1 /* 2 * Copyright (C) 2008 The Android Open Source Project 3 * 4 * Licensed under the Apache License, Version 2.0 (the "License"); 5 * you may not use this file except in compliance with the License. 6 * You may obtain a copy of the License at 7 * 8 * http://www.apache.org/licenses/LICENSE-2.0 9 * 10 * Unless required by applicable law or agreed to in writing, software 11 * distributed under the License is distributed on an "AS IS" BASIS, 12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. 13 * See the License for the specific language governing permissions and 14 * limitations under the License. 15 */ 16 17 #ifndef ANDROID_AUDIOSYSTEM_H_ 18 #define ANDROID_AUDIOSYSTEM_H_ 19 20 #include <sys/types.h> 21 22 #include <media/AudioDeviceTypeAddr.h> 23 #include <media/AudioPolicy.h> 24 #include <media/AudioProductStrategy.h> 25 #include <media/AudioVolumeGroup.h> 26 #include <media/AudioIoDescriptor.h> 27 #include <media/IAudioFlingerClient.h> 28 #include <media/IAudioPolicyServiceClient.h> 29 #include <media/MicrophoneInfo.h> 30 #include <system/audio.h> 31 #include <system/audio_effect.h> 32 #include <system/audio_policy.h> 33 #include <utils/Errors.h> 34 #include <utils/Mutex.h> 35 #include <vector> 36 37 namespace android { 38 39 typedef void (*audio_error_callback)(status_t err); 40 typedef void (*dynamic_policy_callback)(int event, String8 regId, int val); 41 typedef void (*record_config_callback)(int event, 42 const record_client_info_t *clientInfo, 43 const audio_config_base_t *clientConfig, 44 std::vector<effect_descriptor_t> clientEffects, 45 const audio_config_base_t *deviceConfig, 46 std::vector<effect_descriptor_t> effects, 47 audio_patch_handle_t patchHandle, 48 audio_source_t source); 49 50 class IAudioFlinger; 51 class IAudioPolicyService; 52 class String8; 53 54 class AudioSystem 55 { 56 public: 57 58 // FIXME Declare in binder opcode order, similarly to IAudioFlinger.h and IAudioFlinger.cpp 59 60 /* These are static methods to control the system-wide AudioFlinger 61 * only privileged processes can have access to them 62 */ 63 64 // mute/unmute microphone 65 static status_t muteMicrophone(bool state); 66 static status_t isMicrophoneMuted(bool *state); 67 68 // set/get master volume 69 static status_t setMasterVolume(float value); 70 static status_t getMasterVolume(float* volume); 71 72 // mute/unmute audio outputs 73 static status_t setMasterMute(bool mute); 74 static status_t getMasterMute(bool* mute); 75 76 // set/get stream volume on specified output 77 static status_t setStreamVolume(audio_stream_type_t stream, float value, 78 audio_io_handle_t output); 79 static status_t getStreamVolume(audio_stream_type_t stream, float* volume, 80 audio_io_handle_t output); 81 82 // mute/unmute stream 83 static status_t setStreamMute(audio_stream_type_t stream, bool mute); 84 static status_t getStreamMute(audio_stream_type_t stream, bool* mute); 85 86 // set audio mode in audio hardware 87 static status_t setMode(audio_mode_t mode); 88 89 // returns true in *state if tracks are active on the specified stream or have been active 90 // in the past inPastMs milliseconds 91 static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs); 92 // returns true in *state if tracks are active for what qualifies as remote playback 93 // on the specified stream or have been active in the past inPastMs milliseconds. Remote 94 // playback isn't mutually exclusive with local playback. 95 static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state, 96 uint32_t inPastMs); 97 // returns true in *state if a recorder is currently recording with the specified source 98 static status_t isSourceActive(audio_source_t source, bool *state); 99 100 // set/get audio hardware parameters. The function accepts a list of parameters 101 // key value pairs in the form: key1=value1;key2=value2;... 102 // Some keys are reserved for standard parameters (See AudioParameter class). 