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 <hardware/audio_effect.h> 21 #include <media/AudioPolicy.h> 22 #include <media/IAudioFlingerClient.h> 23 #include <media/IAudioPolicyServiceClient.h> 24 #include <system/audio.h> 25 #include <system/audio_policy.h> 26 #include <utils/Errors.h> 27 #include <utils/Mutex.h> 28 29 namespace android { 30 31 typedef void (*audio_error_callback)(status_t err); 32 33 class IAudioFlinger; 34 class IAudioPolicyService; 35 class String8; 36 37 class AudioSystem 38 { 39 public: 40 41 /* These are static methods to control the system-wide AudioFlinger 42 * only privileged processes can have access to them 43 */ 44 45 // mute/unmute microphone 46 static status_t muteMicrophone(bool state); 47 static status_t isMicrophoneMuted(bool *state); 48 49 // set/get master volume 50 static status_t setMasterVolume(float value); 51 static status_t getMasterVolume(float* volume); 52 53 // mute/unmute audio outputs 54 static status_t setMasterMute(bool mute); 55 static status_t getMasterMute(bool* mute); 56 57 // set/get stream volume on specified output 58 static status_t setStreamVolume(audio_stream_type_t stream, float value, 59 audio_io_handle_t output); 60 static status_t getStreamVolume(audio_stream_type_t stream, float* volume, 61 audio_io_handle_t output); 62 63 // mute/unmute stream 64 static status_t setStreamMute(audio_stream_type_t stream, bool mute); 65 static status_t getStreamMute(audio_stream_type_t stream, bool* mute); 66 67 // set audio mode in audio hardware 68 static status_t setMode(audio_mode_t mode); 69 70 // returns true in *state if tracks are active on the specified stream or have been active 71 // in the past inPastMs milliseconds 72 static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs); 73 // returns true in *state if tracks are active for what qualifies as remote playback 74 // on the specified stream or have been active in the past inPastMs milliseconds. Remote 75 // playback isn't mutually exclusive with local playback. 76 static status_t isStreamActiveRemotely(audio_stream_type_t stream, bool *state, 77 uint32_t inPastMs); 78 // returns true in *state if a recorder is currently recording with the specified source 79 static status_t isSourceActive(audio_source_t source, bool *state); 80 81 // set/get audio hardware parameters. The function accepts a list of parameters 82 // key value pairs in the form: key1=value1;key2=value2;... 83 // Some keys are reserved for standard parameters (See AudioParameter class). 84 // The versions with audio_io_handle_t are intended for internal media framework use only. 85 static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 86 static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); 87 // The versions without audio_io_handle_t are intended for JNI. 88 static status_t setParameters(const String8& keyValuePairs); 89 static String8 getParameters(const String8& keys); 90 91 static void setErrorCallback(audio_error_callback cb); 92 93 // helper function to obtain AudioFlinger service handle 94 static const sp<IAudioFlinger> get_audio_flinger(); 95 96 static float linearToLog(int volume); 97 static int logToLinear(float volume); 98 99 // Returned samplingRate and frameCount output values are guaranteed 100 // to be non-zero if status == NO_ERROR 101 static status_t getOutputSamplingRate(uint32_t* samplingRate, 102 audio_stream_type_t stream); 103 static status_t getOutputFrameCount(size_t* frameCount, 104 audio_stream_type_t stream); 105 static status_t getOutputLatency(uint32_t* latency, 106 audio_stream_type_t stream); 107 static status_t getSamplingRate(audio_io_handle_t output, 108 uint32_t* samplingRate); 109 // returns the number of frames per audio HAL write buffer. Corresponds to 110 // audio_stream->get_buffer_size()/audio_stream_out_frame_size() 111 static status_t getFrameCount(audio_io_handle_t output, 112 size_t* frameCount); 113 // returns the audio output stream latency in ms. Corresponds to 114 // audio_stream_out->get_latency() 115 static status_t getLatency(audio_io_handle_t output, 116 uint32_t* latency); 117 118 // return status NO_ERROR implies *buffSize > 0 119 static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, 120 audio_channel_mask_t channelMask, size_t* buffSize); 121 122 static status_t setVoiceVolume(float volume); 123 124 // return the number of audio frames written by AudioFlinger to audio HAL and 125 // audio dsp to DAC since the specified output I/O handle has exited standby. 126 // returned status (from utils/Errors.h) can be: 127 // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data 128 // - INVALID_OPERATION: Not supported on current hardware platform 129 // - BAD_VALUE: invalid parameter 130 // NOTE: this feature is not supported on all hardware platforms and it is 131 // necessary to check returned status before using the returned values. 132 static status_t getRenderPosition(audio_io_handle_t output, 133 uint32_t *halFrames, 134 uint32_t *dspFrames); 135 136 // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid 137 static uint32_t getInputFramesLost(audio_io_handle_t ioHandle); 138 139 // Allocate a new unique ID for use as an audio session ID or I/O handle. 