AudioFlinger.h revision d3bb645f0b7f567b033b8664499d685f8ec10628
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 "FastCapture.h" 62#include "FastMixer.h" 63#include <media/nbaio/NBAIO.h> 64#include "AudioWatchdog.h" 65#include "AudioStreamOut.h" 66#include "SpdifStreamOut.h" 67#include "AudioHwDevice.h" 68 69#include <powermanager/IPowerManager.h> 70 71#include <media/nbaio/NBLog.h> 72#include <private/media/AudioTrackShared.h> 73 74namespace android { 75 76struct audio_track_cblk_t; 77struct effect_param_cblk_t; 78class AudioMixer; 79class AudioBuffer; 80class AudioResampler; 81class DeviceHalInterface; 82class DevicesFactoryHalInterface; 83class EffectsFactoryHalInterface; 84class FastMixer; 85class PassthruBufferProvider; 86class RecordBufferConverter; 87class ServerProxy; 88 89// ---------------------------------------------------------------------------- 90 91static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 92 93 94// Max shared memory size for audio tracks and audio records per client process 95static const size_t kClientSharedHeapSizeBytes = 1024*1024; 96// Shared memory size multiplier for non low ram devices 97static const size_t kClientSharedHeapSizeMultiplier = 4; 98 99#define INCLUDING_FROM_AUDIOFLINGER_H 100 101class AudioFlinger : 102 public BinderService<AudioFlinger>, 103 public BnAudioFlinger 104{ 105 friend class BinderService<AudioFlinger>; // for AudioFlinger() 106 107public: 108 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 109 110 virtual status_t dump(int fd, const Vector<String16>& args); 111 112 // IAudioFlinger interface, in binder opcode order 113 virtual sp<IAudioTrack> createTrack( 114 audio_stream_type_t streamType, 115 uint32_t sampleRate, 116 audio_format_t format, 117 audio_channel_mask_t channelMask, 118 size_t *pFrameCount, 119 audio_output_flags_t *flags, 120 const sp<IMemory>& sharedBuffer, 121 audio_io_handle_t output, 122 pid_t pid, 123 pid_t tid, 124 audio_session_t *sessionId, 125 int clientUid, 126 status_t *status /*non-NULL*/, 127 audio_port_handle_t portId); 128 129 virtual sp<IAudioRecord> openRecord( 130 audio_io_handle_t input, 131 uint32_t sampleRate, 132 audio_format_t format, 133 audio_channel_mask_t channelMask, 134 const String16& opPackageName, 135 size_t *pFrameCount, 136 audio_input_flags_t *flags, 137 pid_t pid, 138 pid_t tid, 139 int clientUid, 140 audio_session_t *sessionId, 141 size_t *notificationFrames, 142 sp<IMemory>& cblk, 143 sp<IMemory>& buffers, 144 status_t *status /*non-NULL*/, 145 audio_port_handle_t portId); 146 147 virtual uint32_t sampleRate(audio_io_handle_t ioHandle) const; 148 virtual audio_format_t format(audio_io_handle_t output) const; 149 virtual size_t frameCount(audio_io_handle_t ioHandle) const; 150 virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const; 151 virtual uint32_t latency(audio_io_handle_t output) const; 152 153 virtual status_t setMasterVolume(float value); 154 virtual status_t setMasterMute(bool muted); 155 156 virtual float masterVolume() const; 157 virtual bool masterMute() const; 158 159 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 160 audio_io_handle_t output); 161 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 162 163 virtual float streamVolume(audio_stream_type_t stream, 164 audio_io_handle_t output) const; 165 virtual bool streamMute(audio_stream_type_t stream) const; 166 167 virtual status_t setMode(audio_mode_t mode); 168 169 virtual status_t setMicMute(bool state); 170 virtual bool getMicMute() const; 171 172 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 173 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 174 175 virtual void registerClient(const sp<IAudioFlingerClient>& client); 176 177 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 178 audio_channel_mask_t channelMask) const; 179 180 virtual status_t openOutput(audio_module_handle_t module, 181 audio_io_handle_t *output, 182 audio_config_t *config, 183 audio_devices_t *devices, 184 const String8& address, 185 uint32_t *latencyMs, 186 audio_output_flags_t flags); 187 188 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 189 audio_io_handle_t output2); 190 191 virtual status_t closeOutput(audio_io_handle_t output); 192 193 virtual status_t suspendOutput(audio_io_handle_t output); 194 195 virtual status_t restoreOutput(audio_io_handle_t output); 196 197 virtual status_t openInput(audio_module_handle_t module, 198 audio_io_handle_t *input, 199 audio_config_t *config, 200 audio_devices_t *device, 201 const String8& address, 202 audio_source_t source, 203 audio_input_flags_t flags); 204 205 virtual status_t closeInput(audio_io_handle_t input); 206 207 virtual status_t invalidateStream(audio_stream_type_t stream); 208 209 virtual status_t setVoiceVolume(float volume); 210 211 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 212 audio_io_handle_t output) const; 213 214 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 215 216 // This is the binder API. For the internal API see nextUniqueId(). 