/* * Copyright (C) 2007 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_AUDIO_HARDWARE_INTERFACE_H #define ANDROID_AUDIO_HARDWARE_INTERFACE_H #include #include #include #include #include #include #include #include #include #include #include namespace android_audio_legacy { using android::Vector; using android::String16; using android::String8; // ---------------------------------------------------------------------------- /** * AudioStreamOut is the abstraction interface for the audio output hardware. * * It provides information about various properties of the audio output hardware driver. */ class AudioStreamOut { public: virtual ~AudioStreamOut() = 0; /** return audio sampling rate in hz - eg. 44100 */ virtual uint32_t sampleRate() const = 0; /** returns size of output buffer - eg. 4800 */ virtual size_t bufferSize() const = 0; /** * returns the output channel mask */ virtual uint32_t channels() const = 0; /** * return audio format in 8bit or 16bit PCM format - * eg. AudioSystem:PCM_16_BIT */ virtual int format() const = 0; /** * return the frame size (number of bytes per sample). */ uint32_t frameSize() const { return audio_channel_count_from_out_mask(channels())* ((format()==AUDIO_FORMAT_PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); } /** * return the audio hardware driver latency in milli seconds. */ virtual uint32_t latency() const = 0; /** * Use this method in situations where audio mixing is done in the * hardware. This method serves as a direct interface with hardware, * allowing you to directly set the volume as apposed to via the framework. * This method might produce multiple PCM outputs or hardware accelerated * codecs, such as MP3 or AAC. */ virtual status_t setVolume(float left, float right) = 0; /** write audio buffer to driver. Returns number of bytes written */ virtual ssize_t write(const void* buffer, size_t bytes) = 0; /** * Put the audio hardware output into standby mode. Returns * status based on include/utils/Errors.h */ virtual status_t standby() = 0; /** dump the state of the audio output device */ virtual status_t dump(int fd, const Vector& args) = 0; // set/get audio output 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). // If the implementation does not accept a parameter change while the output is // active but the parameter is acceptable otherwise, it must return INVALID_OPERATION. // The audio flinger will put the output in standby and then change the parameter value. virtual status_t setParameters(const String8& keyValuePairs) = 0; virtual String8 getParameters(const String8& keys) = 0; // return the number of audio frames written by the audio dsp to DAC since // the output has exited standby virtual status_t getRenderPosition(uint32_t *dspFrames) = 0; /** * get the local time at which the next write to the audio driver will be * presented */ virtual status_t getNextWriteTimestamp(int64_t *timestamp); /** * Return a recent count of the number of audio frames presented to an external observer. */ virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp) = 0; }; /** * AudioStreamIn is the abstraction interface for the audio input hardware. * * It defines the various properties of the audio hardware input driver. */ class AudioStreamIn { public: virtual ~AudioStreamIn() = 0; /** return audio sampling rate in hz - eg. 44100 */ virtual uint32_t sampleRate() const = 0; /** return the input buffer size allowed by audio driver */ virtual size_t bufferSize() const = 0; /** return input channel mask */ virtual uint32_t channels() const = 0; /** * return audio format in 8bit or 16bit PCM format - * eg. AudioSystem:PCM_16_BIT */ virtual int format() const = 0; /** * return the frame size (number of bytes per sample). */ uint32_t frameSize() const { return audio_channel_count_from_in_mask(channels())* ((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); } /** set the input gain for the audio driver. This method is for * for future use */ virtual status_t setGain(float gain) = 0; /** read audio buffer in from audio driver */ virtual ssize_t read(void* buffer, ssize_t bytes) = 0; /** dump the state of the audio input device */ virtual status_t dump(int fd, const Vector& args) = 0; /** * Put the audio hardware input into standby mode. Returns * status based on include/utils/Errors.h */ virtual status_t standby() = 0; // set/get audio input 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). // If the implementation does not accept a parameter change while the output is // active but the parameter is acceptable otherwise, it must return