103 // The versions with audio_io_handle_t are intended for internal media framework use only. 104 static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 105 static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); 106 // The versions without audio_io_handle_t are intended for JNI. 107 static status_t setParameters(const String8& keyValuePairs); 108 static String8 getParameters(const String8& keys); 109 110 static void setErrorCallback(audio_error_callback cb); 111 static void setDynPolicyCallback(dynamic_policy_callback cb); 112 static void setRecordConfigCallback(record_config_callback); 113 114 // helper function to obtain AudioFlinger service handle 115 static const sp<IAudioFlinger> get_audio_flinger(); 116 117 static float linearToLog(int volume); 118 static int logToLinear(float volume); 119 static size_t calculateMinFrameCount( 120 uint32_t afLatencyMs, uint32_t afFrameCount, uint32_t afSampleRate, 121 uint32_t sampleRate, float speed /*, uint32_t notificationsPerBufferReq*/); 122 123 // Returned samplingRate and frameCount output values are guaranteed 124 // to be non-zero if status == NO_ERROR 125 // FIXME This API assumes a route, and so should be deprecated. 126 static status_t getOutputSamplingRate(uint32_t* samplingRate, 127 audio_stream_type_t stream); 128 // FIXME This API assumes a route, and so should be deprecated. 129 static status_t getOutputFrameCount(size_t* frameCount, 130 audio_stream_type_t stream); 131 // FIXME This API assumes a route, and so should be deprecated. 132 static status_t getOutputLatency(uint32_t* latency, 133 audio_stream_type_t stream); 134 // returns the audio HAL sample rate 135 static status_t getSamplingRate(audio_io_handle_t ioHandle, 136 uint32_t* samplingRate); 137 // For output threads with a fast mixer, returns the number of frames per normal mixer buffer. 138 // For output threads without a fast mixer, or for input, this is same as getFrameCountHAL(). 139 static status_t getFrameCount(audio_io_handle_t ioHandle, 140 size_t* frameCount); 141 // returns the audio output latency in ms. Corresponds to 142 // audio_stream_out->get_latency() 143 static status_t getLatency(audio_io_handle_t output, 144 uint32_t* latency); 145 146 // return status NO_ERROR implies *buffSize > 0 147 // FIXME This API assumes a route, and so should deprecated. 148 static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 149 audio_channel_mask_t channelMask, size_t* buffSize); 150 151 static status_t setVoiceVolume(float volume); 152 153 // return the number of audio frames written by AudioFlinger to audio HAL and 154 // audio dsp to DAC since the specified output has exited standby. 155 // returned status (from utils/Errors.h) can be: 156 // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data 157 // - INVALID_OPERATION: Not supported on current hardware platform 158 // - BAD_VALUE: invalid parameter 159 // NOTE: this feature is not supported on all hardware platforms and it is 160 // necessary to check returned status before using the returned values. 161 static status_t getRenderPosition(audio_io_handle_t output, 162 uint32_t *halFrames, 163 uint32_t *dspFrames); 164 165 // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid 166 static uint32_t getInputFramesLost(audio_io_handle_t ioHandle); 167 168 // Allocate a new unique ID for use as an audio session ID or I/O handle. 169 // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead. 170 // FIXME If AudioFlinger were to ever exhaust the unique ID namespace, 171 // this method could fail by returning either a reserved ID like AUDIO_UNIQUE_ID_ALLOCATE 172 // or an unspecified existing unique ID. 173 static audio_unique_id_t newAudioUniqueId(audio_unique_id_use_t use); 174 175 static void acquireAudioSessionId(audio_session_t audioSession, pid_t pid); 176 static void releaseAudioSessionId(audio_session_t audioSession, pid_t pid); 177 178 // Get the HW synchronization source used for an audio session. 