include/hardware/audio.h

/*
 * Copyright (C) 2011 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_HAL_INTERFACE_H
#define ANDROID_AUDIO_HAL_INTERFACE_H

#include <stdint.h>
#include <strings.h>
#include <sys/cdefs.h>
#include <sys/types.h>
#include <time.h>

#include <cutils/bitops.h>

#include <hardware/hardware.h>
#include <system/audio.h>
#include <hardware/audio_effect.h>

__BEGIN_DECLS

/**
 * The id of this module
 */
#define AUDIO_HARDWARE_MODULE_ID "audio"

/**
 * Name of the audio devices to open
 */
#define AUDIO_HARDWARE_INTERFACE "audio_hw_if"


/* Use version 0.1 to be compatible with first generation of audio hw module with version_major
 * hardcoded to 1. No audio module API change.
 */
#define AUDIO_MODULE_API_VERSION_0_1 HARDWARE_MODULE_API_VERSION(0, 1)
#define AUDIO_MODULE_API_VERSION_CURRENT AUDIO_MODULE_API_VERSION_0_1

/* First generation of audio devices had version hardcoded to 0. all devices with versions < 1.0
 * will be considered of first generation API.
 */
#define AUDIO_DEVICE_API_VERSION_0_0 HARDWARE_DEVICE_API_VERSION(0, 0)
#define AUDIO_DEVICE_API_VERSION_1_0 HARDWARE_DEVICE_API_VERSION(1, 0)
#define AUDIO_DEVICE_API_VERSION_2_0 HARDWARE_DEVICE_API_VERSION(2, 0)
#define AUDIO_DEVICE_API_VERSION_3_0 HARDWARE_DEVICE_API_VERSION(3, 0)
#define AUDIO_DEVICE_API_VERSION_3_1 HARDWARE_DEVICE_API_VERSION(3, 1)
#define AUDIO_DEVICE_API_VERSION_3_2 HARDWARE_DEVICE_API_VERSION(3, 2)
#define AUDIO_DEVICE_API_VERSION_CURRENT AUDIO_DEVICE_API_VERSION_3_2
/* Minimal audio HAL version supported by the audio framework */
#define AUDIO_DEVICE_API_VERSION_MIN AUDIO_DEVICE_API_VERSION_2_0

/**************************************/

/**
 *  standard audio parameters that the HAL may need to handle
 */

/**
 *  audio device parameters
 */

/* TTY mode selection */
#define AUDIO_PARAMETER_KEY_TTY_MODE "tty_mode"
#define AUDIO_PARAMETER_VALUE_TTY_OFF "tty_off"
#define AUDIO_PARAMETER_VALUE_TTY_VCO "tty_vco"
#define AUDIO_PARAMETER_VALUE_TTY_HCO "tty_hco"
#define AUDIO_PARAMETER_VALUE_TTY_FULL "tty_full"

/* Hearing Aid Compatibility - Telecoil (HAC-T) mode on/off */
#define AUDIO_PARAMETER_KEY_HAC "HACSetting"
#define AUDIO_PARAMETER_VALUE_HAC_ON "ON"
#define AUDIO_PARAMETER_VALUE_HAC_OFF "OFF"

/* A2DP sink address set by framework */
#define AUDIO_PARAMETER_A2DP_SINK_ADDRESS "a2dp_sink_address"

/* A2DP source address set by framework */
#define AUDIO_PARAMETER_A2DP_SOURCE_ADDRESS "a2dp_source_address"

/* Bluetooth SCO wideband */
#define AUDIO_PARAMETER_KEY_BT_SCO_WB "bt_wbs"

/* BT SCO headset name for debug */
#define AUDIO_PARAMETER_KEY_BT_SCO_HEADSET_NAME "bt_headset_name"

/* BT SCO HFP control */
#define AUDIO_PARAMETER_KEY_HFP_ENABLE            "hfp_enable"
#define AUDIO_PARAMETER_KEY_HFP_SET_SAMPLING_RATE "hfp_set_sampling_rate"
#define AUDIO_PARAMETER_KEY_HFP_VOLUME            "hfp_volume"

/* Set screen orientation */
#define AUDIO_PARAMETER_KEY_ROTATION "rotation"

