/* * Copyright (C) 2008 The Android Open Source Project * Copyright (c) 2012-2013, The Linux Foundation. All rights reserved. * * Not a Contribution, Apache license notifications and license are retained * for attribution purposes only. * * 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 #include #include /* XXX: Should be include by all the users instead */ #include namespace android { typedef void (*audio_error_callback)(status_t err); class IAudioPolicyService; class String8; class AudioSystem { public: /* 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(audio_stream_type_t stream, float value, audio_io_handle_t output); static status_t getStreamVolume(audio_stream_type_t stream, float* volume, audio_io_handle_t output); // mute/unmute stream static status_t setStreamMute(audio_stream_type_t stream, bool mute); static status_t getStreamMute(audio_stream_type_t stream, bool* mute); // set audio mode in audio hardware static status_t setMode(audio_mode_t mode); // returns true in *state if tracks are active on the specified stream or has been active // in the past inPastMs milliseconds static status_t isStreamActive(audio_stream_type_t stream, bool *state, uint32_t inPastMs = 0); // returns true in *state if a recorder is currently recording with the specified source static status_t isSourceActive(audio_source_t source, 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, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); static status_t getOutputFrameCount(int* frameCount, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); static status_t getOutputLatency(uint32_t* latency, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); static status_t getSamplingRate(audio_io_handle_t output, audio_stream_type_t streamType, int* samplingRate); // returns the number of frames per audio HAL write buffer. Corresponds to // audio_stream->get_buffer_size()/audio_stream_frame_size() static status_t getFrameCount(audio_io_handle_t output, audio_stream_type_t stream, int* frameCount); // returns the audio output stream latency in ms. Corresponds to // audio_stream_out->get_latency() static status_t getLatency(audio_io_handle_t output, audio_stream_type_t stream, uint32_t* latency); // DEPRECATED static status_t getOutputSamplingRate(int* samplingRate, int stream = AUDIO_STREAM_DEFAULT); // DEPRECATED static status_t getOutputFrameCount(int* frameCount, int stream = AUDIO_STREAM_DEFAULT); static bool routedToA2dpOutput(audio_stream_type_t streamType); static status_t getInputBufferSize(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask, size_t* buffSize); static status_t setVoiceVolume(float volume); #ifdef QCOM_FM_ENABLED static status_t setFmVolume(float volume); #endif // return the number of audio frames written by AudioFlinger to audio HAL and // audio dsp to DAC since the output on which the specified 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, audio_stream_type_t stream = AUDIO_STREAM_DEFAULT); // return the number of input frames lost by HAL implementation, or 0 if the handle is invalid static unsigned int getInputFramesLost(audio_io_handle_t ioHandle); static int newAudioSessionId(); static void acquireAudioSessionId(int audioSession); static void releaseAudioSessionId(int audioSession); // 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, #ifdef QCOM_HARDWARE A2DP_OUTPUT_STATE, EFFECT_CONFIG_CHANGED, #endif NUM_CONFIG_EVENTS }; // audio output descriptor used to cache output configurations in client process to avoid frequent calls // through IAudioFlinger class OutputDescriptor { public: OutputDescriptor() : samplingRate(0), format(AUDIO_FORMAT_DEFAULT), channels(0), frameCount(0), latency(0) {} uint32_t samplingRate; int32_t format; int32_t channels; size_t frameCount; uint32_t latency; }; // Events used to synchronize actions between audio sessions. // For instance SYNC_EVENT_PRESENTATION_COMPLETE can be used to delay recording start until playback // is complete on another audio session. // See definitions in MediaSyncEvent.java enum sync_event_t { SYNC_EVENT_SAME = -1, // used internally to indicate restart with same event SYNC_EVENT_NONE = 0, SYNC_EVENT_PRESENTATION_COMPLETE, // // Define new events here: SYNC_EVENT_START, SYNC_EVENT_STOP, SYNC_EVENT_TIME ... // SYNC_EVENT_CNT, }; // Timeout for synchronous record start. Prevents from blocking the record thread forever // if the trigger event is not fired. static const uint32_t kSyncRecordStartTimeOutMs = 30000; // // IAudioPolicyService interface (see AudioPolicyInterface for method descriptions) // static status_t setDeviceConnectionState(audio_devices_t device, audio_policy_dev_state_t state, const char *device_address); static audio_policy_dev_state_t getDeviceConnectionState(audio_devices_t device, const char *device_address); static status_t setPhoneState(audio_mode_t state); static status_t setForceUse(audio_policy_force_use_t usage, audio_policy_forced_cfg_t config); static audio_policy_forced_cfg_t getForceUse(audio_policy_force_use_t usage); static audio_io_handle_t getOutput(audio_stream_type_t stream, uint32_t samplingRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = AUDIO_CHANNEL_OUT_STEREO, audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE); static status_t startOutput(audio_io_handle_t output, audio_stream_type_t stream, int session = 0); static status_t stopOutput(audio_io_handle_t output, audio_stream_type_t stream, int session = 0); static void releaseOutput(audio_io_handle_t output); static audio_io_handle_t getInput(audio_source_t inputSource, uint32_t samplingRate = 0, audio_format_t format = AUDIO_FORMAT_DEFAULT, audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_MONO, int sessionId = 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(audio_stream_type_t stream, int indexMin, int indexMax); static status_t setStreamVolumeIndex(audio_stream_type_t stream, int index, audio_devices_t device); static status_t getStreamVolumeIndex(audio_stream_type_t stream, int *index, audio_devices_t device); static uint32_t getStrategyForStream(audio_stream_type_t stream); static audio_devices_t getDevicesForStream(audio_stream_type_t stream); static audio_io_handle_t getOutputForEffect(const effect_descriptor_t *desc); static status_t registerEffect(const effect_descriptor_t *desc, audio_io_handle_t io, uint32_t strategy, int session, int id); static status_t unregisterEffect(int id); static status_t setEffectEnabled(int id, bool enabled); // clear stream to output mapping cache (gStreamOutputMap) // and output configuration cache (gOutputs) static void clearAudioConfigCache(); static const sp& get_audio_policy_service(); // helpers for android.media.AudioManager.getProperty(), see description there for meaning static int32_t getPrimaryOutputSamplingRate(); static int32_t getPrimaryOutputFrameCount(); // ---------------------------------------------------------------------------- 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 up-to-date in client process virtual void ioConfigChanged(int event, audio_io_handle_t ioHandle, const 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 audio_format_t gPrevInFormat; static audio_channel_mask_t gPrevInChannelMask; static sp gAudioPolicyService; // mapping between stream types and outputs static DefaultKeyedVector gStreamOutputMap; // list of output descriptors containing cached parameters // (sampling rate, framecount, channel count...) static DefaultKeyedVector gOutputs; }; }; // namespace android #endif /*ANDROID_AUDIOSYSTEM_H_*/