| /* |
| * Copyright (C) 2008 The Android Open Source Project |
| * |
| * 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 <utils/RefBase.h> |
| #include <utils/threads.h> |
| #include <media/IAudioFlinger.h> |
| |
| 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, |
| 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, |
| 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, |
| 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 |
| }; |
| |
| /* 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<IAudioFlinger>& 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); |
| |
| // 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); |
| // |
| // 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_DEFAULT = 0x8000, |
| DEVICE_OUT_ALL = (DEVICE_OUT_EARPIECE | DEVICE_OUT_SPEAKER | DEVICE_OUT_WIRED_HEADSET | |
| DEVICE_OUT_WIRED_HEADPHONE | 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_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_BLUETOOTH_SCO_HEADSET = 0x80000, |
| DEVICE_IN_WIRED_HEADSET = 0x100000, |
| DEVICE_IN_AUX_DIGITAL = 0x200000, |
| DEVICE_IN_VOICE_CALL = 0x400000, |
| DEVICE_IN_BACK_MIC = 0x800000, |
| 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_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 |
| }; |
| |
| // 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, |
| 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 status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream); |
| static status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream); |
| 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 const sp<IAudioPolicyService>& 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); |
| |
| private: |
| |
| class AudioFlingerClient: public IBinder::DeathRecipient, public BnAudioFlingerClient |
| { |
| public: |
| AudioFlingerClient() { |
| } |
| |
| // DeathRecipient |
| virtual void binderDied(const wp<IBinder>& 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<IBinder>& who); |
| }; |
| |
| static sp<AudioFlingerClient> gAudioFlingerClient; |
| static sp<AudioPolicyServiceClient> gAudioPolicyServiceClient; |
| friend class AudioFlingerClient; |
| friend class AudioPolicyServiceClient; |
| |
| static Mutex gLock; |
| static sp<IAudioFlinger> 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 sp<IAudioPolicyService> gAudioPolicyService; |
| |
| // mapping between stream types and outputs |
| static DefaultKeyedVector<int, audio_io_handle_t> gStreamOutputMap; |
| // list of output descritor containing cached parameters (sampling rate, framecount, channel count...) |
| static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; |
| }; |
| |
| class AudioParameter { |
| |
| public: |
| AudioParameter() {} |
| AudioParameter(const String8& keyValuePairs); |
| virtual ~AudioParameter(); |
| |
| // reserved parameter keys for changeing 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 |
| static const char *keyRouting; |
| static const char *keySamplingRate; |
| static const char *keyFormat; |
| static const char *keyChannels; |
| static const char *keyFrameCount; |
| |
| 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 <String8, String8> mParameters; |
| }; |
| |
| }; // namespace android |
| |
| #endif /*ANDROID_AUDIOSYSTEM_H_*/ |