/**
 *  audio stream parameters
 */

/* Enable AANC */
#define AUDIO_PARAMETER_KEY_AANC "aanc_enabled"

/**************************************/

/* common audio stream parameters and operations */
struct audio_stream {

    /**
     * Return the sampling rate in Hz - eg. 44100.
     */
    uint32_t (*get_sample_rate)(const struct audio_stream *stream);

    /* currently unused - use set_parameters with key
     *    AUDIO_PARAMETER_STREAM_SAMPLING_RATE
     */
    int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate);

    /**
     * Return size of input/output buffer in bytes for this stream - eg. 4800.
     * It should be a multiple of the frame size.  See also get_input_buffer_size.
     */
    size_t (*get_buffer_size)(const struct audio_stream *stream);

    /**
     * Return the channel mask -
     *  e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
     */
    audio_channel_mask_t (*get_channels)(const struct audio_stream *stream);

    /**
     * Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT
     */
    audio_format_t (*get_format)(const struct audio_stream *stream);

    /* currently unused - use set_parameters with key
     *     AUDIO_PARAMETER_STREAM_FORMAT
     */
    int (*set_format)(struct audio_stream *stream, audio_format_t format);

    /**
     * Put the audio hardware input/output into standby mode.
     * Driver should exit from standby mode at the next I/O operation.
     * Returns 0 on success and <0 on failure.
     */
    int (*standby)(struct audio_stream *stream);

    /** dump the state of the audio input/output device */
    int (*dump)(const struct audio_stream *stream, int fd);

    /** Return the set of device(s) which this stream is connected to */
    audio_devices_t (*get_device)(const struct audio_stream *stream);

    /**
     * Currently unused - set_device() corresponds to set_parameters() with key
     * AUDIO_PARAMETER_STREAM_ROUTING for both input and output.
     * AUDIO_PARAMETER_STREAM_INPUT_SOURCE is an additional information used by
     * input streams only.
     */
    int (*set_device)(struct audio_stream *stream, audio_devices_t device);

    /**
     * set/get audio stream parameters. The function accepts a list of
     * parameter 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 -ENOSYS.
     *
     * The audio flinger will put the stream in standby and then change the
     * parameter value.
     */
    int (*set_parameters)(struct audio_stream *stream, const char *kv_pairs);

    /*
     * Returns a pointer to a heap allocated string. The caller is responsible
     * for freeing the memory for it using free().
     */
    char * (*get_parameters)(const struct audio_stream *stream,
                             const char *keys);
    int (*add_audio_effect)(const struct audio_stream *stream,
                             effect_handle_t effect);
    int (*remove_audio_effect)(const struct audio_stream *stream,
                             effect_handle_t effect);
};
typedef struct audio_stream audio_stream_t;

/* type of asynchronous write callback events. Mutually exclusive */
typedef enum {
    STREAM_CBK_EVENT_WRITE_READY, /* non blocking write completed */
    STREAM_CBK_EVENT_DRAIN_READY,  /* drain completed */
    STREAM_CBK_EVENT_ERROR, /* stream hit some error, let AF take action */
} stream_callback_event_t;

typedef enum {
    STREAM_EVENT_CBK_TYPE_CODEC_FORMAT_CHANGED, /* codec format of the stream changed */
} stream_event_callback_type_t;

typedef int (*stream_callback_t)(stream_callback_event_t event, void *param, void *cookie);

typedef int (*stream_event_callback_t)(stream_event_callback_type_t event,
                                       void *param, void *cookie);

/* type of drain requested to audio_stream_out->drain(). Mutually exclusive */
typedef enum {
    AUDIO_DRAIN_ALL,            /* drain() returns when all data has been played */
    AUDIO_DRAIN_EARLY_NOTIFY    /* drain() returns a short time before all data
                                   from the current track has been played to
                                   give time for gapless track switch */
} audio_drain_type_t;

typedef struct source_metadata {
    size_t track_count;
    /** Array of metadata of each track connected to this source. */
    struct playback_track_metadata* tracks;
} source_metadata_t;

typedef struct sink_metadata {
    size_t track_count;
    /** Array of metadata of each track connected to this sink. */
    struct record_track_metadata* tracks;
} sink_metadata_t;

