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