AudioFlinger.h revision cc85abcf4ac398dca240db356b8b4db052b415a4
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 <stdint.h> 23#include <sys/types.h> 24#include <limits.h> 25 26#include <common_time/cc_helper.h> 27 28#include <cutils/compiler.h> 29 30#include <media/IAudioFlinger.h> 31#include <media/IAudioFlingerClient.h> 32#include <media/IAudioTrack.h> 33#include <media/IAudioRecord.h> 34#include <media/AudioSystem.h> 35#include <media/AudioTrack.h> 36 37#include <utils/Atomic.h> 38#include <utils/Errors.h> 39#include <utils/threads.h> 40#include <utils/SortedVector.h> 41#include <utils/TypeHelpers.h> 42#include <utils/Vector.h> 43 44#include <binder/BinderService.h> 45#include <binder/MemoryDealer.h> 46 47#include <system/audio.h> 48#include <hardware/audio.h> 49#include <hardware/audio_policy.h> 50 51#include <media/AudioBufferProvider.h> 52#include <media/ExtendedAudioBufferProvider.h> 53 54#include "FastCapture.h" 55#include "FastMixer.h" 56#include <media/nbaio/NBAIO.h> 57#include "AudioWatchdog.h" 58#include "AudioMixer.h" 59#include "AudioStreamOut.h" 60#include "SpdifStreamOut.h" 61#include "AudioHwDevice.h" 62 63#include <powermanager/IPowerManager.h> 64 65#include <media/nbaio/NBLog.h> 66#include <private/media/AudioTrackShared.h> 67 68namespace android { 69 70struct audio_track_cblk_t; 71struct effect_param_cblk_t; 72class AudioMixer; 73class AudioBuffer; 74class AudioResampler; 75class FastMixer; 76class PassthruBufferProvider; 77class ServerProxy; 78 79// ---------------------------------------------------------------------------- 80 81// The macro FCC_2 highlights some (but not all) places where there are are 2-channel assumptions. 82// This is typically due to legacy implementation of stereo input or output. 83// Search also for "2", "left", "right", "[0]", "[1]", ">> 16", "<< 16", etc. 84#define FCC_2 2 // FCC_2 = Fixed Channel Count 2 85// The macro FCC_8 highlights places where there are 8-channel assumptions. 86// This is typically due to audio mixer and resampler limitations. 87#define FCC_8 8 // FCC_8 = Fixed Channel Count 8 88 89static const nsecs_t kDefaultStandbyTimeInNsecs = seconds(3); 90 91#define INCLUDING_FROM_AUDIOFLINGER_H 92 93class AudioFlinger : 94 public BinderService<AudioFlinger>, 95 public BnAudioFlinger 96{ 97 friend class BinderService<AudioFlinger>; // for AudioFlinger() 98public: 99 static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; } 100 101 virtual status_t dump(int fd, const Vector<String16>& args); 102 103 // IAudioFlinger interface, in binder opcode order 104 virtual sp<IAudioTrack> createTrack( 105 audio_stream_type_t streamType, 106 uint32_t sampleRate, 107 audio_format_t format, 108 audio_channel_mask_t channelMask, 109 size_t *pFrameCount, 110 IAudioFlinger::track_flags_t *flags, 111 const sp<IMemory>& sharedBuffer, 112 audio_io_handle_t output, 113 pid_t tid, 114 int *sessionId, 115 int clientUid, 116 status_t *status /*non-NULL*/); 117 118 virtual sp<IAudioRecord> openRecord( 119 audio_io_handle_t input, 120 uint32_t sampleRate, 121 audio_format_t format, 122 audio_channel_mask_t channelMask, 123 const String16& opPackageName, 124 size_t *pFrameCount, 125 IAudioFlinger::track_flags_t *flags, 126 pid_t tid, 127 int *sessionId, 128 size_t *notificationFrames, 129 sp<IMemory>& cblk, 130 sp<IMemory>& buffers, 131 status_t *status /*non-NULL*/); 132 133 virtual uint32_t sampleRate(audio_io_handle_t output) const; 134 virtual audio_format_t format(audio_io_handle_t output) const; 135 virtual size_t frameCount(audio_io_handle_t output) const; 136 virtual uint32_t latency(audio_io_handle_t output) const; 137 138 virtual status_t setMasterVolume(float value); 139 virtual status_t setMasterMute(bool muted); 140 141 virtual float masterVolume() const; 142 virtual bool masterMute() const; 143 144 virtual status_t