferProvider.h> 56 #include <media/AudioMixer.h> 57 #include <media/ExtendedAudioBufferProvider.h> 58 #include <media/LinearMap.h> 59 #include <media/VolumeShaper.h> 60 61 #include <audio_utils/SimpleLog.h> 62 63 #include "FastCapture.h" 64 #include "FastMixer.h" 65 #include <media/nbaio/NBAIO.h> 66 #include "AudioWatchdog.h" 67 #include "AudioStreamOut.h" 68 #include "SpdifStreamOut.h" 69 #include "AudioHwDevice.h" 70 71 #include <powermanager/IPowerManager.h> 72 73 #include <media/nbaio/NBLog.h> 74 #include <private/media/AudioTrackShared.h> 75 76 namespace android { 77 78 struct audio_track_cblk_t; 79 struct effect_param_cblk_t; 80 class AudioMixer; 81 class AudioBuffer; 82 class AudioResampler; 83 class DeviceHalInterface; 84 class DevicesFactoryHalInterface; 85 class EffectsFactoryHalInterface; 86 class FastMixer; 87 class PassthruBufferProvider; 88 class RecordBufferConverter; 89 class ServerProxy; 90 91 // ---------------------------------------------------------------------------- 92 93 static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 94 95 96 // Max shared memory size for audio tracks and audio records per client process 97 static const size_t kClientSharedHeapSizeBytes = 1024*1024; 98 // Shared memory size multiplier for non low ram devices 99 static const size_t kClientSharedHeapSizeMultiplier = 4; 100 101 #define INCLUDING_FROM_AUDIOFLINGER_H 102 103 class AudioFlinger : 104 public BinderService<AudioFlinger>, 105 public BnAudioFlinger 106 { 107 friend class BinderService<AudioFlinger>; // for AudioFlinger() 108 109 public: getServiceName()110 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 111 112 virtual status_t dump(int fd, const Vector<String16>& args); 113 114 // IAudioFlinger interface, in binder opcode order 115 virtual sp<IAudioTrack> createTrack( 116 audio_stream_type_t streamType, 117 uint32_t sampleRate, 118 audio_format_t format, 119 audio_channel_mask_t channelMask, 120 size_t *pFrameCount, 121 audio_output_flags_t *flags, 122 const sp<IMemory>& sharedBuffer, 123 audio_io_handle_t output, 124 pid_t pid, 125 pid_t tid, 126 audio_session_t *sessionId, 127 int clientUid, 128 status_t *status /*non-NULL*/, 129 audio_port_handle_t portId); 130 131 virtual sp<IAudioRecord> openRecord( 132 audio_io_handle_t input, 133 uint32_t sampleRate, 134 audio_format_t format, 135 audio_channel_mask_t channelMask, 136 const String16& opPackageName, 137 size_t *pFrameCount, 138 audio_input_flags_t *flags, 139 pid_t pid, 140 pid_t tid, 141 int clientUid, 142 audio_session_t *sessionId, 143 size_t *notificationFrames, 144 sp<IMemory>& cblk, 145 sp<IMemory>& buffers, 146 status_t *status /*non-NULL*/, 147 audio_port_handle_t portId); 148 149 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 150 virtual audio_format_t format(audio_io_handle_t output) const; 151 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 152 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 153 virtual uint32_t latency(audio_io_handle_t output) const; 154 155 virtual status_t setMasterVolume(float value); 156 virtual status_t setMasterMute(bool muted); 157 158 virtual float masterVolume() const; 159 virtual bool masterMute() const; 160 161 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 162 audio_io_handle_t output); 163 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 164 165 virtual float streamVolume(audio_stream_type_t stream, 166 audio_io_handle_t output) const; 167 virtual bool streamMute(audio_stream_type_t stream) const; 168 169 virtual status_t setMode(audio_mode_t mode); 170 171 virtual status_t setMicMute(bool state); 172 virtual bool getMicMute() const; 173 174 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 175 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 176 177 virtual void registerClient(const sp<IAudioFlingerClient>& client); 178 179 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 180 audio_channel_mask_t channelMask) const; 181 182 virtual status_t openOutput(audio_module_handle_t module, 183 audio_io_handle_t *output, 184 audio_config_t *config, 185 audio_devices_t *devices, 186 const String8& address, 187 uint32_t *latencyMs, 188 audio_output_flags_t flags); 189 190 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 191 audio_io_handle_t