/* * Copyright (C) 2008 The Android Open Source Project * Copyright (c) 2009-2011, The Linux Foundation. All rights reserved. * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ #ifndef ANDROID_AUDIOSYSTEM_H_ #define ANDROID_AUDIOSYSTEM_H_ #include #include #include namespace android { typedef void (*audio_error_callback)(status_t err); typedef int audio_io_handle_t; class IAudioPolicyService; class String8; class AudioSystem { public: enum stream_type { DEFAULT =-1, VOICE_CALL = 0, SYSTEM = 1, RING = 2, MUSIC = 3, ALARM = 4, NOTIFICATION = 5, BLUETOOTH_SCO = 6, ENFORCED_AUDIBLE = 7, // Sounds that cannot be muted by user and must be routed to speaker DTMF = 8, TTS = 9, FM = 10, NUM_STREAM_TYPES }; // Audio sub formats (see AudioSystem::audio_format). enum pcm_sub_format { PCM_SUB_16_BIT = 0x1, // must be 1 for backward compatibility PCM_SUB_8_BIT = 0x2, // must be 2 for backward compatibility }; // MP3 sub format field definition : can use 11 LSBs in the same way as MP3 frame header to specify // bit rate, stereo mode, version... enum mp3_sub_format { //TODO }; // AMR NB/WB sub format field definition: specify frame block interleaving, bandwidth efficient or octet aligned, // encoding mode for recording... enum amr_sub_format { //TODO }; // AAC sub format field definition: specify profile or bitrate for recording... enum aac_sub_format { //TODO }; // VORBIS sub format field definition: specify quality for recording... enum vorbis_sub_format { //TODO }; // Audio format consists in a main format field (upper 8 bits) and a sub format field (lower 24 bits). // The main format indicates the main codec type. The sub format field indicates options and parameters // for each format. The sub format is mainly used for record to indicate for instance the requested bitrate // or profile. It can also be used for certain formats to give informations not present in the encoded // audio stream (e.g. octet alignement for AMR). enum audio_format { INVALID_FORMAT = -1, FORMAT_DEFAULT = 0, PCM = 0x00000000, // must be 0 for backward compatibility MP3 = 0x01000000, AMR_NB = 0x02000000, AMR_WB = 0x03000000, AAC = 0x04000000, HE_AAC_V1 = 0x05000000, HE_AAC_V2 = 0x06000000, VORBIS = 0x07000000, EVRC = 0x08000000, QCELP = 0x09000000, VOIP_PCM_INPUT = 0x0A000000, MAIN_FORMAT_MASK = 0xFF000000, SUB_FORMAT_MASK = 0x00FFFFFF, // Aliases PCM_16_BIT = (PCM|PCM_SUB_16_BIT), PCM_8_BIT = (PCM|PCM_SUB_8_BIT) }; // Channel mask definitions must be kept in sync with JAVA values in /media/java/android/media/AudioFormat.java enum audio_channels { // output channels CHANNEL_OUT_FRONT_LEFT = 0x4, CHANNEL_OUT_FRONT_RIGHT = 0x8, CHANNEL_OUT_FRONT_CENTER = 0x10, CHANNEL_OUT_LOW_FREQUENCY = 0x20, CHANNEL_OUT_BACK_LEFT = 0x40, CHANNEL_OUT_BACK_RIGHT = 0x80, CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 0x100, CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200, CHANNEL_OUT_BACK_CENTER = 0x400, CHANNEL_OUT_MONO = CHANNEL_OUT_FRONT_LEFT, CHANNEL_OUT_STEREO = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT), CHANNEL_OUT_QUAD = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT), CHANNEL_OUT_SURROUND = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_BACK_CENTER), CHANNEL_OUT_5POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT), CHANNEL_OUT_7POINT1 = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER), CHANNEL_OUT_ALL = (CHANNEL_OUT_FRONT_LEFT | CHANNEL_OUT_FRONT_RIGHT | CHANNEL_OUT_FRONT_CENTER | CHANNEL_OUT_LOW_FREQUENCY | CHANNEL_OUT_BACK_LEFT | CHANNEL_OUT_BACK_RIGHT | CHANNEL_OUT_FRONT_LEFT_OF_CENTER | CHANNEL_OUT_FRONT_RIGHT_OF_CENTER | CHANNEL_OUT_BACK_CENTER), // input channels CHANNEL_IN_LEFT = 0x4, CHANNEL_IN_RIGHT = 0x8, CHANNEL_IN_FRONT = 0x10, CHANNEL_IN_BACK = 0x20, CHANNEL_IN_LEFT_PROCESSED = 0x40, CHANNEL_IN_RIGHT_PROCESSED = 0x80, CHANNEL_IN_FRONT_PROCESSED = 0x100, CHANNEL_IN_BACK_PROCESSED = 0x200, CHANNEL_IN_PRESSURE = 0x400, CHANNEL_IN_X_AXIS = 0x800, CHANNEL_IN_Y_AXIS = 0x1000, CHANNEL_IN_Z_AXIS = 0x2000, CHANNEL_IN_VOICE_UPLINK = 0x4000, CHANNEL_IN_VOICE_DNLINK = 0x8000, CHANNEL_IN_MONO = CHANNEL_IN_FRONT, CHANNEL_IN_STEREO = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT), CHANNEL_IN_ALL = (CHANNEL_IN_LEFT | CHANNEL_IN_RIGHT | CHANNEL_IN_FRONT | CHANNEL_IN_BACK| CHANNEL_IN_LEFT_PROCESSED | CHANNEL_IN_RIGHT_PROCESSED | CHANNEL_IN_FRONT_PROCESSED | CHANNEL_IN_BACK_PROCESSED| CHANNEL_IN_PRESSURE | CHANNEL_IN_X_AXIS | CHANNEL_IN_Y_AXIS | CHANNEL_IN_Z_AXIS | CHANNEL_IN_VOICE_UPLINK | CHANNEL_IN_VOICE_DNLINK) }; enum audio_mode { MODE_INVALID = -2, MODE_CURRENT = -1, MODE_NORMAL = 0, MODE_RINGTONE, MODE_IN_CALL, MODE_IN_COMMUNICATION, NUM_MODES // not a valid entry, denotes end-of-list }; enum audio_in_acoustics { AGC_ENABLE = 0x0001, AGC_DISABLE = 0, NS_ENABLE = 0x0002, NS_DISABLE = 0, TX_IIR_ENABLE = 0x0004, TX_DISABLE = 0 }; // special audio session values enum audio_sessions { SESSION_OUTPUT_STAGE = -1, // session for effects attached to a particular output stream // (value must be less than 0) SESSION_OUTPUT_MIX = 0, // session for effects applied to output mix. These effects can // be moved by audio policy manager to another output stream // (value must be 0) }; /* These are static methods to control the system-wide AudioFlinger * only privileged processes can have access to them */ // mute/unmute microphone static status_t muteMicrophone(bool state); static status_t isMicrophoneMuted(bool *state); // set/get master volume static status_t setMasterVolume(float value); static status_t getMasterVolume(float* volume); // mute/unmute audio outputs static status_t setMasterMute(bool mute); static status_t getMasterMute(bool* mute); // set/get stream volume on specified output static status_t setStreamVolume(int stream, float value, int output); static status_t getStreamVolume(int stream, float* volume, int output); // mute/unmute stream static status_t setStreamMute(int stream, bool mute); static status_t getStreamMute(int stream, bool* mute); // set audio mode in audio hardware (see AudioSystem::audio_mode) static status_t setMode(int mode); // returns true in *state if tracks are active on the specified stream static status_t isStreamActive(int stream, bool *state); // set/get audio hardware parameters. The function accepts a list of parameters // key value pairs in the form: key1=value1;key2=value2;... // Some keys are reserved for standard parameters (See AudioParameter class). static status_t setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs); static String8 getParameters(audio_io_handle_t ioHandle, const String8& keys); static void setErrorCallback(audio_error_callback cb); // helper function to obtain AudioFlinger service handle static const sp& get_audio_flinger(); static float linearToLog(int volume); static int logToLinear(float volume); static status_t getOutputSamplingRate(int* samplingRate, int stream = DEFAULT); static status_t getOutputFrameCount(int* frameCount, int