setStreamVolume(audio_stream_type_t stream, float value, 145 audio_io_handle_t output); 146 virtual status_t setStreamMute(audio_stream_type_t stream, bool muted); 147 148 virtual float streamVolume(audio_stream_type_t stream, 149 audio_io_handle_t output) const; 150 virtual bool streamMute(audio_stream_type_t stream) const; 151 152 virtual status_t setMode(audio_mode_t mode); 153 154 virtual status_t setMicMute(bool state); 155 virtual bool getMicMute() const; 156 157 virtual status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 158 virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) const; 159 160 virtual void registerClient(const sp<IAudioFlingerClient>& client); 161 162 virtual size_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 163 audio_channel_mask_t channelMask) const; 164 165 virtual status_t openOutput(audio_module_handle_t module, 166 audio_io_handle_t *output, 167 audio_config_t *config, 168 audio_devices_t *devices, 169 const String8& address, 170 uint32_t *latencyMs, 171 audio_output_flags_t flags); 172 173 virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, 174 audio_io_handle_t output2); 175 176 virtual status_t closeOutput(audio_io_handle_t output); 177 178 virtual status_t suspendOutput(audio_io_handle_t output); 179 180 virtual status_t restoreOutput(audio_io_handle_t output); 181 182 virtual status_t openInput(audio_module_handle_t module, 183 audio_io_handle_t *input, 184 audio_config_t *config, 185 audio_devices_t *device, 186 const String8& address, 187 audio_source_t source, 188 audio_input_flags_t flags); 189 190 virtual status_t closeInput(audio_io_handle_t input); 191 192 virtual status_t invalidateStream(audio_stream_type_t stream); 193 194 virtual status_t setVoiceVolume(float volume); 195 196 virtual status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, 197 audio_io_handle_t output) const; 198 199 virtual uint32_t getInputFramesLost(audio_io_handle_t ioHandle) const; 200 201 virtual audio_unique_id_t newAudioUniqueId(); 202 203 virtual void acquireAudioSessionId(int audioSession, pid_t pid); 204 205 virtual void releaseAudioSessionId(int audioSession, pid_t pid); 206 207 virtual status_t queryNumberEffects(uint32_t *numEffects) const; 208 209 virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const; 210 211 virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid, 212 effect_descriptor_t *descriptor) const; 213 214 virtual sp<IEffect> createEffect( 215 effect_descriptor_t *pDesc, 216 const sp<IEffectClient>& effectClient, 217 int32_t priority, 218 audio_io_handle_t io, 219 int sessionId, 220 const String16& opPackageName, 221 status_t *status /*non-NULL*/, 222 int *id, 223 int *enabled); 224 225 virtual status_t moveEffects(int sessionId, audio_io_handle_t srcOutput, 226 audio_io_handle_t dstOutput); 227 228 virtual audio_module_handle_t loadHwModule(const char *name); 229 230 virtual uint32_t getPrimaryOutputSamplingRate(); 231 virtual size_t getPrimaryOutputFrameCount(); 232 233 virtual status_t setLowRamDevice(bool isLowRamDevice); 234 235 /* List available audio ports and their attributes */ 236 virtual status_t listAudioPorts(unsigned int *num_ports, 237 struct audio_port *ports); 238 239 /* Get attributes for a given audio port */ 240 virtual status_t getAudioPort(struct audio_port *port); 241 242 /* Create an audio patch between several source and sink ports */ 243 virtual status_t createAudioPatch(const struct audio_patch *patch, 244 audio_patch_handle_t *handle); 245 246 /* Release an audio patch */ 247 virtual status_t releaseAudioPatch(audio_patch_handle_t handle); 248 249 /* List existing audio patches */ 250 virtual status_t listAudioPatches(unsigned int *num_patches, 251 struct audio_patch *patches); 252 253 /* Set audio port configuration */ 254 virtual status_t setAudioPortConfig(const struct