/* HAL version 3.2 and higher only. */
typedef struct source_metadata_v7 {
    size_t track_count;
    /** Array of metadata of each track connected to this source. */
    struct playback_track_metadata_v7* tracks;
} source_metadata_v7_t;

/* HAL version 3.2 and higher only. */
typedef struct sink_metadata_v7 {
    size_t track_count;
    /** Array of metadata of each track connected to this sink. */
    struct record_track_metadata_v7* tracks;
} sink_metadata_v7_t;

/**
 * audio_stream_out is the abstraction interface for the audio output hardware.
 *
 * It provides information about various properties of the audio output
 * hardware driver.
 */
struct audio_stream_out {
    /**
     * Common methods of the audio stream out.  This *must* be the first member of audio_stream_out
     * as users of this structure will cast a audio_stream to audio_stream_out pointer in contexts
     * where it's known the audio_stream references an audio_stream_out.
     */
    struct audio_stream common;

    /**
     * Return the audio hardware driver estimated latency in milliseconds.
     */
    uint32_t (*get_latency)(const struct audio_stream_out *stream);

    /**
     * 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.
     */
    int (*set_volume)(struct audio_stream_out *stream, float left, float right);

    /**
     * Write audio buffer to driver. Returns number of bytes written, or a
     * negative status_t. If at least one frame was written successfully prior to the error,
     * it is suggested that the driver return that successful (short) byte count
     * and then return an error in the subsequent call.
     *
     * If set_callback() has previously been called to enable non-blocking mode
     * the write() is not allowed to block. It must write only the number of
     * bytes that currently fit in the driver/hardware buffer and then return
     * this byte count. If this is less than the requested write size the
     * callback function must be called when more space is available in the
     * driver/hardware buffer.
     */
    ssize_t (*write)(struct audio_stream_out *stream, const void* buffer,
                     size_t bytes);

    /* return the number of audio frames written by the audio dsp to DAC since
     * the output has exited standby
     */
    int (*get_render_position)(const struct audio_stream_out *stream,
                               uint32_t *dsp_frames);

    /**
     * get the local time at which the next write to the audio driver will be presented.
     * The units are microseconds, where the epoch is decided by the local audio HAL.
     */
    int (*get_next_write_timestamp)(const struct audio_stream_out *stream,
                                    int64_t *timestamp);

    /**
     * set the callback function for notifying completion of non-blocking
     * write and drain.
     * Calling this function implies that all future write() and drain()
     * must be non-blocking and use the callback to signal completion.
     */
    int (*set_callback)(struct audio_stream_out *stream,
            stream_callback_t callback, void *cookie);

    /**
     * Notifies to the audio driver to stop playback however the queued buffers are
     * retained by the hardware. Useful for implementing pause/resume. Empty implementation
     * if not supported however should be implemented for hardware with non-trivial
     * latency. In the pause state audio hardware could still be using power. User may
     * consider calling suspend after a timeout.
     *
     * Implementation of this function is mandatory for offloaded playback.
     */
    int (*pause)(struct audio_stream_out* stream);

    /**
     * Notifies to the audio driver to resume playback following a pause.
     * Returns error if called without matching pause.
     *
     * Implementation of this function is mandatory for offloaded playback.
     */
    int (*resume)(struct audio_stream_out* stream);

    /**
     * Requests notification when data buffered by the driver/hardware has
     * been played. If set_callback() has previously been called to enable
     * non-blocking mode, the drain() must not block, instead it should return
     * quickly and completion of the drain is notified through the callback.
     * If set_callback() has not been called, the drain() must block until
     * completion.
     * If type==AUDIO_DRAIN_ALL, the drain completes when all previously written
     * data has been played.
     * If type==AUDIO_DRAIN_EARLY_NOTIFY, the drain completes shortly before all
     * data for the current track has played to allow time for the framework
     * to perform a gapless track switch.
     *
     * Drain must return immediately on stop() and flush() call
     *
     * Implementation of this function is mandatory for offloaded playback.
     */
    int (*drain)(struct audio_stream_out* stream, audio_drain_type_t type );

