| /* |
| * 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> |
| |
| #include <system/audio.h> |
| #include <system/audio_policy.h> |
| |
| /* XXX: Should be include by all the users instead */ |
| #include <media/AudioParameter.h> |
| |
| 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); |
| |
| // 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, 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); |
| |
| // 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); |
| |
| 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, |
| 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, |
| uint32_t channels = 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, |
| uint32_t channels = 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(effect_descriptor_t *desc); |
| static status_t registerEffect(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<IAudioPolicyService>& get_audio_policy_service(); |
| |
| // ---------------------------------------------------------------------------- |
| |
| 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 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<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 audio_format_t gPrevInFormat; |
| static audio_channel_mask_t gPrevInChannelMask; |
| |
| static sp<IAudioPolicyService> gAudioPolicyService; |
| |
| // mapping between stream types and outputs |
| static DefaultKeyedVector<audio_stream_type_t, audio_io_handle_t> gStreamOutputMap; |
| // list of output descriptors containing cached parameters |
| // (sampling rate, framecount, channel count...) |
| static DefaultKeyedVector<audio_io_handle_t, OutputDescriptor *> gOutputs; |
| }; |
| |
| }; // namespace android |
| |
| #endif /*ANDROID_AUDIOSYSTEM_H_*/ |