217 virtual audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 218 219 virtual void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 220 221 virtual void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 222 223 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 224 225 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 226 227 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 228 effect_descriptor_t *descriptor) const; 229 230 virtual sp<IEffect> createEffect( 231 effect_descriptor_t *pDesc, 232 const sp<IEffectClient>& effectClient, 233 int32_t priority, 234 audio_io_handle_t io, 235 audio_session_t sessionId, 236 const String16& opPackageName, 237 pid_t pid, 238 status_t *status /*non-NULL*/, 239 int *id, 240 int *enabled); 241 242 virtual status_t moveEffects(audio_session_t sessionId, audio_io_handle_t srcOutput, 243 audio_io_handle_t dstOutput); 244 245 virtual audio_module_handle_t loadHwModule(const char *name); 246 247 virtual uint32_t getPrimaryOutputSamplingRate(); 248 virtual size_t getPrimaryOutputFrameCount(); 249 250 virtual status_t setLowRamDevice(bool isLowRamDevice); 251 252 /* List available audio ports and their attributes */ 253 virtual status_t listAudioPorts(unsigned int *num_ports, 254 struct audio_port *ports); 255 256 /* Get attributes for a given audio port */ 257 virtual status_t getAudioPort(struct audio_port *port); 258 259 /* Create an audio patch between several source and sink ports */ 260 virtual status_t createAudioPatch(const struct audio_patch *patch, 261 audio_patch_handle_t *handle); 262 263 /* Release an audio patch */ 264 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 265 266 /* List existing audio patches */ 267 virtual status_t listAudioPatches(unsigned int *num_patches, 268 struct audio_patch *patches); 269 270 /* Set audio port configuration */ 271 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 272 273 /* Get the HW synchronization source used for an audio session */ 274 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 275 276 /* Indicate JAVA services are ready (scheduling, power management ...) */ 277 virtual status_t systemReady(); 278 279 virtual status_t onTransact( 280 uint32_t code, 281 const Parcel& data, 282 Parcel* reply, 283 uint32_t flags); 284 285 // end of IAudioFlinger interface 286 287 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 288 void unregisterWriter(const sp<NBLog::Writer>& writer); 289 sp<EffectsFactoryHalInterface> getEffectsFactory(); 290 291 status_t openMmapStream(MmapStreamInterface::stream_direction_t direction, 292 const audio_attributes_t *attr, 293 audio_config_base_t *config, 294 const MmapStreamInterface::Client& client, 295 audio_port_handle_t *deviceId, 296 const sp<MmapStreamCallback>& callback, 297 sp<MmapStreamInterface>& interface); 298private: 299 static const size_t kLogMemorySize = 40 * 1024; 300 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 301 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 302 // for as long as possible. The memory is only freed when it is needed for another log writer. 303 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 304 Mutex mUnregisteredWritersLock; 305 306public: 307 308 class SyncEvent; 309 310 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 311 312 class SyncEvent : public RefBase { 313 public: 314 SyncEvent(AudioSystem::sync_event_t type, 315 audio_session_t triggerSession, 316 audio_session_t listenerSession, 317 sync_event_callback_t callBack, 318 wp<RefBase> cookie) 319 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 320 mCallback(callBack), mCookie(cookie) 321 {} 322 323 virtual ~SyncEvent() {} 324 325 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 326 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 327 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 328 AudioSystem::sync_event_t type() const { return mType; } 329 audio_session_t triggerSession() const { return mTriggerSession; } 330 audio_session_t listenerSession() const { return mListenerSession; } 331 wp<RefBase> cookie() const { return mCookie; } 332 333 private: 334 const AudioSystem::sync_event_t mType; 335 const audio_session_t mTriggerSession; 336 const audio_session_t mListenerSession; 337 sync_event_callback_t mCallback; 338 const wp<RefBase> mCookie; 339 mutable Mutex