    /**
     * Notifies to the audio driver to flush the queued data. Stream must already
     * be paused before calling flush().
     *
     * Implementation of this function is mandatory for offloaded playback.
     */
   int (*flush)(struct audio_stream_out* stream);

    /**
     * Return a recent count of the number of audio frames presented to an external observer.
     * This excludes frames which have been written but are still in the pipeline.
     * The count is not reset to zero when output enters standby.
     * Also returns the value of CLOCK_MONOTONIC as of this presentation count.
     * The returned count is expected to be 'recent',
     * but does not need to be the most recent possible value.
     * However, the associated time should correspond to whatever count is returned.
     * Example:  assume that N+M frames have been presented, where M is a 'small' number.
     * Then it is permissible to return N instead of N+M,
     * and the timestamp should correspond to N rather than N+M.
     * The terms 'recent' and 'small' are not defined.
     * They reflect the quality of the implementation.
     *
     * 3.0 and higher only.
     */
    int (*get_presentation_position)(const struct audio_stream_out *stream,
                               uint64_t *frames, struct timespec *timestamp);

    /**
     * Called by the framework to start a stream operating in mmap mode.
     * create_mmap_buffer must be called before calling start()
     *
     * \note Function only implemented by streams operating in mmap mode.
     *
     * \param[in] stream the stream object.
     * \return 0 in case of success.
     *         -ENOSYS if called out of sequence or on non mmap stream
     */
    int (*start)(const struct audio_stream_out* stream);

    /**
     * Called by the framework to stop a stream operating in mmap mode.
     * Must be called after start()
     *
     * \note Function only implemented by streams operating in mmap mode.
     *
     * \param[in] stream the stream object.
     * \return 0 in case of success.
     *         -ENOSYS if called out of sequence or on non mmap stream
     */
    int (*stop)(const struct audio_stream_out* stream);

    /**
     * Called by the framework to retrieve information on the mmap buffer used for audio
     * samples transfer.
     *
     * \note Function only implemented by streams operating in mmap mode.
     *
     * \param[in] stream the stream object.
     * \param[in] min_size_frames minimum buffer size requested. The actual buffer
     *        size returned in struct audio_mmap_buffer_info can be larger.
     * \param[out] info address at which the mmap buffer information should be returned.
     *
     * \return 0 if the buffer was allocated.
     *         -ENODEV in case of initialization error
     *         -EINVAL if the requested buffer size is too large
     *         -ENOSYS if called out of sequence (e.g. buffer already allocated)
     */
    int (*create_mmap_buffer)(const struct audio_stream_out *stream,
                              int32_t min_size_frames,
                              struct audio_mmap_buffer_info *info);

    /**
     * Called by the framework to read current read/write position in the mmap buffer
     * with associated time stamp.
     *
     * \note Function only implemented by streams operating in mmap mode.
     *
     * \param[in] stream the stream object.
     * \param[out] position address at which the mmap read/write position should be returned.
     *
     * \return 0 if the position is successfully returned.
     *         -ENODATA if the position cannot be retrieved
     *         -ENOSYS if called before create_mmap_buffer()
     */
    int (*get_mmap_position)(const struct audio_stream_out *stream,
                             struct audio_mmap_position *position);

    /**
     * Called when the metadata of the stream's source has been changed.
     * @param source_metadata Description of the audio that is played by the clients.
     */
    void (*update_source_metadata)(struct audio_stream_out *stream,
                                   const struct source_metadata* source_metadata);

    /**
     * Set the callback function for notifying events for an output stream.
     */
    int (*set_event_callback)(struct audio_stream_out *stream,
                              stream_event_callback_t callback,
                              void *cookie);

    /**
     * Called when the metadata of the stream's source has been changed.
     * HAL version 3.2 and higher only.
     * @param source_metadata Description of the audio that is played by the clients.
     */
    void (*update_source_metadata_v7)(struct audio_stream_out *stream,
                                      const struct source_metadata_v7* source_metadata);