stream = DEFAULT); static status_t getOutputLatency(uint32_t* latency, int stream = DEFAULT); static bool routedToA2dpOutput(int streamType); static status_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount, size_t* buffSize); static status_t setVoiceVolume(float volume); static status_t setFmVolume(float volume); // return the number of audio frames written by AudioFlinger to audio HAL and // audio dsp to DAC since the output on which the specificed stream is playing // has exited standby. // returned status (from utils/Errors.h) can be: // - NO_ERROR: successful operation, halFrames and dspFrames point to valid data // - INVALID_OPERATION: Not supported on current hardware platform // - BAD_VALUE: invalid parameter // NOTE: this feature is not supported on all hardware platforms and it is // necessary to check returned status before using the returned values. static status_t getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames, int stream = DEFAULT); static unsigned int getInputFramesLost(audio_io_handle_t ioHandle); static int newAudioSessionId(); // // AudioPolicyService interface // enum audio_devices { // output devices DEVICE_OUT_EARPIECE = 0x1, DEVICE_OUT_SPEAKER = 0x2, DEVICE_OUT_WIRED_HEADSET = 0x4, DEVICE_OUT_WIRED_HEADPHONE = 0x8, DEVICE_OUT_BLUETOOTH_SCO = 0x10, DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 0x20, DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 0x40, DEVICE_OUT_BLUETOOTH_A2DP = 0x80, DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100, DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 0x200, DEVICE_OUT_AUX_DIGITAL = 0x400, DEVICE_OUT_FM = 0x800, DEVICE_OUT_ANC_HEADSET = 0x1000, DEVICE_OUT_ANC_HEADPHONE = 0x2000, DEVICE_OUT_FM_TX = 0x4000, DEVICE_OUT_DEFAULT = 0x8000, // Since no free bits available in output device , using free bits from input device list DEVICE_OUT_DIRECTOUTPUT = 0x8000000, DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET | DEVICE_OUT_WIRED_HEADPHONE | DEVICE_OUT_FM | DEVICE_OUT_BLUETOOTH_SCO | DEVICE_OUT_BLUETOOTH_SCO_HEADSET | DEVICE_OUT_BLUETOOTH_SCO_CARKIT | DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER | DEVICE_OUT_AUX_DIGITAL | DEVICE_OUT_FM | DEVICE_OUT_ANC_HEADSET | DEVICE_OUT_ANC_HEADPHONE | DEVICE_OUT_FM_TX | DEVICE_OUT_DIRECTOUTPUT | DEVICE_OUT_DEFAULT), DEVICE_OUT_ALL_A2DP = (DEVICE_OUT_BLUETOOTH_A2DP | DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES | DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER), // input devices DEVICE_IN_COMMUNICATION = 0x10000, DEVICE_IN_AMBIENT = 0x20000, DEVICE_IN_BUILTIN_MIC = 0x40000, DEVICE_IN_VOICE_CALL = 0x80000, DEVICE_IN_BACK_MIC = 0x100000, DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x200000, DEVICE_IN_WIRED_HEADSET = 0x400000, DEVICE_IN_AUX_DIGITAL = 0x800000, DEVICE_IN_ANC_HEADSET = 0x1000000, DEVICE_IN_FM_RX = 0x2000000, DEVICE_IN_FM_RX_A2DP = 0x4000000, DEVICE_IN_DEFAULT = 0x80000000, DEVICE_IN_ALL = (DEVICE_IN_COMMUNICATION | DEVICE_IN_AMBIENT | DEVICE_IN_BUILTIN_MIC | DEVICE_IN_BLUETOOTH_SCO_HEADSET | DEVICE_IN_WIRED_HEADSET | DEVICE_IN_AUX_DIGITAL | DEVICE_IN_VOICE_CALL | DEVICE_IN_BACK_MIC | DEVICE_IN_FM_RX | DEVICE_IN_FM_RX_A2DP | DEVICE_IN_ANC_HEADSET | DEVICE_IN_DEFAULT) }; // device connection states used for setDeviceConnectionState() enum device_connection_state { DEVICE_STATE_UNAVAILABLE, DEVICE_STATE_AVAILABLE, NUM_DEVICE_STATES }; // request to open a direct output with getOutput() (by