audio_port_config *config); 255 256 /* Get the HW synchronization source used for an audio session */ 257 virtual audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 258 259 virtual status_t onTransact( 260 uint32_t code, 261 const Parcel& data, 262 Parcel* reply, 263 uint32_t flags); 264 265 // end of IAudioFlinger interface 266 267 sp<NBLog::Writer> newWriter_l(size_t size, const char *name); 268 void unregisterWriter(const sp<NBLog::Writer>& writer); 269private: 270 static const size_t kLogMemorySize = 40 * 1024; 271 sp<MemoryDealer> mLogMemoryDealer; // == 0 when NBLog is disabled 272 // When a log writer is unregistered, it is done lazily so that media.log can continue to see it 273 // for as long as possible. The memory is only freed when it is needed for another log writer. 274 Vector< sp<NBLog::Writer> > mUnregisteredWriters; 275 Mutex mUnregisteredWritersLock; 276public: 277 278 class SyncEvent; 279 280 typedef void (*sync_event_callback_t)(const wp<SyncEvent>& event) ; 281 282 class SyncEvent : public RefBase { 283 public: 284 SyncEvent(AudioSystem::sync_event_t type, 285 int triggerSession, 286 int listenerSession, 287 sync_event_callback_t callBack, 288 wp<RefBase> cookie) 289 : mType(type), mTriggerSession(triggerSession), mListenerSession(listenerSession), 290 mCallback(callBack), mCookie(cookie) 291 {} 292 293 virtual ~SyncEvent() {} 294 295 void trigger() { Mutex::Autolock _l(mLock); if (mCallback) mCallback(this); } 296 bool isCancelled() const { Mutex::Autolock _l(mLock); return (mCallback == NULL); } 297 void cancel() { Mutex::Autolock _l(mLock); mCallback = NULL; } 298 AudioSystem::sync_event_t type() const { return mType; } 299 int triggerSession() const { return mTriggerSession; } 300 int listenerSession() const { return mListenerSession; } 301 wp<RefBase> cookie() const { return mCookie; } 302 303 private: 304 const AudioSystem::sync_event_t mType; 305 const int mTriggerSession; 306 const int mListenerSession; 307 sync_event_callback_t mCallback; 308 const wp<RefBase> mCookie; 309 mutable Mutex mLock; 310 }; 311 312 sp<SyncEvent> createSyncEvent(AudioSystem::sync_event_t type, 313 int triggerSession, 314 int listenerSession, 315 sync_event_callback_t callBack, 316 wp<RefBase> cookie); 317 318private: 319 320 audio_mode_t getMode() const { return mMode; } 321 322 bool btNrecIsOff() const { return mBtNrecIsOff; } 323 324 AudioFlinger() ANDROID_API; 325 virtual ~AudioFlinger(); 326 327 // call in any IAudioFlinger method that accesses mPrimaryHardwareDev 328 status_t initCheck() const { return mPrimaryHardwareDev == NULL ? 329 NO_INIT : NO_ERROR; } 330 331 // RefBase 332 virtual void onFirstRef(); 333 334 AudioHwDevice* findSuitableHwDev_l(audio_module_handle_t module, 335 audio_devices_t devices); 336 void purgeStaleEffects_l(); 337 338 // Set kEnableExtendedChannels to true to enable greater than stereo output 339 // for the MixerThread and device sink. Number of channels allowed is 340 // FCC_2 <= channels <= AudioMixer::MAX_NUM_CHANNELS. 341 static const bool kEnableExtendedChannels = true; 342 343 // Returns true if channel mask is permitted for the PCM sink in the MixerThread 344 static inline bool isValidPcmSinkChannelMask(audio_channel_mask_t channelMask) { 345 switch (audio_channel_mask_get_representation(channelMask)) { 346 case AUDIO_CHANNEL_REPRESENTATION_POSITION: { 347 uint32_t channelCount = FCC_2; // stereo is default 348 if (kEnableExtendedChannels) { 349 channelCount = audio_channel_count_from_out_mask(channelMask); 350 if (channelCount < FCC_2 // mono is not supported at this time 351 || channelCount > AudioMixer::MAX_NUM_CHANNELS) { 352 return false; 353 } 354 } 355 // check that channelMask is the "canonical" one we expect for the channelCount. 