    /**
     * Returns the Dual Mono mode presentation setting.
     *
     * \param[in] stream the stream object.
     * \param[out] mode current setting of Dual Mono mode.
     *
     * \return 0 if the position is successfully returned.
     *         -EINVAL if the arguments are invalid
     *         -ENOSYS if the function is not available
     */
    int (*get_dual_mono_mode)(struct audio_stream_out *stream, audio_dual_mono_mode_t *mode);

    /**
     * Sets the Dual Mono mode presentation on the output device.
     *
     * \param[in] stream the stream object.
     * \param[in] mode selected Dual Mono mode.
     *
     * \return 0 in case of success.
     *         -EINVAL if the arguments are invalid
     *         -ENOSYS if the function is not available
     */
    int (*set_dual_mono_mode)(struct audio_stream_out *stream, const audio_dual_mono_mode_t mode);

    /**
     * Returns the Audio Description Mix level in dB.
     *
     * \param[in] stream the stream object.
     * \param[out] leveldB the current Audio Description Mix Level in dB.
     *
     * \return 0 in case of success.
     *         -EINVAL if the arguments are invalid
     *         -ENOSYS if the function is not available
     */
    int (*get_audio_description_mix_level)(struct audio_stream_out *stream, float *leveldB);

    /**
     * Sets the Audio Description Mix level in dB.
     *
     * \param[in] stream the stream object.
     * \param[in] leveldB Audio Description Mix Level in dB.
     *
     * \return 0 in case of success.
     *         -EINVAL if the arguments are invalid
     *         -ENOSYS if the function is not available
     */
    int (*set_audio_description_mix_level)(struct audio_stream_out *stream, const float leveldB);

    /**
     * Retrieves current playback rate parameters.
     *
     * \param[in] stream the stream object.
     * \param[out] playbackRate current playback parameters.
     *
     * \return 0 in case of success.
     *         -EINVAL if the arguments are invalid
     *         -ENOSYS if the function is not available
     */
    int (*get_playback_rate_parameters)(struct audio_stream_out *stream,
                                        audio_playback_rate_t *playbackRate);

    /**
     * Sets the playback rate parameters that control playback behavior.
     *
     * \param[in] stream the stream object.
     * \param[in] playbackRate playback parameters.
     *
     * \return 0 in case of success.
     *         -EINVAL if the arguments are invalid
     *         -ENOSYS if the function is not available
     */
    int (*set_playback_rate_parameters)(struct audio_stream_out *stream,
                                        const audio_playback_rate_t *playbackRate);
};
typedef struct audio_stream_out audio_stream_out_t;

struct audio_stream_in {
    /**
     * Common methods of the audio stream in.  This *must* be the first member of audio_stream_in
     * as users of this structure will cast a audio_stream to audio_stream_in pointer in contexts
     * where it's known the audio_stream references an audio_stream_in.
     */
    struct audio_stream common;

    /** set the input gain for the audio driver. This method is for
     *  for future use */
    int (*set_gain)(struct audio_stream_in *stream, float gain);

    /** Read audio buffer in from audio driver. Returns number of bytes read, or a
     *  negative status_t. If at least one frame was read prior to the error,
     *  read should return that byte count and then return an error in the subsequent call.
     */
    ssize_t (*read)(struct audio_stream_in *stream, void* buffer,
                    size_t bytes);

    /**
     * Return the amount 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
     */
    uint32_t (*get_input_frames_lost)(struct audio_stream_in *stream);

    /**
     * Return a recent count of the number of audio frames received and
     * the clock time associated with that frame count.
     *
     * frames is the total frame count received. This should be as early in
     *     the capture pipeline as possible. In general,
     *     frames should be non-negative and should not go "backwards".
     *
     * time is the clock MONOTONIC time when frames was measured. In general,
     *     time should be a positive quantity and should not go "backwards".
     *
     * The status returned is 0 on success, -ENOSYS if the device is not
     * ready/available, or -EINVAL if the arguments are null or otherwise invalid.
     */
    int (*get_capture_position)(const struct audio_stream_in *stream,
                                int64_t *frames, int64_t *time);

    /**
     * Called by the framework to start a stream operating in mmap mode.
     * create_mmap_buffer must be called before calling start()
     *
     * \note Function only implemented by streams operating in mmap mode.
     *
     * \param[in] stream the stream object.
     * \return 0 in case off success.
     *         -ENOSYS if called out of sequence or on non mmap stream
     */
    int (*start)(const struct audio_stream_in* stream);