output2); 192 193 virtual status_t closeOutput(audio_io_handle_t output); 194 195 virtual status_t suspendOutput(audio_io_handle_t output); 196 197 virtual status_t restoreOutput(audio_io_handle_t output); 198 199 virtual status_t openInput(audio_module_handle_t module, 200 audio_io_handle_t *input, 201 audio_config_t *config, 202 audio_devices_t *device, 203 const String8& address, 204 audio_source_t source, 205 audio_input_flags_t flags); 206 207 virtual status_t closeInput(audio_io_handle_t input); 208 209 virtual status_t invalidateStream(audio_stream_type_t stream); 210 211 virtual status_t setVoiceVolume(float volume); 212 213 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 214 audio_io_handle_t output) const; 215 216 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 217 218 // This is the binder API. For the internal API see nextUniqueId(). 219 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 220 221 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 222 223 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 224 225 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 226 227 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 228 229 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 230 effect_descriptor_t *descriptor) const; 231 232 virtual sp<IEffect> createEffect( 233 effect_descriptor_t *pDesc, 234 const sp<IEffectClient>& effectClient, 235 int32_t priority, 236 audio_io_handle_t io, 237 audio_session_t sessionId, 238 const String16& opPackageName, 239 pid_t pid, 240 status_t *status /*non-NULL*/, 241 int *id, 242 int *enabled); 243 244 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 245 audio_io_handle_t dstOutput); 246 247 virtual audio_module_handle_t loadHwModule(const char *name); 248 249 virtual uint32_t getPrimaryOutputSamplingRate(); 250 virtual size_t getPrimaryOutputFrameCount(); 251 252 virtual status_t setLowRamDevice(bool isLowRamDevice); 253 254 /* List available audio ports and their attributes */ 255 virtual status_t listAudioPorts(unsigned int *num_ports, 256 struct audio_port *ports); 257 258 /* Get attributes for a given audio port */ 259 virtual status_t getAudioPort(struct audio_port *port); 260 261 /* Create an audio patch between several source and sink ports */ 262 virtual status_t createAudioPatch(const struct audio_patch *patch, 263 audio_patch_handle_t *handle); 264 265 /* Release an audio patch */ 266 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 267 268 /* List existing audio patches */ 269 virtual status_t listAudioPatches(unsigned int *num_patches, 270 struct audio_patch *patches); 271 272 /* Set audio port configuration */ 273 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 274 275 /* Get the HW synchronization source used for an audio session */ 276 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 277 278 /* Indicate JAVA services are ready (scheduling, power management ...) */ 279 virtual status_t systemReady(); 280 281 virtual status_t onTransact( 282 uint32_t code, 283 const Parcel& data, 284 Parcel* reply, 285 uint32_t flags); 286 287 // end of IAudioFlinger interface 288 289 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 290 void unregisterWriter(const sp<NBLog::Writer>& writer); 291 sp<EffectsFactoryHalInterface> getEffectsFactory(); 292 293 status_t openMmapStream(MmapStreamInterface::stream_direction_t direction, 294 const audio_attributes_t *attr, 295 audio_config_base_t *config, 296 const MmapStreamInterface::Client& client, 297 audio_port_handle_t *deviceId, 298 const sp<MmapStreamCallback>& callback, 299 sp<MmapStreamInterface>& interface); 300 private: 301 static const size_t kLogMemorySize = 40 * 1024; 302 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 303 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 304 // for as long as possible. The memory is only freed when it is needed for another log writer. 