179 // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs 180 // or no HW sync source is used. 181 static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 182 183 // Indicate JAVA services are ready (scheduling, power management ...) 184 static status_t systemReady(); 185 186 // Returns the number of frames per audio HAL buffer. 187 // Corresponds to audio_stream->get_buffer_size()/audio_stream_in_frame_size() for input. 188 // See also getFrameCount(). 189 static status_t getFrameCountHAL(audio_io_handle_t ioHandle, 190 size_t* frameCount); 191 192 // Events used to synchronize actions between audio sessions. 193 // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until 194 // playback is complete on another audio session. 195 // See definitions in MediaSyncEvent.java 196 enum sync_event_t { 197 SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event 198 SYNC_EVENT_NONE = 0, 199 SYNC_EVENT_PRESENTATION_COMPLETE, 200 201 // 202 // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ... 203 // 204 SYNC_EVENT_CNT, 205 }; 206 207 // Timeout for synchronous record start. Prevents from blocking the record thread forever 208 // if the trigger event is not fired. 209 static const uint32_t kSyncRecordStartTimeOutMs = 30000; 210 211 // 212 // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) 213 // 214 static void onNewAudioModulesAvailable(); 215 static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, 216 const char *device_address, const char *device_name, 217 audio_format_t encodedFormat); 218 static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, 219 const char *device_address); 220 static status_t handleDeviceConfigChange(audio_devices_t device, 221 const char *device_address, 222 const char *device_name, 223 audio_format_t encodedFormat); 224 static status_t setPhoneState(audio_mode_t state); 225 static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); 226 static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); 227 228 static status_t getOutputForAttr(audio_attributes_t *attr, 229 audio_io_handle_t *output, 230 audio_session_t session, 231 audio_stream_type_t *stream, 232 pid_t pid, 233 uid_t uid, 234 const audio_config_t *config, 235 audio_output_flags_t flags, 236 audio_port_handle_t *selectedDeviceId, 237 audio_port_handle_t *portId, 238 std::vector<audio_io_handle_t> *secondaryOutputs); 239 static status_t startOutput(audio_port_handle_t portId); 240 static status_t stopOutput(audio_port_handle_t portId); 241 static void releaseOutput(audio_port_handle_t portId); 242 243 // Client must successfully hand off the handle reference to AudioFlinger via createRecord(), 244 // or release it with releaseInput(). 245 static status_t getInputForAttr(const audio_attributes_t *attr, 246 audio_io_handle_t *input, 247 audio_unique_id_t riid, 248 audio_session_t session, 249 pid_t pid, 250 uid_t uid, 251 const String16& opPackageName, 252 const audio_config_base_t *config, 253 audio_input_flags_t flags, 254 audio_port_handle_t *selectedDeviceId, 255 audio_port_handle_t *portId); 256 257 static status_t startInput(audio_port_handle_t portId); 258 static status_t stopInput(audio_port_handle_t portId); 259 static void releaseInput(audio_port_handle_t portId); 260 static status_t initStreamVolume(audio_stream_type_t stream, 261 int indexMin, 262 int indexMax); 263 static status_t setStreamVolumeIndex(audio_stream_type_t stream, 264 int index, 265 audio_devices_t device); 266 static status_t getStreamVolumeIndex(audio_stream_type_t stream, 267 int *index, 268 audio_devices_t device); 269 270 static status_t setVolumeIndexForAttributes(const audio_attributes_t &attr, 271 int index, 272 audio_devices_t device); 273 static status_t getVolumeIndexForAttributes(const audio_attributes_t &attr, 274 int &index, 275 audio_devices_t device); 276 277 static status_t getMaxVolumeIndexForAttributes(const audio_attributes_t &attr, int &index); 278 279 static status_t getMinVolumeIndexForAttributes(const audio_attributes_t &attr, int &index); 280 281 static uint32_t getStrategyForStream(audio_stream_type_t stream); 282 