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