305 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 306 Mutex mUnregisteredWritersLock; 307 308 public: 309 310 class SyncEvent; 311 312 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 313 314 class SyncEvent : public RefBase { 315 public: SyncEvent(AudioSystem::sync_event_t type,audio_session_t triggerSession,audio_session_t listenerSession,sync_event_callback_t callBack,wp<RefBase> cookie)316 SyncEvent(AudioSystem::sync_event_t type, 317 audio_session_t triggerSession, 318 audio_session_t listenerSession, 319 sync_event_callback_t callBack, 320 wp<RefBase> cookie) 321 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 322 mCallback(callBack), mCookie(cookie) 323 {} 324 ~SyncEvent()325 virtual ~SyncEvent() {} 326 trigger()327 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } isCancelled()328 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } cancel()329 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } type()330 AudioSystem::sync_event_t type() const { return mType; } triggerSession()331 audio_session_t triggerSession() const { return mTriggerSession; } listenerSession()332 audio_session_t listenerSession() const { return mListenerSession; } cookie()333 wp<RefBase> cookie() const { return mCookie; } 334 335 private: 336 const AudioSystem::sync_event_t mType; 337 const audio_session_t mTriggerSession; 338 const audio_session_t mListenerSession; 339 sync_event_callback_t mCallback; 340 const wp<RefBase> mCookie; 341 mutable Mutex mLock; 342 }; 343 344 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 345 audio_session_t triggerSession, 346 audio_session_t listenerSession, 347 sync_event_callback_t callBack, 348 const wp<RefBase>& cookie); 349 350 private: 351 getMode()352 audio_mode_t getMode() const { return mMode; } 353 btNrecIsOff()354 bool btNrecIsOff() const { return mBtNrecIsOff; } 355 356 AudioFlinger() ANDROID_API; 357 virtual ~AudioFlinger(); 358 359 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev initCheck()360 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 361 NO_INIT : NO_ERROR; } 362 363 // RefBase 364 virtual void onFirstRef(); 365 366 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 367 audio_devices_t devices); 368 void purgeStaleEffects_l(); 369 370 // Set kEnableExtendedChannels to true to enable greater than stereo output 371 // for the MixerThread and device sink. Number of channels allowed is 372 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 373 static const bool kEnableExtendedChannels = true; 374 375 // Returns true if channel mask is permitted for the PCM sink in the MixerThread isValidPcmSinkChannelMask(audio_channel_mask_t channelMask)376 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 377 switch (audio_channel_mask_get_representation(channelMask)) { 378 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 379 uint32_t channelCount = FCC_2; // stereo is default 380 if (kEnableExtendedChannels) { 381 channelCount = audio_channel_count_from_out_mask(channelMask); 382 if (channelCount < FCC_2 // mono is not supported at this time 383 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 384 return false; 385 } 386 } 387 // check that channelMask is the "canonical" one we expect for the channelCount. 388 return channelMask == audio_channel_out_mask_from_count(channelCount); 389 } 390 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 391 if (kEnableExtendedChannels) { 392 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 393 if (channelCount >= FCC_2 // mono is not supported at this time 394 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 395 return true; 396 } 397 } 398 return false; 399 default: 400 return false; 401 } 402 } 403 404 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 405 static const bool kEnableExtendedPrecision = true; 406 407 // Returns true if format is permitted for the PCM sink in the MixerThread isValidPcmSinkFormat(audio_format_t format)408 static inline bool isValidPcmSinkFormat(audio_format_t format) { 409 switch (format) { 410 case AUDIO_FORMAT_PCM_16_BIT: 411 return true; 412 case AUDIO_FORMAT_PCM_FLOAT: 413 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 414 case AUDIO_FORMAT_PCM_32_BIT: 415 case AUDIO_FORMAT_PCM_8_24_BIT: 416 return kEnableExtendedPrecision; 417 default: 418 return false; 419 } 420 } 421 422 // standby delay for MIXER and DUPLICATING playback threads is read from property 423 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 424 static nsecs_t mStandbyTimeInNsecs; 425 426 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 427 // AudioFlinger::setParameters() updates, other threads read w/o lock 428 static uint32_t mScreenState; 429 430 // Internal dump utilities. 