static audio_devices_t getDevicesForStream(audio_stream_type_t stream); 283 284 static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); 285 static status_t registerEffect(const effect_descriptor_t *desc, 286 audio_io_handle_t io, 287 uint32_t strategy, 288 audio_session_t session, 289 int id); 290 static status_t unregisterEffect(int id); 291 static status_t setEffectEnabled(int id, bool enabled); 292 static status_t moveEffectsToIo(const std::vector<int>& ids, audio_io_handle_t io); 293 294 // clear stream to output mapping cache (gStreamOutputMap) 295 // and output configuration cache (gOutputs) 296 static void clearAudioConfigCache(); 297 298 static const sp<IAudioPolicyService> get_audio_policy_service(); 299 300 // helpers for android.media.AudioManager.getProperty(), see description there for meaning 301 static uint32_t getPrimaryOutputSamplingRate(); 302 static size_t getPrimaryOutputFrameCount(); 303 304 static status_t setLowRamDevice(bool isLowRamDevice, int64_t totalMemory); 305 306 static status_t setAllowedCapturePolicy(uid_t uid, audio_flags_mask_t flags); 307 308 // Check if hw offload is possible for given format, stream type, sample rate, 309 // bit rate, duration, video and streaming or offload property is enabled 310 static bool isOffloadSupported(const audio_offload_info_t& info); 311 312 // check presence of audio flinger service. 313 // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise 314 static status_t checkAudioFlinger(); 315 316 /* List available audio ports and their attributes */ 317 static status_t listAudioPorts(audio_port_role_t role, 318 audio_port_type_t type, 319 unsigned int *num_ports, 320 struct audio_port *ports, 321 unsigned int *generation); 322 323 /* Get attributes for a given audio port */ 324 static status_t getAudioPort(struct audio_port *port); 325 326 /* Create an audio patch between several source and sink ports */ 327 static status_t createAudioPatch(const struct audio_patch *patch, 328 audio_patch_handle_t *handle); 329 330 /* Release an audio patch */ 331 static status_t releaseAudioPatch(audio_patch_handle_t handle); 332 333 /* List existing audio patches */ 334 static status_t listAudioPatches(unsigned int *num_patches, 335 struct audio_patch *patches, 336 unsigned int *generation); 337 /* Set audio port configuration */ 338 static status_t setAudioPortConfig(const struct audio_port_config *config); 339 340 341 static status_t acquireSoundTriggerSession(audio_session_t *session, 342 audio_io_handle_t *ioHandle, 343 audio_devices_t *device); 344 static status_t releaseSoundTriggerSession(audio_session_t session); 345 346 static audio_mode_t getPhoneState(); 347 348 static status_t registerPolicyMixes(const Vector<AudioMix>& mixes, bool registration); 349 350 static status_t setUidDeviceAffinities(uid_t uid, const Vector<AudioDeviceTypeAddr>& devices); 351 352 static status_t removeUidDeviceAffinities(uid_t uid); 353 354 static status_t startAudioSource(const struct audio_port_config *source, 355 const audio_attributes_t *attributes, 356 audio_port_handle_t *portId); 357 static status_t stopAudioSource(audio_port_handle_t portId); 358 359 static status_t setMasterMono(bool mono); 360 static status_t getMasterMono(bool *mono); 361 362 static status_t setMasterBalance(float balance); 363 static status_t getMasterBalance(float *balance); 364 365 static float getStreamVolumeDB( 366 audio_stream_type_t stream, int index, audio_devices_t device); 367 368 static status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones); 369 370 static status_t getHwOffloadEncodingFormatsSupportedForA2DP( 371 std::vector<audio_format_t> *formats); 372 373 // numSurroundFormats holds the maximum number of formats and bool value allowed in the array. 374 // When numSurroundFormats is 0, surroundFormats and surroundFormatsEnabled will not be 375 // populated. The actual number of surround formats should be returned at numSurroundFormats. 