356 return channelMask == audio_channel_out_mask_from_count(channelCount); 357 } 358 default: 359 return false; 360 } 361 } 362 363 // Set kEnableExtendedPrecision to true to use extended precision in MixerThread 364 static const bool kEnableExtendedPrecision = true; 365 366 // Returns true if format is permitted for the PCM sink in the MixerThread 367 static inline bool isValidPcmSinkFormat(audio_format_t format) { 368 switch (format) { 369 case AUDIO_FORMAT_PCM_16_BIT: 370 return true; 371 case AUDIO_FORMAT_PCM_FLOAT: 372 case AUDIO_FORMAT_PCM_24_BIT_PACKED: 373 case AUDIO_FORMAT_PCM_32_BIT: 374 case AUDIO_FORMAT_PCM_8_24_BIT: 375 return kEnableExtendedPrecision; 376 default: 377 return false; 378 } 379 } 380 381 // standby delay for MIXER and DUPLICATING playback threads is read from property 382 // ro.audio.flinger_standbytime_ms or defaults to kDefaultStandbyTimeInNsecs 383 static nsecs_t mStandbyTimeInNsecs; 384 385 // incremented by 2 when screen state changes, bit 0 == 1 means "off" 386 // AudioFlinger::setParameters() updates, other threads read w/o lock 387 static uint32_t mScreenState; 388 389 // Internal dump utilities. 390 static const int kDumpLockRetries = 50; 391 static const int kDumpLockSleepUs = 20000; 392 static bool dumpTryLock(Mutex& mutex); 393 void dumpPermissionDenial(int fd, const Vector<String16>& args); 394 void dumpClients(int fd, const Vector<String16>& args); 395 void dumpInternals(int fd, const Vector<String16>& args); 396 397 // --- Client --- 398 class Client : public RefBase { 399 public: 400 Client(const sp<AudioFlinger>& audioFlinger, pid_t pid); 401 virtual ~Client(); 402 sp<MemoryDealer> heap() const; 403 pid_t pid() const { return mPid; } 404 sp<AudioFlinger> audioFlinger() const { return mAudioFlinger; } 405 406 bool reserveTimedTrack(); 407 void releaseTimedTrack(); 408 409 private: 410 Client(const Client&); 411 Client& operator = (const Client&); 412 const sp<AudioFlinger> mAudioFlinger; 413 const sp<MemoryDealer> mMemoryDealer; 414 const pid_t mPid; 415 416 Mutex mTimedTrackLock; 417 int mTimedTrackCount; 418 }; 419 420 // --- Notification Client --- 421 class NotificationClient : public IBinder::DeathRecipient { 422 public: 423 NotificationClient(const sp<AudioFlinger>& audioFlinger, 424 const sp<IAudioFlingerClient>& client, 425 pid_t pid); 426 virtual ~NotificationClient(); 427 428 sp<IAudioFlingerClient> audioFlingerClient() const { return mAudioFlingerClient; } 429 430 // IBinder::DeathRecipient 431 virtual void binderDied(const wp<IBinder>& who); 432 433 private: 434 NotificationClient(const NotificationClient&); 435 NotificationClient& operator = (const NotificationClient&); 436 437 const sp<AudioFlinger> mAudioFlinger; 438 const pid_t mPid; 439 const sp<IAudioFlingerClient> mAudioFlingerClient; 440 }; 441 442 class TrackHandle; 443 class RecordHandle; 444 class RecordThread; 445 class PlaybackThread; 446 class MixerThread; 447 class DirectOutputThread; 448 class OffloadThread; 449 class DuplicatingThread; 450 class AsyncCallbackThread; 451 class Track; 452 class RecordTrack; 453 class EffectModule; 454 class EffectHandle; 455 class EffectChain; 456 457 struct AudioStreamIn; 458 459 struct stream_type_t { 460 stream_type_t() 461 : volume(1.0f), 462 mute(false) 463 { 464 } 465 float volume; 466 bool mute; 467 }; 468 469 // --- PlaybackThread --- 470 471#include "Threads.h" 472 473#include "Effects.h" 474 475#include "PatchPanel.h" 476 477 // server side of the client's IAudioTrack 478 class TrackHandle : public android::BnAudioTrack { 479 public: 480 TrackHandle(const sp<PlaybackThread::Track>& track); 481 virtual ~TrackHandle(); 482 virtual sp<IMemory> getCblk() const; 483 virtual status_t start(); 484 virtual void stop(); 485 virtual void flush(); 486 virtual void pause(); 487 virtual status_t attachAuxEffect(int effectId); 488 virtual status_t allocateTimedBuffer(size_t size, 489 sp<IMemory>* buffer); 490 