INVALID_OPERATION. // The audio flinger will put the input in standby and then change the parameter value. virtual status_t setParameters(const String8& keyValuePairs) = 0; virtual String8 getParameters(const String8& keys) = 0; // Return the number of input frames lost in the audio driver since the last call of this function. // Audio driver is expected to reset the value to 0 and restart counting upon returning the current value by this function call. // Such loss typically occurs when the user space process is blocked longer than the capacity of audio driver buffers. // Unit: the number of input audio frames virtual unsigned int getInputFramesLost() const = 0; virtual status_t addAudioEffect(effect_handle_t effect) = 0; virtual status_t removeAudioEffect(effect_handle_t effect) = 0; }; /** * AudioHardwareInterface.h defines the interface to the audio hardware abstraction layer. * * The interface supports setting and getting parameters, selecting audio routing * paths, and defining input and output streams. * * AudioFlinger initializes the audio hardware and immediately opens an output stream. * You can set Audio routing to output to handset, speaker, Bluetooth, or a headset. * * The audio input stream is initialized when AudioFlinger is called to carry out * a record operation. */ class AudioHardwareInterface { public: virtual ~AudioHardwareInterface() {} /** * check to see if the audio hardware interface has been initialized. * return status based on values defined in include/utils/Errors.h */ virtual status_t initCheck() = 0; /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */ virtual status_t setVoiceVolume(float volume) = 0; /** * set the audio volume for all audio activities other than voice call. * Range between 0.0 and 1.0. If any value other than NO_ERROR is returned, * the software mixer will emulate this capability. */ virtual status_t setMasterVolume(float volume) = 0; /** * Get the current master volume value for the HAL, if the HAL supports * master volume control. AudioFlinger will query this value from the * primary audio HAL when the service starts and use the value for setting * the initial master volume across all HALs. */ virtual status_t getMasterVolume(float *volume) = 0; /** * setMode is called when the audio mode changes. NORMAL mode is for * standard audio playback, RINGTONE when a ringtone is playing, and IN_CALL * when a call is in progress. */ virtual status_t setMode(int mode) = 0; // mic mute virtual status_t setMicMute(bool state) = 0; virtual status_t getMicMute(bool* state) = 0; // set/get global audio parameters virtual status_t setParameters(const String8& keyValuePairs) = 0; virtual String8 getParameters(const String8& keys) = 0; // Returns audio input buffer size according to parameters passed or 0 if one of the // parameters is not supported virtual size_t getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0; /** This method creates and opens the audio hardware output stream */ virtual AudioStreamOut* openOutputStream( uint32_t devices, int *format=0, uint32_t *channels=0, uint32_t *sampleRate=0, status_t *status=0) = 0; virtual AudioStreamOut* openOutputStreamWithFlags( uint32_t devices, audio_output_flags_t flags=(audio_output_flags_t)0, int *format=0, uint32_t *channels=0, uint32_t *sampleRate=0, status_t *status=0) = 0; virtual void closeOutputStream(AudioStreamOut* out) = 0; /** This method creates and opens the audio hardware input stream */ virtual AudioStreamIn* openInputStream( uint32_t devices, int *format, uint32_t *channels, uint32_t *sampleRate, status_t *status, AudioSystem::audio_in_acoustics acoustics) = 0; virtual void closeInputStream(AudioStreamIn* in) = 0; /**This method dumps the state of the audio hardware */ virtual status_t dumpState(int fd, const Vector& args) = 0; virtual status_t setMasterMute(bool muted) = 0; static AudioHardwareInterface* create(); virtual int createAudioPatch(unsigned int num_sources, const struct audio_port_config *sources, unsigned int num_sinks, const struct audio_port_config *sinks, audio_patch_handle_t *handle) = 0; virtual int releaseAudioPatch(audio_patch_handle_t handle) = 0; virtual int getAudioPort(struct audio_port *port) = 0; virtual int setAudioPortConfig(const struct audio_port_config *config) = 0; protected: virtual status_t dump(int fd, const Vector& args) = 0; }; // ---------------------------------------------------------------------------- extern "C" AudioHardwareInterface* createAudioHardware(void); }; // namespace android #endif // ANDROID_AUDIO_HARDWARE_INTERFACE_H