mLock; 340 }; 341 342 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 343 audio_session_t triggerSession, 344 audio_session_t listenerSession, 345 sync_event_callback_t callBack, 346 const wp<RefBase>& cookie); 347 348private: 349 350 audio_mode_t getMode() const { return mMode; } 351 352 bool btNrecIsOff() const { return mBtNrecIsOff; } 353 354 AudioFlinger() ANDROID_API; 355 virtual ~AudioFlinger(); 356 357 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 358 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 359 NO_INIT : NO_ERROR; } 360 361 // RefBase 362 virtual void onFirstRef(); 363 364 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 365 audio_devices_t devices); 366 void purgeStaleEffects_l(); 367 368 // Set kEnableExtendedChannels to true to enable greater than stereo output 369 // for the MixerThread and device sink. Number of channels allowed is 370 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 371 static const bool kEnableExtendedChannels = true; 372 373 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 374 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 375 switch (audio_channel_mask_get_representation(channelMask)) { 376 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 377 uint32_t channelCount = FCC_2; // stereo is default 378 if (kEnableExtendedChannels) { 379 channelCount = audio_channel_count_from_out_mask(channelMask); 380 if (channelCount < FCC_2 // mono is not supported at this time 381 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 382 return false; 383 } 384 } 385 // check that channelMask is the "canonical" one we expect for the channelCount. 386 return channelMask == audio_channel_out_mask_from_count(channelCount); 387 } 388 case AUDIO_CHANNEL_REPRESENTATION_INDEX: 389 if (kEnableExtendedChannels) { 390 const uint32_t channelCount = audio_channel_count_from_out_mask(channelMask); 391 if (channelCount >= FCC_2 // mono is not supported at this time 392 && channelCount <= AudioMixer::MAX_NUM_CHANNELS) { 393 return true; 394 } 395 } 396 return false; 397 default: 398 return false; 399 } 400 } 401 402 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 403 static const bool kEnableExtendedPrecision = true; 404 405 // Returns true if format is permitted for the PCM sink in the MixerThread 406 static inline bool isValidPcmSinkFormat(audio_format_t format) { 407 switch (format) { 408 case AUDIO_FORMAT_PCM_16_BIT: 409 return true; 410 case AUDIO_FORMAT_PCM_FLOAT: 411 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 412 case AUDIO_FORMAT_PCM_32_BIT: 413 case AUDIO_FORMAT_PCM_8_24_BIT: 414 return kEnableExtendedPrecision; 415 default: 416 return false; 417 } 418 } 419 420 // standby delay for MIXER and DUPLICATING playback threads is read from property 421 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 422 static nsecs_t mStandbyTimeInNsecs; 423 424 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 425 // AudioFlinger::setParameters() updates, other threads read w/o lock 426 static uint32_t mScreenState; 427 428 // Internal dump utilities. 429 static const int kDumpLockRetries = 50; 430 static const int kDumpLockSleepUs = 20000; 431 static bool dumpTryLock(Mutex& mutex); 432 void dumpPermissionDenial(int fd, const Vector<String16>& args); 433 void dumpClients(int fd, const Vector<String16>& args); 434 void dumpInternals(int fd, const Vector<String16>& args); 435 436 // --- Client --- 437 class Client : public RefBase { 438 public: 439 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 440 virtual ~Client(); 441 sp<MemoryDealer> heap() const; 442 pid_t pid() const { return mPid; } 443 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 444 445 private: 446 Client(const Client&); 447 Client& operator = (const Client&); 448 const sp<AudioFlinger> mAudioFlinger; 449 sp<MemoryDealer> mMemoryDealer; 450 const pid_t mPid; 451 }; 452 453 // --- Notification Client --- 454 class NotificationClient : public IBinder::DeathRecipient { 455 public: 456 NotificationClient(const sp<AudioFlinger>& audioFlinger, 457 const sp<IAudioFlingerClient>& client, 458 pid_t pid); 459 virtual ~NotificationClient(); 460 461 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 462 463 // IBinder::DeathRecipient 464 virtual void binderDied(const wp<IBinder>& who); 465 466 private: 467 NotificationClient(const NotificationClient&); 468 NotificationClient& operator = (const NotificationClient&); 469 470 const sp<AudioFlinger> mAudioFlinger; 471 const pid_t mPid; 472 const sp<IAudioFlingerClient> mAudioFlingerClient; 473 }; 474 475 class TrackHandle; 476 class RecordHandle; 477 class RecordThread; 478 class PlaybackThread; 479 class