virtual status_t queueTimedBuffer(const sp<IMemory>& buffer, 491 int64_t pts); 492 virtual status_t setMediaTimeTransform(const LinearTransform& xform, 493 int target); 494 virtual status_t setParameters(const String8& keyValuePairs); 495 virtual status_t getTimestamp(AudioTimestamp& timestamp); 496 virtual void signal(); // signal playback thread for a change in control block 497 498 virtual status_t onTransact( 499 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 500 501 private: 502 const sp<PlaybackThread::Track> mTrack; 503 }; 504 505 // server side of the client's IAudioRecord 506 class RecordHandle : public android::BnAudioRecord { 507 public: 508 RecordHandle(const sp<RecordThread::RecordTrack>& recordTrack); 509 virtual ~RecordHandle(); 510 virtual status_t start(int /*AudioSystem::sync_event_t*/ event, int triggerSession); 511 virtual void stop(); 512 virtual status_t onTransact( 513 uint32_t code, const Parcel& data, Parcel* reply, uint32_t flags); 514 private: 515 const sp<RecordThread::RecordTrack> mRecordTrack; 516 517 // for use from destructor 518 void stop_nonvirtual(); 519 }; 520 521 522 PlaybackThread *checkPlaybackThread_l(audio_io_handle_t output) const; 523 MixerThread *checkMixerThread_l(audio_io_handle_t output) const; 524 RecordThread *checkRecordThread_l(audio_io_handle_t input) const; 525 sp<RecordThread> openInput_l(audio_module_handle_t module, 526 audio_io_handle_t *input, 527 audio_config_t *config, 528 audio_devices_t device, 529 const String8& address, 530 audio_source_t source, 531 audio_input_flags_t flags); 532 sp<PlaybackThread> openOutput_l(audio_module_handle_t module, 533 audio_io_handle_t *output, 534 audio_config_t *config, 535 audio_devices_t devices, 536 const String8& address, 537 audio_output_flags_t flags); 538 539 void closeOutputFinish(sp<PlaybackThread> thread); 540 void closeInputFinish(sp<RecordThread> thread); 541 542 // no range check, AudioFlinger::mLock held 543 bool streamMute_l(audio_stream_type_t stream) const 544 { return mStreamTypes[stream].mute; } 545 // no range check, doesn't check per-thread stream volume, AudioFlinger::mLock held 546 float streamVolume_l(audio_stream_type_t stream) const 547 { return mStreamTypes[stream].volume; } 548 void ioConfigChanged(audio_io_config_event event, 549 const sp<AudioIoDescriptor>& ioDesc); 550 551 // Allocate an audio_io_handle_t, session ID, effect ID, or audio_module_handle_t. 552 // They all share the same ID space, but the namespaces are actually independent 553 // because there are separate KeyedVectors for each kind of ID. 554 // The return value is uint32_t, but is cast to signed for some IDs. 555 // FIXME This API does not handle rollover to zero (for unsigned IDs), 556 // or from positive to negative (for signed IDs). 557 // Thus it may fail by returning an ID of the wrong sign, 558 // or by returning a non-unique ID. 559 uint32_t nextUniqueId(); 560 561 status_t moveEffectChain_l(int sessionId, 562 PlaybackThread *srcThread, 563 PlaybackThread *dstThread, 564 bool reRegister); 565 // return thread associated with primary hardware device, or NULL 566 PlaybackThread *primaryPlaybackThread_l() const; 567 audio_devices_t primaryOutputDevice_l() const; 568 569 sp<PlaybackThread> getEffectThread_l(int sessionId, int EffectId); 570 571 572 void removeClient_l(pid_t pid); 573 void removeNotificationClient(pid_t pid); 574 bool isNonOffloadableGlobalEffectEnabled_l(); 575 void onNonOffloadableGlobalEffectEnable(); 576 577 // Store an effect chain to mOrphanEffectChains keyed vector. 578 // Called when a thread exits and effects are still attached to it. 579 // If effects are later created on the same session, they will reuse the same 580 // effect chain and same instances in the effect library. 