AudioSystem.h revision be916aa1267e2e6b1c148f51d11bcbbc79cb864c
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 <utils/RefBase.h> 21#include <utils/threads.h> 22#include <media/IAudioFlinger.h> 23#include <media/AudioCommon.h> 24 25namespace android { 26 27typedef void (*audio_error_callback)(status_t err); 28typedef int audio_io_handle_t; 29 30class IAudioPolicyService; 31class String8; 32 33class AudioSystem 34{ 35public: 36 37 enum stream_type { 38 DEFAULT =-1, 39 VOICE_CALL = 0, 40 SYSTEM = 1, 41 RING = 2, 42 MUSIC = 3, 43 ALARM = 4, 44 NOTIFICATION = 5, 45 BLUETOOTH_SCO = 6, 46 ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker 47 DTMF = 8, 48 TTS = 9, 49 NUM_STREAM_TYPES 50 }; 51 52 // Audio sub formats (see AudioSystem::audio_format). 53 enum pcm_sub_format { 54 PCM_SUB_16_BIT = PCM_FORMAT_S15, // must be 1 for backward compatibility 55 PCM_SUB_8_BIT = PCM_FORMAT_U8, // must be 2 for backward compatibility 56 }; 57 58 // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify 59 // bit rate, stereo mode, version... 60 enum mp3_sub_format { 61 //TODO 62 }; 63 64 // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned, 65 // encoding mode for recording... 66 enum amr_sub_format { 67 //TODO 68 }; 69 70 // AAC sub format field definition: specify profile or bitrate for recording... 71 enum aac_sub_format { 72 //TODO 73 }; 74 75 // VORBIS sub format field definition: specify quality for recording... 76 enum vorbis_sub_format { 77 //TODO 78 }; 79 80 // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits). 81 // The main format indicates the main codec type. The sub format field indicates options and parameters 82 // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate 83 // or profile. It can also be used for certain formats to give informations not present in the encoded 84 // audio stream (e.g. octet alignement for AMR). 85 enum audio_format { 86 INVALID_FORMAT = -1, 87 FORMAT_DEFAULT = 0, 88 PCM = 0x00000000, // must be 0 for backward compatibility 89 MP3 = 0x01000000, 90 AMR_NB = 0x02000000, 91 AMR_WB = 0x03000000, 92 AAC = 0x04000000, 93 HE_AAC_V1 = 0x05000000, 94 HE_AAC_V2 = 0x06000000, 95 VORBIS = 0x07000000, 96 MAIN_FORMAT_MASK = 0xFF000000, 97 SUB_FORMAT_MASK = 0x00FFFFFF, 98 // Aliases 99 PCM_16_BIT = (PCM|PCM_SUB_16_BIT), 100 PCM_8_BIT = (PCM|PCM_SUB_8_BIT) 101 }; 102 103 104 // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java 105 enum audio_channels { 106 // output channels 107 CHANNEL_OUT_FRONT_LEFT = CHANNEL_FRONT_LEFT, 108 CHANNEL_OUT_FRONT_RIGHT = CHANNEL_FRONT_RIGHT, 109 CHANNEL_OUT_FRONT_CENTER = CHANNEL_FRONT_CENTER, 110 CHANNEL_OUT_LOW_FREQUENCY = CHANNEL_LOW_FREQUENCY, 111 CHANNEL_OUT_BACK_LEFT = CHANNEL_BACK_LEFT, 112 CHANNEL_OUT_BACK_RIGHT = CHANNEL_BACK_RIGHT, 113 CHANNEL_OUT_FRONT_LEFT_OF_CENTER = CHANNEL_FRONT_LEFT_OF_CENTER, 114 CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = CHANNEL_FRONT_RIGHT_OF_CENTER, 115 CHANNEL_OUT_BACK_CENTER = CHANNEL_BACK_CENTER, 116 CHANNEL_OUT_MONO = CHANNEL_MONO, 117 CHANNEL_OUT_STEREO = CHANNEL_STEREO, 118 CHANNEL_OUT_QUAD = CHANNEL_QUAD, 119 CHANNEL_OUT_SURROUND = CHANNEL_SURROUND, 120 CHANNEL_OUT_5POINT1 = CHANNEL_5POINT1, 121 CHANNEL_OUT_7POINT1 = CHANNEL_7POINT1, 122 CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | 123 CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | 124 CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER), 125 126 // input channels 127 CHANNEL_IN_LEFT = 0x4, 128 CHANNEL_IN_RIGHT = 0x8, 129 CHANNEL_IN_FRONT = 0x10, 130 CHANNEL_IN_BACK = 0x20, 131 CHANNEL_IN_LEFT_PROCESSED = 0x40, 132 CHANNEL_IN_RIGHT_PROCESSED = 0x80, 133 CHANNEL_IN_FRONT_PROCESSED = 0x100, 134 CHANNEL_IN_BACK_PROCESSED = 0x200, 135 CHANNEL_IN_PRESSURE = 0x400, 136 CHANNEL_IN_X_AXIS = 0x800, 137 CHANNEL_IN_Y_AXIS = 0x1000, 138 CHANNEL_IN_Z_AXIS = 0x2000, 139 CHANNEL_IN_VOICE_UPLINK = 0x4000, 140 CHANNEL_IN_VOICE_DNLINK = 0x8000, 141 CHANNEL_IN_MONO = CHANNEL_IN_FRONT, 142 CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT), 143 CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK| 144 CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED| 145 CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS | 146 CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK) 147 }; 148 149 enum audio_mode { 150 MODE_INVALID = -2, 151 MODE_CURRENT = -1, 152 MODE_NORMAL = 0, 153 MODE_RINGTONE, 154 MODE_IN_CALL, 155 NUM_MODES // not a valid entry, denotes end-of-list 156 }; 157 158 enum audio_in_acoustics { 159 AGC_ENABLE = 0x0001, 160 AGC_DISABLE = 0, 161 NS_ENABLE = 0x0002, 162 NS_DISABLE = 0, 163 TX_IIR_ENABLE = 0x0004, 164 TX_DISABLE = 0 165 }; 166 167 /* These are static methods to control the system-wide AudioFlinger 168 * only privileged processes can have access to them 169 */ 170 171 // mute/unmute microphone 172 static status_t muteMicrophone(bool state); 173 static status_t isMicrophoneMuted(bool *state); 174 175 // set/get master volume 176 static status_t setMasterVolume(float value); 177 static status_t getMasterVolume(float* volume); 178 // mute/unmute audio outputs 179 static status_t setMasterMute(bool mute); 180 static status_t getMasterMute(bool* mute); 181 182 // set/get stream volume on specified output 183 static status_t setStreamVolume(int stream, float value, int output); 184 static status_t getStreamVolume(int stream, float* volume, int output); 185 186 // mute/unmute stream 187 static status_t setStreamMute(int stream, bool mute); 188 static status_t getStreamMute(int stream, bool* mute); 189 190 // set audio mode in audio hardware (see AudioSystem::audio_mode) 191 static status_t setMode(int mode); 192 193 // returns true in *state if tracks are active on the specified stream 194 static status_t isStreamActive(int stream, bool *state); 195 196 // set/get audio hardware parameters. The function accepts a list of parameters 197 // key value pairs in the form: key1=value1;key2=value2;... 198 // Some keys are reserved for standard parameters (See AudioParameter class). 199 static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); 200 static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); 201 202 static void setErrorCallback(audio_error_callback cb); 203 204 // helper function to obtain AudioFlinger service handle 205 static const sp<IAudioFlinger>& get_audio_flinger(); 206 207 static float linearToLog(int volume); 208 static int logToLinear(float volume); 209 210 static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT); 211 static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT); 212 static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT); 213 214 static bool routedToA2dpOutput(int streamType); 215 216 static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount, 217 size_t* buffSize); 218 219 static status_t setVoiceVolume(float volume); 220 221 // return the number of audio frames written by AudioFlinger to audio HAL and 222 // audio dsp to DAC since the output on which the specificed stream is playing 223 // has exited standby. 224 // returned status (from utils/Errors.h) can be: 225 // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data 226 // - INVALID_OPERATION: Not supported on current hardware platform 227 // - BAD_VALUE: invalid parameter 228 // NOTE: this feature is not supported on all hardware platforms and it is 229 // necessary to check returned status before using the returned values. 230 static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT); 231 232 static unsigned int getInputFramesLost(audio_io_handle_t ioHandle); 233 234 static int newAudioSessionId(); 235 // 236 // AudioPolicyService interface 237 // 238 239 enum audio_devices { 240 // output devices 241 DEVICE_OUT_EARPIECE = DEVICE_EARPIECE, 242 DEVICE_OUT_SPEAKER = DEVICE_SPEAKER, 243 DEVICE_OUT_WIRED_HEADSET = DEVICE_WIRED_HEADSET, 244 DEVICE_OUT_WIRED_HEADPHONE = DEVICE_WIRED_HEADPHONE, 245 DEVICE_OUT_BLUETOOTH_SCO = DEVICE_BLUETOOTH_SCO, 246 DEVICE_OUT_BLUETOOTH_SCO_HEADSET = DEVICE_BLUETOOTH_SCO_HEADSET, 247 DEVICE_OUT_BLUETOOTH_SCO_CARKIT = DEVICE_BLUETOOTH_SCO_CARKIT, 248 DEVICE_OUT_BLUETOOTH_A2DP = DEVICE_BLUETOOTH_A2DP, 249 DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = DEVICE_BLUETOOTH_A2DP_HEADPHONES, 250 DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = DEVICE_BLUETOOTH_A2DP_SPEAKER, 251 DEVICE_OUT_AUX_DIGITAL = DEVICE_AUX_DIGITAL, 252 DEVICE_OUT_DEFAULT = 0x8000, 253 DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET | 254 DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET | 255 DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | 256 DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_DEFAULT), 257 DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | 258 DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), 259 260 // input devices 261 DEVICE_IN_COMMUNICATION = 0x10000, 262 DEVICE_IN_AMBIENT = 0x20000, 263 DEVICE_IN_BUILTIN_MIC = 0x40000, 264 DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000, 265 DEVICE_IN_WIRED_HEADSET = 0x100000, 266 DEVICE_IN_AUX_DIGITAL = 0x200000, 267 DEVICE_IN_VOICE_CALL = 0x400000, 268 DEVICE_IN_BACK_MIC = 0x800000, 269 DEVICE_IN_DEFAULT = 0x80000000, 270 271 DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC | 272 DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL | 273 DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_DEFAULT) 274 }; 275 276 // device connection states used for setDeviceConnectionState() 277 enum device_connection_state { 278 DEVICE_STATE_UNAVAILABLE, 279 DEVICE_STATE_AVAILABLE, 280 NUM_DEVICE_STATES 281 }; 282 283 // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks) 284 enum output_flags { 285 OUTPUT_FLAG_INDIRECT = 0x0, 286 OUTPUT_FLAG_DIRECT = 0x1 287 }; 288 289 // device categories used for setForceUse() 290 enum forced_config { 291 FORCE_NONE, 292 FORCE_SPEAKER, 293 FORCE_HEADPHONES, 294 FORCE_BT_SCO, 295 FORCE_BT_A2DP, 296 FORCE_WIRED_ACCESSORY, 297 FORCE_BT_CAR_DOCK, 298 FORCE_BT_DESK_DOCK, 299 NUM_FORCE_CONFIG, 300 FORCE_DEFAULT = FORCE_NONE 301 }; 302 303 // usages used for setForceUse() 304 enum force_use { 305 FOR_COMMUNICATION, 306 FOR_MEDIA, 307 FOR_RECORD, 308 FOR_DOCK, 309 NUM_FORCE_USE 310 }; 311 312 // types of io configuration change events received with ioConfigChanged() 313 enum io_config_event { 314 OUTPUT_OPENED, 315 OUTPUT_CLOSED, 316 OUTPUT_CONFIG_CHANGED, 317 INPUT_OPENED, 318 INPUT_CLOSED, 319 INPUT_CONFIG_CHANGED, 320 STREAM_CONFIG_CHANGED, 321 NUM_CONFIG_EVENTS 322 }; 323 324 // audio output descritor used to cache output configurations in client process to avoid frequent calls 325 // through IAudioFlinger 326 class OutputDescriptor { 327 public: 328 OutputDescriptor() 329 : samplingRate(0), format(0), channels(0), frameCount(0), latency(0) {} 330 331 uint32_t samplingRate; 332 int32_t format; 333 int32_t channels; 334 size_t frameCount; 335 uint32_t latency; 336 }; 337 338 // 339 // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) 340 // 341 static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address); 342 static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address); 343 static status_t setPhoneState(int state); 344 static status_t setRingerMode(uint32_t mode, uint32_t mask); 345 static status_t setForceUse(force_use usage, forced_config config); 346 static forced_config getForceUse(force_use usage); 347 static audio_io_handle_t getOutput(stream_type stream, 348 uint32_t samplingRate = 0, 349 uint32_t format = FORMAT_DEFAULT, 350 uint32_t channels = CHANNEL_OUT_STEREO, 351 output_flags flags = OUTPUT_FLAG_INDIRECT); 352 static status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream); 353 static status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream); 354 static void releaseOutput(audio_io_handle_t output); 355 static audio_io_handle_t getInput(int inputSource, 356 uint32_t samplingRate = 0, 357 uint32_t format = FORMAT_DEFAULT, 358 uint32_t channels = CHANNEL_IN_MONO, 359 audio_in_acoustics acoustics = (audio_in_acoustics)0); 360 static status_t startInput(audio_io_handle_t input); 361 static status_t stopInput(audio_io_handle_t input); 362 static void releaseInput(audio_io_handle_t input); 363 static status_t initStreamVolume(stream_type stream, 364 int