431 static const int kDumpLockRetries = 50; 432 static const int kDumpLockSleepUs = 20000; 433 static bool dumpTryLock(Mutex& mutex); 434 void dumpPermissionDenial(int fd, const Vector<String16>& args); 435 void dumpClients(int fd, const Vector<String16>& args); 436 void dumpInternals(int fd, const Vector<String16>& args); 437 438 // --- Client --- 439 class Client : public RefBase { 440 public: 441 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 442 virtual ~Client(); 443 sp<MemoryDealer> heap() const; pid()444 pid_t pid() const { return mPid; } audioFlinger()445 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 446 447 private: 448 Client(const Client&); 449 Client& operator = (const Client&); 450 const sp<AudioFlinger> mAudioFlinger; 451 sp<MemoryDealer> mMemoryDealer; 452 const pid_t mPid; 453 }; 454 455 // --- Notification Client --- 456 class NotificationClient : public IBinder::DeathRecipient { 457 public: 458 NotificationClient(const sp<AudioFlinger>& audioFlinger, 459 const sp<IAudioFlingerClient>& client, 460 pid_t pid); 461 virtual ~NotificationClient(); 462 audioFlingerClient()463 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 464 465 // IBinder::DeathRecipient 466 virtual void binderDied(const wp<IBinder>& who); 467 468 private: 469 NotificationClient(const NotificationClient&); 470 NotificationClient& operator = (const NotificationClient&); 471 472 const sp<AudioFlinger> mAudioFlinger; 473 const pid_t mPid; 474 const sp<IAudioFlingerClient> mAudioFlingerClient; 475 }; 476 477 // --- MediaLogNotifier --- 478 // Thread in charge of notifying MediaLogService to start merging. 479 // Receives requests from AudioFlinger's binder activity. It is used to reduce the amount of 480 // binder calls to MediaLogService in case of bursts of AudioFlinger binder calls. 481 class MediaLogNotifier : public Thread { 482 public: 483 MediaLogNotifier(); 484 485 // Requests a MediaLogService notification. It's ignored if there has recently been another 486 void requestMerge(); 487 private: 488 // Every iteration blocks waiting for a request, then interacts with MediaLogService to 489 // start merging. 490 // As every MediaLogService binder call is expensive, once it gets a request it ignores the 491 // following ones for a period of time. 492 virtual bool threadLoop() override; 493 494 bool mPendingRequests; 495 496 // Mutex and condition variable around mPendingRequests' value 497 Mutex mMutex; 498 Condition mCond; 499 500 // Duration of the sleep period after a processed request 501 static const int kPostTriggerSleepPeriod = 1000000; 502 }; 503 504 const sp<MediaLogNotifier> mMediaLogNotifier; 505 506 // This is a helper that is called during incoming binder calls. 507 void requestLogMerge(); 508 509 class TrackHandle; 510 class RecordHandle; 511 class RecordThread; 512 class PlaybackThread; 513 class MixerThread; 514 class DirectOutputThread; 515 class OffloadThread; 516 class DuplicatingThread; 517 class AsyncCallbackThread; 518 class Track; 519 class RecordTrack; 520 class EffectModule; 521 class EffectHandle; 522 class EffectChain; 523 524 struct AudioStreamIn; 525 526 struct stream_type_t { stream_type_tstream_type_t527 stream_type_t() 528 : volume(1.0f), 529 mute(false) 530 { 531 } 532 float volume; 533 bool mute; 534 }; 535 536 // --- PlaybackThread --- 537 538 #include "Threads.h" 539 540 #include "Effects.h" 541 542 #include "PatchPanel.h" 543 544 // server side of the client's IAudioTrack 545 class TrackHandle : public android::BnAudioTrack { 546 public: 547 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 548 virtual ~TrackHandle(); 549 virtual sp<IMemory> getCblk() const; 550 virtual status_t start(); 551 virtual void stop(); 552 virtual void flush(); 553 virtual void pause(); 554 virtual status_t attachAuxEffect(int effectId); 555 virtual status_t setParameters(const String8& keyValuePairs); 556 virtual VolumeShaper::Status applyVolumeShaper( 557 const sp<VolumeShaper::Configuration>& configuration, 558 const sp<VolumeShaper::Operation>& operation) override; 559 virtual sp<VolumeShaper::State> getVolumeShaperState(int id) override; 560 virtual status_t getTimestamp(AudioTimestamp& timestamp); 