581 // return ALREADY_EXISTS if a chain with the same session already exists in 582 // mOrphanEffectChains. Note that this should never happen as there is only one 583 // chain for a given session and it is attached to only one thread at a time. 584 status_t putOrphanEffectChain_l(const sp<EffectChain>& chain); 585 // Get an effect chain for the specified session in mOrphanEffectChains and remove 586 // it if found. Returns 0 if not found (this is the most common case). 587 sp<EffectChain> getOrphanEffectChain_l(audio_session_t session); 588 // Called when the last effect handle on an effect instance is removed. If this 589 // effect belongs to an effect chain in mOrphanEffectChains, the chain is updated 590 // and removed from mOrphanEffectChains if it does not contain any effect. 591 // Return true if the effect was found in mOrphanEffectChains, false otherwise. 592 bool updateOrphanEffectChains(const sp<EffectModule>& effect); 593 594 void broacastParametersToRecordThreads_l(const String8& keyValuePairs); 595 596 // AudioStreamIn is immutable, so their fields are const. 597 // For emphasis, we could also make all pointers to them be "const *", 598 // but that would clutter the code unnecessarily. 599 600 struct AudioStreamIn { 601 AudioHwDevice* const audioHwDev; 602 audio_stream_in_t* const stream; 603 604 audio_hw_device_t* hwDev() const { return audioHwDev->hwDevice(); } 605 606 AudioStreamIn(AudioHwDevice *dev, audio_stream_in_t *in) : 607 audioHwDev(dev), stream(in) {} 608 }; 609 610 // for mAudioSessionRefs only 611 struct AudioSessionRef { 612 AudioSessionRef(int sessionid, pid_t pid) : 613 mSessionid(sessionid), mPid(pid), mCnt(1) {} 614 const int mSessionid; 615 const pid_t mPid; 616 int mCnt; 617 }; 618 619 mutable Mutex mLock; 620 // protects mClients and mNotificationClients. 621 // must be locked after mLock and ThreadBase::mLock if both must be locked 622 // avoids acquiring AudioFlinger::mLock from inside thread loop. 623 mutable Mutex mClientLock; 624 // protected by mClientLock 625 DefaultKeyedVector< pid_t, wp<Client> > mClients; // see ~Client() 626 627 mutable Mutex mHardwareLock; 628 // NOTE: If both mLock and mHardwareLock mutexes must be held, 629 // always take mLock before mHardwareLock 630 631 // These two fields are immutable after onFirstRef(), so no lock needed to access 632 AudioHwDevice* mPrimaryHardwareDev; // mAudioHwDevs[0] or NULL 633 DefaultKeyedVector<audio_module_handle_t, AudioHwDevice*> mAudioHwDevs; 634 635 // for dump, indicates which hardware operation is currently in progress (but not stream ops) 636 enum hardware_call_state { 637 AUDIO_HW_IDLE = 0, // no operation in progress 638 AUDIO_HW_INIT, // init_check 639 AUDIO_HW_OUTPUT_OPEN, // open_output_stream 640 AUDIO_HW_OUTPUT_CLOSE, // unused 641 AUDIO_HW_INPUT_OPEN, // unused 642 AUDIO_HW_INPUT_CLOSE, // unused 643 AUDIO_HW_STANDBY, // unused 644 AUDIO_HW_SET_MASTER_VOLUME, // set_master_volume 645 AUDIO_HW_GET_ROUTING, // unused 646 AUDIO_HW_SET_ROUTING, // unused 647 AUDIO_HW_GET_MODE, // unused 648 AUDIO_HW_SET_MODE, // set_mode 649 AUDIO_HW_GET_MIC_MUTE, // get_mic_mute 650 AUDIO_HW_SET_MIC_MUTE, // set_mic_mute 651 AUDIO_HW_SET_VOICE_VOLUME, // set_voice_volume 652 AUDIO_HW_SET_PARAMETER, // set_parameters 653 AUDIO_HW_GET_INPUT_BUFFER_SIZE, // get_input_buffer_size 654 AUDIO_HW_GET_MASTER_VOLUME, // get_master_volume 655 AUDIO_HW_GET_PARAMETER, // get_parameters 656 AUDIO_HW_SET_MASTER_MUTE, // set_master_mute 657 AUDIO_HW_GET_MASTER_MUTE, // get_master_mute 658 }; 659 660 mutable hardware_call_state mHardwareStatus; // for dump only 661 662 663 DefaultKeyedVector< audio_io_handle_t, sp<PlaybackThread> > mPlaybackThreads; 664 stream_type_t mStreamTypes[AUDIO_STREAM_CNT]; 665 666 // member variables below are protected by mLock 667 float mMasterVolume; 668 bool mMasterMute; 669 // end of variables protected by mLock 670 671 