opposition to sharing an output with other AudioTracks) enum output_flags { OUTPUT_FLAG_INDIRECT = 0x0, OUTPUT_FLAG_DIRECT = 0x1, OUTPUT_FLAG_SESSION = 0x2 }; // device categories used for setForceUse() enum forced_config { FORCE_NONE, FORCE_SPEAKER, FORCE_HEADPHONES, FORCE_BT_SCO, FORCE_BT_A2DP, FORCE_WIRED_ACCESSORY, FORCE_BT_CAR_DOCK, FORCE_BT_DESK_DOCK, NUM_FORCE_CONFIG, FORCE_DEFAULT = FORCE_NONE }; // usages used for setForceUse() enum force_use { FOR_COMMUNICATION, FOR_MEDIA, FOR_RECORD, FOR_DOCK, NUM_FORCE_USE }; // types of io configuration change events received with ioConfigChanged() enum io_config_event { OUTPUT_OPENED, OUTPUT_CLOSED, OUTPUT_CONFIG_CHANGED, INPUT_OPENED, INPUT_CLOSED, INPUT_CONFIG_CHANGED, STREAM_CONFIG_CHANGED, A2DP_OUTPUT_STATE, EFFECT_CONFIG_CHANGED, NUM_CONFIG_EVENTS }; // audio output descritor used to cache output configurations in client process to avoid frequent calls // through IAudioFlinger class OutputDescriptor { public: OutputDescriptor() : samplingRate(0), format(0), channels(0), frameCount(0), latency(0) {} uint32_t samplingRate; int32_t format; int32_t channels; size_t frameCount; uint32_t latency; }; // // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) // static status_t setDeviceConnectionState(audio_devices device, device_connection_state state, const char *device_address); static device_connection_state getDeviceConnectionState(audio_devices device, const char *device_address); static status_t setPhoneState(int state); static status_t setRingerMode(uint32_t mode, uint32_t mask); static status_t setForceUse(force_use usage, forced_config config); static forced_config getForceUse(force_use usage); static audio_io_handle_t getOutput(stream_type stream, uint32_t samplingRate = 0, uint32_t format = FORMAT_DEFAULT, uint32_t channels = CHANNEL_OUT_STEREO, output_flags flags = OUTPUT_FLAG_INDIRECT); static audio_io_handle_t getSession(stream_type stream, uint32_t format = FORMAT_DEFAULT, output_flags flags = OUTPUT_FLAG_DIRECT, int32_t sessionId = -1); static void closeSession(audio_io_handle_t output); static status_t pauseSession(audio_io_handle_t output, stream_type stream); static status_t resumeSession(audio_io_handle_t output, stream_type stream); static status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream, int session = 0); static status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream, int session = 0); static void releaseOutput(audio_io_handle_t output); static audio_io_handle_t getInput(int inputSource, uint32_t samplingRate = 0, uint32_t format = FORMAT_DEFAULT, uint32_t channels = CHANNEL_IN_MONO, audio_in_acoustics acoustics = (audio_in_acoustics)0); static status_t startInput(audio_io_handle_t input); static status_t stopInput(audio_io_handle_t input); static void releaseInput(audio_io_handle_t input); static status_t initStreamVolume(stream_type stream, int indexMin, int indexMax); static status_t setStreamVolumeIndex(stream_type stream, int index); static status_t getStreamVolumeIndex(stream_type stream, int *index); static uint32_t getStrategyForStream(stream_type stream); static audio_io_handle_t getOutputForEffect(effect_descriptor_t *desc); static status_t registerEffect(effect_descriptor_t *desc, audio_io_handle_t output, uint32_t strategy, int session, int id); static status_t unregisterEffect(int id); static const sp& get_audio_policy_service(); // ---------------------------------------------------------------------------- static