561 virtual void signal(); // signal playback thread for a change in control block 562 563 virtual status_t onTransact( 564 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 565 566 private: 567 const sp<PlaybackThread::Track> mTrack; 568 }; 569 570 // server side of the client's IAudioRecord 571 class RecordHandle : public android::BnAudioRecord { 572 public: 573 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 574 virtual ~RecordHandle(); 575 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, 576 audio_session_t triggerSession); 577 virtual void stop(); 578 virtual status_t onTransact( 579 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 580 private: 581 const sp<RecordThread::RecordTrack> mRecordTrack; 582 583 // for use from destructor 584 void stop_nonvirtual(); 585 }; 586 587 // Mmap stream control interface implementation. Each MmapThreadHandle controls one 588 // MmapPlaybackThread or MmapCaptureThread instance. 589 class MmapThreadHandle : public MmapStreamInterface { 590 public: 591 explicit MmapThreadHandle(const sp<MmapThread>& thread); 592 virtual ~MmapThreadHandle(); 593 594 // MmapStreamInterface virtuals 595 virtual status_t createMmapBuffer(int32_t minSizeFrames, 596 struct audio_mmap_buffer_info *info); 597 virtual status_t getMmapPosition(struct audio_mmap_position *position); 598 virtual status_t start(const MmapStreamInterface::Client& client, 599 audio_port_handle_t *handle); 600 virtual status_t stop(audio_port_handle_t handle); 601 virtual status_t standby(); 602 603 private: 604 sp<MmapThread> mThread; 605 }; 606 607 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 608 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 609 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 610 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 611 MmapThread *checkMmapThread_l(audio_io_handle_t io) const; 612 VolumeInterface *getVolumeInterface_l(audio_io_handle_t output) const; 613 Vector <VolumeInterface *> getAllVolumeInterfaces_l() const; 614 615 sp<ThreadBase> openInput_l(audio_module_handle_t module, 616 audio_io_handle_t *input, 617 audio_config_t *config, 618 audio_devices_t device, 619 const String8& address, 620 audio_source_t source, 621 audio_input_flags_t flags); 622 sp<ThreadBase> openOutput_l(audio_module_handle_t module, 623 audio_io_handle_t *output, 624 audio_config_t *config, 625 audio_devices_t devices, 626 const String8& address, 627 audio_output_flags_t flags); 628 629 void closeOutputFinish(const sp<PlaybackThread>& thread); 630 void closeInputFinish(const sp<RecordThread>& thread); 631 632 // no range check, AudioFlinger::mLock held streamMute_l(audio_stream_type_t stream)633 bool streamMute_l(audio_stream_type_t stream) const 634 { return mStreamTypes[stream].mute; } 635 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held streamVolume_l(audio_stream_type_t stream)636 float streamVolume_l(audio_stream_type_t stream) const 637 { return mStreamTypes[stream].volume; } 638 void ioConfigChanged(audio_io_config_event event, 639 const sp<AudioIoDescriptor>& ioDesc, 640 pid_t pid = 0); 641 642 // Allocate an audio_unique_id_t. 643 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 644 // audio_module_handle_t, and audio_patch_handle_t. 645 // They all share the same ID space, but the namespaces are actually independent 646 // because there are separate KeyedVectors for each kind of ID. 647 // The return value is cast to the specific type depending on how the ID will be used. 648 // FIXME This API does not handle rollover to zero (for unsigned IDs), 649 // or from positive to negative (for signed IDs). 650 // Thus it may fail by returning an ID of the wrong sign, 651 // or by returning a non-unique ID. 652 // This is the internal API. For the binder API see newAudioUniqueId(). 653 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 654 655 status_t moveEffectChain_l(audio_session_t sessionId, 656 PlaybackThread *srcThread, 657 PlaybackThread *dstThread, 658 bool reRegister); 659 660 // return thread associated with primary hardware device, or NULL 661 PlaybackThread *primaryPlaybackThread_l() const; 662 audio_devices_t primaryOutputDevice_l() const; 663 664 // return the playback thread with smallest HAL buffer size, and prefer fast 665 PlaybackThread *fastPlaybackThread_l() const; 666 667 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 668 669 670 void removeClient_l(pid_t pid); 671 void removeNotificationClient(pid_t pid); 672 bool isNonOffloadableGlobalEffectEnabled_l(); 673 void onNonOffloadableGlobalEffectEnable(); 674 bool isSessionAcquired_l(audio_session_t audioSession); 675 676 // Store an effect chain to mOrphanEffectChains keyed vector. 