DefaultKeyedVector< audio_io_handle_t, sp<RecordThread> > mRecordThreads; 672 673 // protected by mClientLock 674 DefaultKeyedVector< pid_t, sp<NotificationClient> > mNotificationClients; 675 676 volatile int32_t mNextUniqueId; // updated by android_atomic_inc 677 // nextUniqueId() returns uint32_t, but this is declared int32_t 678 // because the atomic operations require an int32_t 679 680 audio_mode_t mMode; 681 bool mBtNrecIsOff; 682 683 // protected by mLock 684 Vector<AudioSessionRef*> mAudioSessionRefs; 685 686 float masterVolume_l() const; 687 bool masterMute_l() const; 688 audio_module_handle_t loadHwModule_l(const char *name); 689 690 Vector < sp<SyncEvent> > mPendingSyncEvents; // sync events awaiting for a session 691 // to be created 692 693 // Effect chains without a valid thread 694 DefaultKeyedVector< audio_session_t , sp<EffectChain> > mOrphanEffectChains; 695 696 // list of sessions for which a valid HW A/V sync ID was retrieved from the HAL 697 DefaultKeyedVector< audio_session_t , audio_hw_sync_t >mHwAvSyncIds; 698private: 699 sp<Client> registerPid(pid_t pid); // always returns non-0 700 701 // for use from destructor 702 status_t closeOutput_nonvirtual(audio_io_handle_t output); 703 void closeOutputInternal_l(sp<PlaybackThread> thread); 704 status_t closeInput_nonvirtual(audio_io_handle_t input); 705 void closeInputInternal_l(sp<RecordThread> thread); 706 void setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId); 707 708 status_t checkStreamType(audio_stream_type_t stream) const; 709 710#ifdef TEE_SINK 711 // all record threads serially share a common tee sink, which is re-created on format change 712 sp<NBAIO_Sink> mRecordTeeSink; 713 sp<NBAIO_Source> mRecordTeeSource; 714#endif 715 716public: 717 718#ifdef TEE_SINK 719 // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file 720 static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id = 0); 721 722 // whether tee sink is enabled by property 723 static bool mTeeSinkInputEnabled; 724 static bool mTeeSinkOutputEnabled; 725 static bool mTeeSinkTrackEnabled; 726 727 // runtime configured size of each tee sink pipe, in frames 728 static size_t mTeeSinkInputFrames; 729 static size_t mTeeSinkOutputFrames; 730 static size_t mTeeSinkTrackFrames; 731 732 // compile-time default size of tee sink pipes, in frames 733 // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes 734 static const size_t kTeeSinkInputFramesDefault = 0x200000; 735 static const size_t kTeeSinkOutputFramesDefault = 0x200000; 736 static const size_t kTeeSinkTrackFramesDefault = 0x200000; 737#endif 738 739 // This method reads from a variable without mLock, but the variable is updated under mLock. So 740 // we might read a stale value, or a value that's inconsistent with respect to other variables. 741 // In this case, it's safe because the return value isn't used for making an important decision. 742 // The reason we don't want to take mLock is because it could block the caller for a long time. 743 bool isLowRamDevice() const { return mIsLowRamDevice; } 744 745private: 746 bool mIsLowRamDevice; 747 bool mIsDeviceTypeKnown; 748 nsecs_t mGlobalEffectEnableTime; // when a global effect was last enabled 749 750 sp<PatchPanel> mPatchPanel; 751 752 uint32_t mPrimaryOutputSampleRate; // sample rate of the primary output, or zero if none 753 // protected by mHardwareLock 754}; 755 756#undef INCLUDING_FROM_AUDIOFLINGER_H 757 758const char *formatToString(audio_format_t format); 759String8 inputFlagsToString(audio_input_flags_t flags); 760String8 outputFlagsToString(audio_output_flags_t flags); 761String8 devicesToString(audio_devices_t devices); 762const char *sourceToString(audio_source_t source); 763 764// ---------------------------------------------------------------------------- 765 766} // namespace android 767 768#endif // ANDROID_AUDIO_FLINGER_H 769