uint32_t popCount(uint32_t u); static bool isOutputDevice(audio_devices device); static bool isInputDevice(audio_devices device); static bool isA2dpDevice(audio_devices device); static bool isBluetoothScoDevice(audio_devices device); static bool isLowVisibility(stream_type stream); static bool isOutputChannel(uint32_t channel); static bool isInputChannel(uint32_t channel); static bool isValidFormat(uint32_t format); static bool isLinearPCM(uint32_t format); static bool isModeInCall(); private: class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient { public: AudioFlingerClient() { } // DeathRecipient virtual void binderDied(const wp& who); // IAudioFlingerClient // indicate a change in the configuration of an output or input: keeps the cached // values for output/input parameters upto date in client process virtual void ioConfigChanged(int event, int ioHandle, void *param2); }; class AudioPolicyServiceClient: public IBinder::DeathRecipient { public: AudioPolicyServiceClient() { } // DeathRecipient virtual void binderDied(const wp& who); }; static sp gAudioFlingerClient; static sp gAudioPolicyServiceClient; friend class AudioFlingerClient; friend class AudioPolicyServiceClient; static Mutex gLock; static sp gAudioFlinger; static audio_error_callback gAudioErrorCallback; static size_t gInBuffSize; // previous parameters for recording buffer size queries static uint32_t gPrevInSamplingRate; static int gPrevInFormat; static int gPrevInChannelCount; static int gPhoneState; static sp gAudioPolicyService; // mapping between stream types and outputs static DefaultKeyedVector gStreamOutputMap; // list of output descritor containing cached parameters (sampling rate, framecount, channel count...) static DefaultKeyedVector gOutputs; }; class AudioParameter { public: AudioParameter() {} AudioParameter(const String8& keyValuePairs); virtual ~AudioParameter(); // reserved parameter keys for changing standard parameters with setParameters() function. // Using these keys is mandatory for AudioFlinger to properly monitor audio output/input // configuration changes and act accordingly. // keyRouting: to change audio routing, value is an int in AudioSystem::audio_devices // keySamplingRate: to change sampling rate routing, value is an int // keyFormat: to change audio format, value is an int in AudioSystem::audio_format // keyChannels: to change audio channel configuration, value is an int in AudioSystem::audio_channels // keyFrameCount: to change audio output frame count, value is an int // keyInputSource: to change audio input source, value is an int in audio_source // (defined in media/mediarecorder.h) static const char *keyRouting; static const char *keySamplingRate; static const char *keyFormat; static const char *keyChannels; static const char *keyFrameCount; static const char *keyInputSource; String8 toString(); status_t add(const String8& key, const String8& value); status_t addInt(const String8& key, const int value); status_t addFloat(const String8& key, const float value); status_t remove(const String8& key); status_t get(const String8& key, String8& value); status_t getInt(const String8& key, int& value); status_t getFloat(const String8& key, float& value); status_t getAt(size_t index, String8& key, String8& value); size_t size() { return mParameters.size(); } private: String8 mKeyValuePairs; KeyedVector mParameters; }; }; // namespace android #endif /*ANDROID_AUDIOSYSTEM_H_*/