677 // Called when a thread exits and effects are still attached to it. 678 // If effects are later created on the same session, they will reuse the same 679 // effect chain and same instances in the effect library. 680 // return ALREADY_EXISTS if a chain with the same session already exists in 681 // mOrphanEffectChains. Note that this should never happen as there is only one 682 // chain for a given session and it is attached to only one thread at a time. 683 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 684 // Get an effect chain for the specified session in mOrphanEffectChains and remove 685 // it if found. Returns 0 if not found (this is the most common case). 686 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 687 // Called when the last effect handle on an effect instance is removed. If this 688 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 689 // and removed from mOrphanEffectChains if it does not contain any effect. 690 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 691 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 692 693 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 694 695 // AudioStreamIn is immutable, so their fields are const. 696 // For emphasis, we could also make all pointers to them be "const *", 697 // but that would clutter the code unnecessarily. 698 699 struct AudioStreamIn { 700 AudioHwDevice* const audioHwDev; 701 sp<StreamInHalInterface> stream; 702 audio_input_flags_t flags; 703 hwDevAudioStreamIn704 sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); } 705 AudioStreamInAudioStreamIn706 AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) : 707 audioHwDev(dev), stream(in), flags(flags) {} 708 }; 709 710 // for mAudioSessionRefs only 711 struct AudioSessionRef { AudioSessionRefAudioSessionRef712 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 713 mSessionid(sessionid), mPid(pid), mCnt(1) {} 714 const audio_session_t mSessionid; 715 const pid_t mPid; 716 int mCnt; 717 }; 718 719 mutable Mutex mLock; 720 // protects mClients and mNotificationClients. 721 // must be locked after mLock and ThreadBase::mLock if both must be locked 722 // avoids acquiring AudioFlinger::mLock from inside thread loop. 723 mutable Mutex mClientLock; 724 // protected by mClientLock 725 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 726 727 mutable Mutex mHardwareLock; 728 // NOTE: If both mLock and mHardwareLock mutexes must be held, 729 // always take mLock before mHardwareLock 730 731 // These two fields are immutable after onFirstRef(), so no lock needed to access 732 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 733 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 734 735 sp<DevicesFactoryHalInterface> mDevicesFactoryHal; 736 737 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 738 enum hardware_call_state { 739 AUDIO_HW_IDLE = 0, // no operation in progress 740 AUDIO_HW_INIT, // init_check 741 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 742 AUDIO_HW_OUTPUT_CLOSE, // unused 743 AUDIO_HW_INPUT_OPEN, // unused 744 AUDIO_HW_INPUT_CLOSE, // unused 745 AUDIO_HW_STANDBY, // unused 746 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 747 AUDIO_HW_GET_ROUTING, // unused 748 AUDIO_HW_SET_ROUTING, // unused 749 AUDIO_HW_GET_MODE, // unused 750 AUDIO_HW_SET_MODE, // set_mode 751 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 752 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 753 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 754 AUDIO_HW_SET_PARAMETER, // set_parameters 755 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 756 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 757 AUDIO_HW_GET_PARAMETER, // get_parameters 758 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 759 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 760 }; 