MixerThread; 480 class DirectOutputThread; 481 class OffloadThread; 482 class DuplicatingThread; 483 class AsyncCallbackThread; 484 class Track; 485 class RecordTrack; 486 class EffectModule; 487 class EffectHandle; 488 class EffectChain; 489 490 struct AudioStreamIn; 491 492 struct stream_type_t { 493 stream_type_t() 494 : volume(1.0f), 495 mute(false) 496 { 497 } 498 float volume; 499 bool mute; 500 }; 501 502 // --- PlaybackThread --- 503 504#include "Threads.h" 505 506#include "Effects.h" 507 508#include "PatchPanel.h" 509 510 // server side of the client's IAudioTrack 511 class TrackHandle : public android::BnAudioTrack { 512 public: 513 explicit TrackHandle(const sp<PlaybackThread::Track>& track); 514 virtual ~TrackHandle(); 515 virtual sp<IMemory> getCblk() const; 516 virtual status_t start(); 517 virtual void stop(); 518 virtual void flush(); 519 virtual void pause(); 520 virtual status_t attachAuxEffect(int effectId); 521 virtual status_t setParameters(const String8& keyValuePairs); 522 virtual VolumeShaper::Status applyVolumeShaper( 523 const sp<VolumeShaper::Configuration>& configuration, 524 const sp<VolumeShaper::Operation>& operation) override; 525 virtual sp<VolumeShaper::State> getVolumeShaperState(int id) override; 526 virtual status_t getTimestamp(AudioTimestamp& timestamp); 527 virtual void signal(); // signal playback thread for a change in control block 528 529 virtual status_t onTransact( 530 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 531 532 private: 533 const sp<PlaybackThread::Track> mTrack; 534 }; 535 536 // server side of the client's IAudioRecord 537 class RecordHandle : public android::BnAudioRecord { 538 public: 539 explicit RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 540 virtual ~RecordHandle(); 541 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, 542 audio_session_t triggerSession); 543 virtual void stop(); 544 virtual status_t onTransact( 545 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 546 private: 547 const sp<RecordThread::RecordTrack> mRecordTrack; 548 549 // for use from destructor 550 void stop_nonvirtual(); 551 }; 552 553 // Mmap stream control interface implementation. Each MmapThreadHandle controls one 554 // MmapPlaybackThread or MmapCaptureThread instance. 555 class MmapThreadHandle : public MmapStreamInterface { 556 public: 557 explicit MmapThreadHandle(const sp<MmapThread>& thread); 558 virtual ~MmapThreadHandle(); 559 560 // MmapStreamInterface virtuals 561 virtual status_t createMmapBuffer(int32_t minSizeFrames, 562 struct audio_mmap_buffer_info *info); 563 virtual status_t getMmapPosition(struct audio_mmap_position *position); 564 virtual status_t start(const MmapStreamInterface::Client& client, 565 audio_port_handle_t *handle); 566 virtual status_t stop(audio_port_handle_t handle); 567 568 private: 569 sp<MmapThread> mThread; 570 }; 571 572 ThreadBase *checkThread_l(audio_io_handle_t ioHandle) const; 573 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 574 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 575 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 576 MmapThread *checkMmapThread_l(audio_io_handle_t io) const; 577 VolumeInterface *getVolumeInterface_l(audio_io_handle_t output) const; 578 Vector <VolumeInterface *> getAllVolumeInterfaces_l() const; 579 580 sp<ThreadBase> openInput_l(audio_module_handle_t module, 581 audio_io_handle_t *input, 582 audio_config_t *config, 583 audio_devices_t device, 584 const String8& address, 585 audio_source_t source, 586 audio_input_flags_t flags); 587 sp<ThreadBase> openOutput_l(audio_module_handle_t module, 588 audio_io_handle_t *output, 589 audio_config_t *config, 590 audio_devices_t devices, 591 const String8& address, 592 audio_output_flags_t flags); 593 594 void closeOutputFinish(const sp<PlaybackThread>& thread); 595 void closeInputFinish(const sp<RecordThread>& thread); 596 597 // no range check, AudioFlinger::mLock held 598 bool streamMute_l(audio_stream_type_t stream) const 599 { return mStreamTypes[stream].mute; } 600 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 601 float streamVolume_l(audio_stream_type_t stream) const 602 { return mStreamTypes[stream].volume; } 603 void ioConfigChanged(audio_io_config_event event, 604 const sp<AudioIoDescriptor>& ioDesc, 605 pid_t pid = 0); 606 607 // Allocate an audio_unique_id_t. 608 // Specific types are audio_io_handle_t, audio_session_t, effect ID (int), 609 // audio_module_handle_t, and audio_patch_handle_t. 610 // They all share the same ID space, but the namespaces are actually independent 611 // because there are separate KeyedVectors for each kind of ID. 612 // The return value is cast to the specific type depending on how the ID will be used. 613 // FIXME This API does not handle rollover to zero (for unsigned IDs), 614 // or from positive to negative (for signed IDs). 