376 static status_t getSurroundFormats(unsigned int *numSurroundFormats, 377 audio_format_t *surroundFormats, 378 bool *surroundFormatsEnabled, 379 bool reported); 380 static status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled); 381 382 static status_t setAssistantUid(uid_t uid); 383 static status_t setA11yServicesUids(const std::vector<uid_t>& uids); 384 385 static bool isHapticPlaybackSupported(); 386 387 static status_t listAudioProductStrategies(AudioProductStrategyVector &strategies); 388 static status_t getProductStrategyFromAudioAttributes(const AudioAttributes &aa, 389 product_strategy_t &productStrategy); 390 391 static audio_attributes_t streamTypeToAttributes(audio_stream_type_t stream); 392 static audio_stream_type_t attributesToStreamType(const audio_attributes_t &attr); 393 394 static status_t listAudioVolumeGroups(AudioVolumeGroupVector &groups); 395 396 static status_t getVolumeGroupFromAudioAttributes(const AudioAttributes &aa, 397 volume_group_t &volumeGroup); 398 399 static status_t setRttEnabled(bool enabled); 400 401 /** 402 * Send audio HAL server process pids to native audioserver process for use 403 * when generating audio HAL servers tombstones 404 */ 405 static status_t setAudioHalPids(const std::vector<pid_t>& pids); 406 407 static status_t setPreferredDeviceForStrategy(product_strategy_t strategy, 408 const AudioDeviceTypeAddr &device); 409 410 static status_t removePreferredDeviceForStrategy(product_strategy_t strategy); 411 412 static status_t getPreferredDeviceForStrategy(product_strategy_t strategy, 413 AudioDeviceTypeAddr &device); 414 415 static status_t getDeviceForStrategy(product_strategy_t strategy, 416 AudioDeviceTypeAddr &device); 417 418 // ---------------------------------------------------------------------------- 419 420 class AudioVolumeGroupCallback : public RefBase 421 { 422 public: 423 AudioVolumeGroupCallback()424 AudioVolumeGroupCallback() {} ~AudioVolumeGroupCallback()425 virtual ~AudioVolumeGroupCallback() {} 426 427 virtual void onAudioVolumeGroupChanged(volume_group_t group, int flags) = 0; 428 virtual void onServiceDied() = 0; 429 430 }; 431 432 static status_t addAudioVolumeGroupCallback(const sp<AudioVolumeGroupCallback>& callback); 433 static status_t removeAudioVolumeGroupCallback(const sp<AudioVolumeGroupCallback>& callback); 434 435 class AudioPortCallback : public RefBase 436 { 437 public: 438 AudioPortCallback()439 AudioPortCallback() {} ~AudioPortCallback()440 virtual ~AudioPortCallback() {} 441 442 virtual void onAudioPortListUpdate() = 0; 443 virtual void onAudioPatchListUpdate() = 0; 444 virtual void onServiceDied() = 0; 445 446 }; 447 448 static status_t addAudioPortCallback(const sp<AudioPortCallback>& callback); 449 static status_t removeAudioPortCallback(const sp<AudioPortCallback>& callback); 450 451 class AudioDeviceCallback : public RefBase 452 { 453 public: 454 AudioDeviceCallback()455 AudioDeviceCallback() {} ~AudioDeviceCallback()456 virtual ~AudioDeviceCallback() {} 457 458 virtual void onAudioDeviceUpdate(audio_io_handle_t audioIo, 459 audio_port_handle_t deviceId) = 0; 460 }; 461 462 static status_t addAudioDeviceCallback(const wp<AudioDeviceCallback>& callback, 463 audio_io_handle_t audioIo, 464 audio_port_handle_t portId); 465 static status_t removeAudioDeviceCallback(const wp<AudioDeviceCallback>& callback, 466 audio_io_handle_t audioIo, 467 audio_port_handle_t portId); 468 469 static audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo); 470 471 private: 472 473 class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient 474 { 475 public: AudioFlingerClient()476 AudioFlingerClient() : 477 mInBuffSize(0), mInSamplingRate(0), 478 mInFormat(AUDIO_FORMAT_DEFAULT), mInChannelMask(AUDIO_CHANNEL_NONE) { 479 } 480 481 void clearIoCache(); 482 status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 483 audio_channel_mask_t channelMask, size_t* buffSize); 484 sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle); 485 486 // DeathRecipient 487 virtual void binderDied(const