761 762 mutable hardware_call_state mHardwareStatus; // for dump only 763 764 765 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 766 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 767 768 // member variables below are protected by mLock 769 float mMasterVolume; 770 bool mMasterMute; 771 // end of variables protected by mLock 772 773 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 774 775 // protected by mClientLock 776 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 777 778 // updated by atomic_fetch_add_explicit 779 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 780 781 audio_mode_t mMode; 782 bool mBtNrecIsOff; 783 784 // protected by mLock 785 Vector<AudioSessionRef*> mAudioSessionRefs; 786 787 float masterVolume_l() const; 788 bool masterMute_l() const; 789 audio_module_handle_t loadHwModule_l(const char *name); 790 791 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 792 // to be created 793 794 // Effect chains without a valid thread 795 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 796 797 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 798 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 799 800 // list of MMAP stream control threads. Those threads allow for wake lock, routing 801 // and volume control for activity on the associated MMAP stream at the HAL. 802 // Audio data transfer is directly handled by the client creating the MMAP stream 803 DefaultKeyedVector< audio_io_handle_t, sp<MmapThread> > mMmapThreads; 804 805 private: 806 sp<Client> registerPid(pid_t pid); // always returns non-0 807 808 // for use from destructor 809 status_t closeOutput_nonvirtual(audio_io_handle_t output); 810 void closeOutputInternal_l(const sp<PlaybackThread>& thread); 811 status_t closeInput_nonvirtual(audio_io_handle_t input); 812 void closeInputInternal_l(const sp<RecordThread>& thread); 813 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 814 815 status_t checkStreamType(audio_stream_type_t stream) const; 816 817 #ifdef TEE_SINK 818 // all record threads serially share a common tee sink, which is re-created on format change 819 sp<NBAIO_Sink> mRecordTeeSink; 820 sp<NBAIO_Source> mRecordTeeSource; 821 #endif 822 823 public: 824 825 #ifdef TEE_SINK 826 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 827 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 828 829 // whether tee sink is enabled by property 830 static bool mTeeSinkInputEnabled; 831 static bool mTeeSinkOutputEnabled; 832 static bool mTeeSinkTrackEnabled; 833 834 // runtime configured size of each tee sink pipe, in frames 835 static size_t mTeeSinkInputFrames; 836 static size_t mTeeSinkOutputFrames; 837 static size_t mTeeSinkTrackFrames; 838 839 // compile-time default size of tee sink pipes, in frames 840 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 841 static const size_t kTeeSinkInputFramesDefault = 0x200000; 842 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 843 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 844 #endif 845 846 // This method reads from a variable without mLock, but the variable is updated under mLock. So 847 // we might read a stale value, or a value that's inconsistent with respect to other variables. 848 // In this case, it's safe because the return value isn't used for making an important decision. 849 // The reason we don't want to take mLock is because it could block the caller for a long time. isLowRamDevice()850 bool isLowRamDevice() const { return mIsLowRamDevice; } 851 852 private: 853 bool mIsLowRamDevice; 854 bool mIsDeviceTypeKnown; 855 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 856 857 sp<PatchPanel> mPatchPanel; 858 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 859 860 bool mSystemReady; 861 }; 862 863 #undef INCLUDING_FROM_AUDIOFLINGER_H 864 865 std::string formatToString(audio_format_t format); 866 std::string inputFlagsToString(audio_input_flags_t flags); 867 std::string outputFlagsToString(audio_output_flags_t flags); 868 std::string devicesToString(audio_devices_t devices); 869 const char *sourceToString(audio_source_t source); 870 871 // ---------------------------------------------------------------------------- 872 873 } // namespace android 874 875 #endif // ANDROID_AUDIO_FLINGER_H 876