615 // Thus it may fail by returning an ID of the wrong sign, 616 // or by returning a non-unique ID. 617 // This is the internal API. For the binder API see newAudioUniqueId(). 618 audio_unique_id_t nextUniqueId(audio_unique_id_use_t use); 619 620 status_t moveEffectChain_l(audio_session_t sessionId, 621 PlaybackThread *srcThread, 622 PlaybackThread *dstThread, 623 bool reRegister); 624 625 // return thread associated with primary hardware device, or NULL 626 PlaybackThread *primaryPlaybackThread_l() const; 627 audio_devices_t primaryOutputDevice_l() const; 628 629 // return the playback thread with smallest HAL buffer size, and prefer fast 630 PlaybackThread *fastPlaybackThread_l() const; 631 632 sp<PlaybackThread> getEffectThread_l(audio_session_t sessionId, int EffectId); 633 634 635 void removeClient_l(pid_t pid); 636 void removeNotificationClient(pid_t pid); 637 bool isNonOffloadableGlobalEffectEnabled_l(); 638 void onNonOffloadableGlobalEffectEnable(); 639 bool isSessionAcquired_l(audio_session_t audioSession); 640 641 // Store an effect chain to mOrphanEffectChains keyed vector. 642 // Called when a thread exits and effects are still attached to it. 643 // If effects are later created on the same session, they will reuse the same 644 // effect chain and same instances in the effect library. 645 // return ALREADY_EXISTS if a chain with the same session already exists in 646 // mOrphanEffectChains. Note that this should never happen as there is only one 647 // chain for a given session and it is attached to only one thread at a time. 648 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 649 // Get an effect chain for the specified session in mOrphanEffectChains and remove 650 // it if found. Returns 0 if not found (this is the most common case). 651 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 652 // Called when the last effect handle on an effect instance is removed. If this 653 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 654 // and removed from mOrphanEffectChains if it does not contain any effect. 655 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 656 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 657 658 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 659 660 // AudioStreamIn is immutable, so their fields are const. 661 // For emphasis, we could also make all pointers to them be "const *", 662 // but that would clutter the code unnecessarily. 663 664 struct AudioStreamIn { 665 AudioHwDevice* const audioHwDev; 666 sp<StreamInHalInterface> stream; 667 audio_input_flags_t flags; 668 669 sp<DeviceHalInterface> hwDev() const { return audioHwDev->hwDevice(); } 670 671 AudioStreamIn(AudioHwDevice *dev, sp<StreamInHalInterface> in, audio_input_flags_t flags) : 672 audioHwDev(dev), stream(in), flags(flags) {} 673 }; 674 675 // for mAudioSessionRefs only 676 struct AudioSessionRef { 677 AudioSessionRef(audio_session_t sessionid, pid_t pid) : 678 mSessionid(sessionid), mPid(pid), mCnt(1) {} 679 const audio_session_t mSessionid; 680 const pid_t mPid; 681 int mCnt; 682 }; 683 684 mutable Mutex mLock; 685 // protects mClients and mNotificationClients. 686 // must be locked after mLock and ThreadBase::mLock if both must be locked 687 // avoids acquiring AudioFlinger::mLock from inside thread loop. 688 mutable Mutex mClientLock; 689 // protected by mClientLock 690 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 691 692 mutable Mutex mHardwareLock; 693 // NOTE: If both mLock and mHardwareLock mutexes must be held, 694 // always take mLock before mHardwareLock 695 696 // These two fields are immutable after onFirstRef(), so no lock needed to access 697 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 698 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 699 700 sp<DevicesFactoryHalInterface> mDevicesFactoryHal; 701 702 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 703 enum hardware_call_state { 704 AUDIO_HW_IDLE = 0, // no operation in progress 705 AUDIO_HW_INIT, // init_check 706 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 707 AUDIO_HW_OUTPUT_CLOSE, // unused 708 AUDIO_HW_INPUT_OPEN, // unused 709 AUDIO_HW_INPUT_CLOSE, // unused 710 AUDIO_HW_STANDBY, // unused 711 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 712 AUDIO_HW_GET_ROUTING, // unused 713 AUDIO_HW_SET_ROUTING, // unused 714 AUDIO_HW_GET_MODE, // unused 715 AUDIO_HW_SET_MODE, // set_mode 716 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 717 