indexMin, 365 int indexMax); 366 static status_t setStreamVolumeIndex(stream_type stream, int index); 367 static status_t getStreamVolumeIndex(stream_type stream, int *index); 368 369 static const sp<IAudioPolicyService>& get_audio_policy_service(); 370 371 // ---------------------------------------------------------------------------- 372 373 static uint32_t popCount(uint32_t u); 374 static bool isOutputDevice(audio_devices device); 375 static bool isInputDevice(audio_devices device); 376 static bool isA2dpDevice(audio_devices device); 377 static bool isBluetoothScoDevice(audio_devices device); 378 static bool isLowVisibility(stream_type stream); 379 static bool isOutputChannel(uint32_t channel); 380 static bool isInputChannel(uint32_t channel); 381 static bool isValidFormat(uint32_t format); 382 static bool isLinearPCM(uint32_t format); 383 384private: 385 386 class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient 387 { 388 public: 389 AudioFlingerClient() { 390 } 391 392 // DeathRecipient 393 virtual void binderDied(const wp<IBinder>& who); 394 395 // IAudioFlingerClient 396 397 // indicate a change in the configuration of an output or input: keeps the cached 398 // values for output/input parameters upto date in client process 399 virtual void ioConfigChanged(int event, int ioHandle, void *param2); 400 }; 401 402 class AudioPolicyServiceClient: public IBinder::DeathRecipient 403 { 404 public: 405 AudioPolicyServiceClient() { 406 } 407 408 // DeathRecipient 409 virtual void binderDied(const wp<IBinder>& who); 410 }; 411 412 static sp<AudioFlingerClient> gAudioFlingerClient; 413 static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; 414 friend class AudioFlingerClient; 415 friend class AudioPolicyServiceClient; 416 417 static Mutex gLock; 418 static sp<IAudioFlinger> gAudioFlinger; 419 static audio_error_callback gAudioErrorCallback; 420 421 static size_t gInBuffSize; 422 // previous parameters for recording buffer size queries 423 static uint32_t gPrevInSamplingRate; 424 static int gPrevInFormat; 425 static int gPrevInChannelCount; 426 427 static sp<IAudioPolicyService> gAudioPolicyService; 428 429 // mapping between stream types and outputs 430 static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap; 431 // list of output descritor containing cached parameters (sampling rate, framecount, channel count...) 432 static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; 433}; 434 435class AudioParameter { 436 437public: 438 AudioParameter() {} 439 AudioParameter(const String8& keyValuePairs); 440 virtual ~AudioParameter(); 441 442 // reserved parameter keys for changeing standard parameters with setParameters() function. 443 // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input 444 // configuration changes and act accordingly. 445 // keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices 446 // keySamplingRate: to change sampling rate routing, value is an int 447 // keyFormat: to change audio format, value is an int in AudioSystem::audio_format 448 // keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels 449 // keyFrameCount: to change audio output frame count, value is an int 450 static const char *keyRouting; 451 static const char *keySamplingRate; 452 static const char *keyFormat; 453 static const char *keyChannels; 454 static const char *keyFrameCount; 455 456 String8 toString(); 457 458 status_t add(const String8& key, const String8& value); 459 status_t addInt(const String8& key, const int value); 460 status_t addFloat(const String8& key, const float value); 461 462 status_t remove(const String8& key); 463 464 status_t get(const String8& key, String8& value); 465 status_t getInt(const String8& key, int& value); 466 status_t getFloat(const String8& key, float& value); 467 status_t getAt(size_t index, String8& key, String8& value); 468 469 size_t size() { return mParameters.size(); } 470 471private: 472 String8 mKeyValuePairs; 473 KeyedVector <String8, String8> mParameters; 474}; 475 476}; // namespace android 477 478#endif /*ANDROID_AUDIOSYSTEM_H_*/ 479