wp<IBinder>& who); 488 489 // IAudioFlingerClient 490 491 // indicate a change in the configuration of an output or input: keeps the cached 492 // values for output/input parameters up-to-date in client process 493 virtual void ioConfigChanged(audio_io_config_event event, 494 const sp<AudioIoDescriptor>& ioDesc); 495 496 497 status_t addAudioDeviceCallback(const wp<AudioDeviceCallback>& callback, 498 audio_io_handle_t audioIo, 499 audio_port_handle_t portId); 500 status_t removeAudioDeviceCallback(const wp<AudioDeviceCallback>& callback, 501 audio_io_handle_t audioIo, 502 audio_port_handle_t portId); 503 504 audio_port_handle_t getDeviceIdForIo(audio_io_handle_t audioIo); 505 506 private: 507 Mutex mLock; 508 DefaultKeyedVector<audio_io_handle_t, sp<AudioIoDescriptor> > mIoDescriptors; 509 510 std::map<audio_io_handle_t, std::map<audio_port_handle_t, wp<AudioDeviceCallback>>> 511 mAudioDeviceCallbacks; 512 // cached values for recording getInputBufferSize() queries 513 size_t mInBuffSize; // zero indicates cache is invalid 514 uint32_t mInSamplingRate; 515 audio_format_t mInFormat; 516 audio_channel_mask_t mInChannelMask; 517 sp<AudioIoDescriptor> getIoDescriptor_l(audio_io_handle_t ioHandle); 518 }; 519 520 class AudioPolicyServiceClient: public IBinder::DeathRecipient, 521 public BnAudioPolicyServiceClient 522 { 523 public: AudioPolicyServiceClient()524 AudioPolicyServiceClient() { 525 } 526 527 int addAudioPortCallback(const sp<AudioPortCallback>& callback); 528 int removeAudioPortCallback(const sp<AudioPortCallback>& callback); isAudioPortCbEnabled()529 bool isAudioPortCbEnabled() const { return (mAudioPortCallbacks.size() != 0); } 530 531 int addAudioVolumeGroupCallback(const sp<AudioVolumeGroupCallback>& callback); 532 int removeAudioVolumeGroupCallback(const sp<AudioVolumeGroupCallback>& callback); isAudioVolumeGroupCbEnabled()533 bool isAudioVolumeGroupCbEnabled() const { return (mAudioVolumeGroupCallback.size() != 0); } 534 535 // DeathRecipient 536 virtual void binderDied(const wp<IBinder>& who); 537 538 // IAudioPolicyServiceClient 539 virtual void onAudioPortListUpdate(); 540 virtual void onAudioPatchListUpdate(); 541 virtual void onAudioVolumeGroupChanged(volume_group_t group, int flags); 542 virtual void onDynamicPolicyMixStateUpdate(String8 regId, int32_t state); 543 virtual void onRecordingConfigurationUpdate(int event, 544 const record_client_info_t *clientInfo, 545 const audio_config_base_t *clientConfig, 546 std::vector<effect_descriptor_t> clientEffects, 547 const audio_config_base_t *deviceConfig, 548 std::vector<effect_descriptor_t> effects, 549 audio_patch_handle_t patchHandle, 550 audio_source_t source); 551 552 private: 553 Mutex mLock; 554 Vector <sp <AudioPortCallback> > mAudioPortCallbacks; 555 Vector <sp <AudioVolumeGroupCallback> > mAudioVolumeGroupCallback; 556 }; 557 558 static audio_io_handle_t getOutput(audio_stream_type_t stream); 559 static const sp<AudioFlingerClient> getAudioFlingerClient(); 560 static sp<AudioIoDescriptor> getIoDescriptor(audio_io_handle_t ioHandle); 561 562 static sp<AudioFlingerClient> gAudioFlingerClient; 563 static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; 564 friend class AudioFlingerClient; 565 friend class AudioPolicyServiceClient; 566 567 static Mutex gLock; // protects gAudioFlinger and gAudioErrorCallback, 568 static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient 569 static sp<IAudioFlinger> gAudioFlinger; 570 static audio_error_callback gAudioErrorCallback; 571 static dynamic_policy_callback gDynPolicyCallback; 572 static record_config_callback gRecordConfigCallback; 573 574 static size_t gInBuffSize; 575 // previous parameters for recording buffer size queries 576 static uint32_t gPrevInSamplingRate; 577 static audio_format_t gPrevInFormat; 578 static audio_channel_mask_t gPrevInChannelMask; 579 580 static sp<IAudioPolicyService> gAudioPolicyService; 581 }; 582 583 }; // namespace android 584 585 #endif /*ANDROID_AUDIOSYSTEM_H_*/ 586