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 718 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 719 AUDIO_HW_SET_PARAMETER, // set_parameters 720 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 721 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 722 AUDIO_HW_GET_PARAMETER, // get_parameters 723 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 724 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 725 }; 726 727 mutable hardware_call_state mHardwareStatus; // for dump only 728 729 730 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 731 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 732 733 // member variables below are protected by mLock 734 float mMasterVolume; 735 bool mMasterMute; 736 // end of variables protected by mLock 737 738 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 739 740 // protected by mClientLock 741 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 742 743 // updated by atomic_fetch_add_explicit 744 volatile atomic_uint_fast32_t mNextUniqueIds[AUDIO_UNIQUE_ID_USE_MAX]; 745 746 audio_mode_t mMode; 747 bool mBtNrecIsOff; 748 749 // protected by mLock 750 Vector<AudioSessionRef*> mAudioSessionRefs; 751 752 float masterVolume_l() const; 753 bool masterMute_l() const; 754 audio_module_handle_t loadHwModule_l(const char *name); 755 756 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 757 // to be created 758 759 // Effect chains without a valid thread 760 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 761 762 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 763 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 764 765 // list of MMAP stream control threads. Those threads allow for wake lock, routing 766 // and volume control for activity on the associated MMAP stream at the HAL. 767 // Audio data transfer is directly handled by the client creating the MMAP stream 768 DefaultKeyedVector< audio_io_handle_t, sp<MmapThread> > mMmapThreads; 769 770private: 771 sp<Client> registerPid(pid_t pid); // always returns non-0 772 773 // for use from destructor 774 status_t closeOutput_nonvirtual(audio_io_handle_t output); 775 void closeOutputInternal_l(const sp<PlaybackThread>& thread); 776 status_t closeInput_nonvirtual(audio_io_handle_t input); 777 void closeInputInternal_l(const sp<RecordThread>& thread); 778 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 779 780 status_t checkStreamType(audio_stream_type_t stream) const; 781 782#ifdef TEE_SINK 783 // all record threads serially share a common tee sink, which is re-created on format change 784 sp<NBAIO_Sink> mRecordTeeSink; 785 sp<NBAIO_Source> mRecordTeeSource; 786#endif 787 788public: 789 790#ifdef TEE_SINK 791 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 792 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 793 794 // whether tee sink is enabled by property 795 static bool mTeeSinkInputEnabled; 796 static bool mTeeSinkOutputEnabled; 797 static bool mTeeSinkTrackEnabled; 798 799 // runtime configured size of each tee sink pipe, in frames 800 static size_t mTeeSinkInputFrames; 801 static size_t mTeeSinkOutputFrames; 802 static size_t mTeeSinkTrackFrames; 803 804 // compile-time default size of tee sink pipes, in frames 805 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 806 static const size_t kTeeSinkInputFramesDefault = 0x200000; 807 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 808 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 809#endif 810 811 // This method reads from a variable without mLock, but the variable is updated under mLock. So 812 // we might read a stale value, or a value that's inconsistent with respect to other variables. 813 // In this case, it's safe because the return value isn't used for making an important decision. 814 // The reason we don't want to take mLock is because it could block the caller for a long time. 815 bool isLowRamDevice() const { return mIsLowRamDevice; } 816 817private: 818 bool mIsLowRamDevice; 819 bool mIsDeviceTypeKnown; 820 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 821 822 sp<PatchPanel> mPatchPanel; 823 sp<EffectsFactoryHalInterface> mEffectsFactoryHal; 824 825 bool mSystemReady; 826}; 827 828#undef INCLUDING_FROM_AUDIOFLINGER_H 829 830std::string formatToString(audio_format_t format); 831std::string inputFlagsToString(audio_input_flags_t flags); 832std::string outputFlagsToString(audio_output_flags_t flags); 833std::string devicesToString(audio_devices_t devices); 834const char *sourceToString(audio_source_t source); 835 836// ---------------------------------------------------------------------------- 837 838} // namespace android 839 840#endif // ANDROID_AUDIO_FLINGER_H 841