140 // If unable to contact AudioFlinger, returns AUDIO_UNIQUE_ID_ALLOCATE instead. 141 // FIXME If AudioFlinger were to ever exhaust the unique ID namespace, 142 // this method could fail by returning either AUDIO_UNIQUE_ID_ALLOCATE 143 // or an unspecified existing unique ID. 144 static audio_unique_id_t newAudioUniqueId(); 145 146 static void acquireAudioSessionId(int audioSession, pid_t pid); 147 static void releaseAudioSessionId(int audioSession, pid_t pid); 148 149 // Get the HW synchronization source used for an audio session. 150 // Return a valid source or AUDIO_HW_SYNC_INVALID if an error occurs 151 // or no HW sync source is used. 152 static audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId); 153 154 // types of io configuration change events received with ioConfigChanged() 155 enum io_config_event { 156 OUTPUT_OPENED, 157 OUTPUT_CLOSED, 158 OUTPUT_CONFIG_CHANGED, 159 INPUT_OPENED, 160 INPUT_CLOSED, 161 INPUT_CONFIG_CHANGED, 162 STREAM_CONFIG_CHANGED, 163 NUM_CONFIG_EVENTS 164 }; 165 166 // audio output descriptor used to cache output configurations in client process to avoid 167 // frequent calls through IAudioFlinger 168 class OutputDescriptor { 169 public: OutputDescriptor()170 OutputDescriptor() 171 : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channelMask(0), frameCount(0), latency(0) 172 {} 173 174 uint32_t samplingRate; 175 audio_format_t format; 176 audio_channel_mask_t channelMask; 177 size_t frameCount; 178 uint32_t latency; 179 }; 180 181 // Events used to synchronize actions between audio sessions. 182 // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until 183 // playback is complete on another audio session. 184 // See definitions in MediaSyncEvent.java 185 enum sync_event_t { 186 SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event 187 SYNC_EVENT_NONE = 0, 188 SYNC_EVENT_PRESENTATION_COMPLETE, 189 190 // 191 // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ... 192 // 193 SYNC_EVENT_CNT, 194 }; 195 196 // Timeout for synchronous record start. Prevents from blocking the record thread forever 197 // if the trigger event is not fired. 198 static const uint32_t kSyncRecordStartTimeOutMs = 30000; 199 200 // 201 // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) 202 // 203 static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, 204 const char *device_address); 205 static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, 206 const char *device_address); 207 static status_t setPhoneState(audio_mode_t state); 208 static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); 209 static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); 210 211 // Client must successfully hand off the handle reference to AudioFlinger via createTrack(), 212 // or release it with releaseOutput(). 213 static audio_io_handle_t getOutput(audio_stream_type_t stream, 214 uint32_t samplingRate = 0, 215 audio_format_t format = AUDIO_FORMAT_DEFAULT, 216 audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, 217 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 218 const audio_offload_info_t *offloadInfo = NULL); 219 static status_t getOutputForAttr(const audio_attributes_t *attr, 220 audio_io_handle_t *output, 221 audio_session_t session, 222 audio_stream_type_t *stream, 223 uint32_t samplingRate = 0, 224 audio_format_t format = AUDIO_FORMAT_DEFAULT, 225 audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, 226 audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE, 227 const audio_offload_info_t *offloadInfo = NULL); 228 static status_t startOutput(audio_io_handle_t output, 229 audio_stream_type_t stream, 230 audio_session_t session); 231 static status_t stopOutput(audio_io_handle_t output, 232 audio_stream_type_t stream, 233 audio_session_t session); 234 static void releaseOutput(audio_io_handle_t output, 235 audio_stream_type_t stream, 236 audio_session_t session); 237 238 // Client must successfully hand off the handle reference to AudioFlinger via openRecord(), 239 // or release it with releaseInput(). 240 static status_t getInputForAttr(const audio_attributes_t *attr, 241 audio_io_handle_t *input, 242 audio_session_t session, 243 uint32_t samplingRate, 244 audio_format_t format, 245 audio_channel_mask_t channelMask, 246 audio_input_flags_t flags); 247 248 static status_t startInput(audio_io_handle_t input, 249 audio_session_t session); 250 static status_t stopInput(audio_io_handle_t input, 251 audio_session_t session); 252 static void releaseInput(audio_io_handle_t input, 253 audio_session_t session); 254 static status_t initStreamVolume(audio_stream_type_t stream, 255 int indexMin, 256 int indexMax); 257 static status_t setStreamVolumeIndex(audio_stream_type_t stream, 258 int index, 259 audio_devices_t device); 260 static status_t getStreamVolumeIndex(audio_stream_type_t stream, 261 int *index, 262 audio_devices_t device); 263 264 static uint32_t getStrategyForStream(audio_stream_type_t stream); 265 static audio_devices_t getDevicesForStream(audio_stream_type_t stream); 266 267 static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); 268 static status_t registerEffect(const effect_descriptor_t *desc, 269 audio_io_handle_t io, 270 uint32_t strategy, 271 int session, 272 int id); 273 static status_t unregisterEffect(int id); 274 static status_t setEffectEnabled(int id, bool enabled); 275 276 // clear stream to output mapping cache (gStreamOutputMap) 277 // and output configuration cache (gOutputs) 278 static void clearAudioConfigCache(); 279 280 static const sp<IAudioPolicyService> get_audio_policy_service(); 281 282 // helpers for android.media.AudioManager.getProperty(), see description there for meaning 283 static uint32_t getPrimaryOutputSamplingRate(); 284 static size_t getPrimaryOutputFrameCount(); 285 286 static status_t setLowRamDevice(bool isLowRamDevice); 287 288 // Check if hw offload is possible for given format, stream type, sample rate, 289 // bit rate, duration, video and streaming or offload property is enabled 290 static bool isOffloadSupported(const audio_offload_info_t& info); 291 292 // check presence of audio flinger service. 293 // returns NO_ERROR if binding to service succeeds, DEAD_OBJECT otherwise 294 static status_t checkAudioFlinger(); 295 296 /* List available audio ports and their attributes */ 297 static status_t listAudioPorts(audio_port_role_t role, 298 audio_port_type_t type, 299 unsigned int *num_ports, 300 struct audio_port *ports, 301 unsigned int *generation); 302 303 /* Get attributes for a given audio port */ 304 static status_t getAudioPort(struct audio_port *port); 305 306 /* Create an audio patch between several source and sink ports */ 307 static status_t createAudioPatch(const struct audio_patch *patch, 308 audio_patch_handle_t *handle); 309 310 /* Release an audio patch */ 311 static status_t releaseAudioPatch(audio_patch_handle_t handle); 312 313 /* List existing audio patches */ 314 static status_t listAudioPatches(unsigned int *num_patches, 315 struct audio_patch *patches, 316 unsigned int *generation); 317 /* Set audio port configuration */ 318 static status_t setAudioPortConfig(const struct audio_port_config *config); 319 320 321 static status_t acquireSoundTriggerSession(audio_session_t *session, 322 audio_io_handle_t *ioHandle, 323 audio_devices_t *device); 324 static status_t releaseSoundTriggerSession(audio_session_t session); 325 326 static audio_mode_t getPhoneState(); 327 328 static status_t registerPolicyMixes(Vector<AudioMix> mixes, bool registration); 329 330 // ---------------------------------------------------------------------------- 331 332 class AudioPortCallback : public RefBase 333 { 334 public: 335 AudioPortCallback()336 AudioPortCallback() {} ~AudioPortCallback()337 virtual ~AudioPortCallback() {} 338 339 virtual void onAudioPortListUpdate() = 0; 340 virtual void onAudioPatchListUpdate() = 0; 341 virtual void onServiceDied() = 0; 342 343 }; 344 345 static void setAudioPortCallback(sp<AudioPortCallback> callBack); 346 347 private: 348 349 class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient 350 { 351 public: AudioFlingerClient()352 AudioFlingerClient() { 353 } 354 355 // DeathRecipient 356 virtual void binderDied(const wp<IBinder>& who); 357 358 // IAudioFlingerClient 359 360 // indicate a change in the configuration of an output or input: keeps the cached 361 // values for output/input parameters up-to-date in client process 362 virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const void *param2); 363 }; 364 365 class AudioPolicyServiceClient: public IBinder::DeathRecipient, 366 public BnAudioPolicyServiceClient 367 { 368 public: AudioPolicyServiceClient()369 AudioPolicyServiceClient() { 370 } 371 372 // DeathRecipient 373 virtual void binderDied(const wp<IBinder>& who); 374 375 // IAudioPolicyServiceClient 376 virtual void onAudioPortListUpdate(); 377 virtual void onAudioPatchListUpdate(); 378 }; 379 380 static sp<AudioFlingerClient> gAudioFlingerClient; 381 static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; 382 friend class AudioFlingerClient; 383 friend class AudioPolicyServiceClient; 384 385 static Mutex gLock; // protects gAudioFlinger and gAudioErrorCallback, 386 static Mutex gLockCache; // protects gOutputs, gPrevInSamplingRate, gPrevInFormat, 387 // gPrevInChannelMask and gInBuffSize 388 static Mutex gLockAPS; // protects gAudioPolicyService and gAudioPolicyServiceClient 389 static Mutex gLockAPC; // protects gAudioPortCallback 390 static sp<IAudioFlinger> gAudioFlinger; 391 static audio_error_callback gAudioErrorCallback; 392 393 static size_t gInBuffSize; 394 // previous parameters for recording buffer size queries 395 static uint32_t gPrevInSamplingRate; 396 static audio_format_t gPrevInFormat; 397 static audio_channel_mask_t gPrevInChannelMask; 398 399 static sp<IAudioPolicyService> gAudioPolicyService; 400 401 // list of output descriptors containing cached parameters 402 // (sampling rate, framecount, channel count...) 403 static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; 404 405 static sp<AudioPortCallback> gAudioPortCallback; 406 }; 407 408 }; // namespace android 409 410 #endif /*ANDROID_AUDIOSYSTEM_H_*/ 411