    /**
     * Called by the framework to stop a stream operating in mmap mode.
     *
     * \note Function only implemented by streams operating in mmap mode.
     *
     * \param[in] stream the stream object.
     * \return 0 in case of success.
     *         -ENOSYS if called out of sequence or on non mmap stream
     */
    int (*stop)(const struct audio_stream_in* stream);

    /**
     * Called by the framework to retrieve information on the mmap buffer used for audio
     * samples transfer.
     *
     * \note Function only implemented by streams operating in mmap mode.
     *
     * \param[in] stream the stream object.
     * \param[in] min_size_frames minimum buffer size requested. The actual buffer
     *        size returned in struct audio_mmap_buffer_info can be larger.
     * \param[out] info address at which the mmap buffer information should be returned.
     *
     * \return 0 if the buffer was allocated.
     *         -ENODEV in case of initialization error
     *         -EINVAL if the requested buffer size is too large
     *         -ENOSYS if called out of sequence (e.g. buffer already allocated)
     */
    int (*create_mmap_buffer)(const struct audio_stream_in *stream,
                              int32_t min_size_frames,
                              struct audio_mmap_buffer_info *info);

    /**
     * Called by the framework to read current read/write position in the mmap buffer
     * with associated time stamp.
     *
     * \note Function only implemented by streams operating in mmap mode.
     *
     * \param[in] stream the stream object.
     * \param[out] position address at which the mmap read/write position should be returned.
     *
     * \return 0 if the position is successfully returned.
     *         -ENODATA if the position cannot be retreived
     *         -ENOSYS if called before mmap_read_position()
     */
    int (*get_mmap_position)(const struct audio_stream_in *stream,
                             struct audio_mmap_position *position);

    /**
     * Called by the framework to read active microphones
     *
     * \param[in] stream the stream object.
     * \param[out] mic_array Pointer to first element on array with microphone info
     * \param[out] mic_count When called, this holds the value of the max number of elements
     *                       allowed in the mic_array. The actual number of elements written
     *                       is returned here.
     *                       if mic_count is passed as zero, mic_array will not be populated,
     *                       and mic_count will return the actual number of active microphones.
     *
     * \return 0 if the microphone array is successfully filled.
     *         -ENOSYS if there is an error filling the data
     */
    int (*get_active_microphones)(const struct audio_stream_in *stream,
                                  struct audio_microphone_characteristic_t *mic_array,
                                  size_t *mic_count);

    /**
     * Called by the framework to instruct the HAL to optimize the capture stream in the
     * specified direction.
     *
     * \param[in] stream    the stream object.
     * \param[in] direction The direction constant (from audio-base.h)
     *   MIC_DIRECTION_UNSPECIFIED  Don't do any directionality processing of the
     *      activated microphone(s).
     *   MIC_DIRECTION_FRONT        Optimize capture for audio coming from the screen-side
     *      of the device.
     *   MIC_DIRECTION_BACK         Optimize capture for audio coming from the side of the
     *      device opposite the screen.
     *   MIC_DIRECTION_EXTERNAL     Optimize capture for audio coming from an off-device
     *      microphone.
     * \return OK if the call is successful, an error code otherwise.
     */
    int (*set_microphone_direction)(const struct audio_stream_in *stream,
                                    audio_microphone_direction_t direction);

    /**
     * Called by the framework to specify to the HAL the desired zoom factor for the selected
     * microphone(s).
     *
     * \param[in] stream    the stream object.
     * \param[in] zoom      the zoom factor.
     * \return OK if the call is successful, an error code otherwise.
     */
    int (*set_microphone_field_dimension)(const struct audio_stream_in *stream,
                                          float zoom);

    /**
     * Called when the metadata of the stream's sink has been changed.
     * @param sink_metadata Description of the audio that is recorded by the clients.
     */
    void (*update_sink_metadata)(struct audio_stream_in *stream,
                                 const struct sink_metadata* sink_metadata);

    /**
     * Called when the metadata of the stream's sink has been changed.
     * HAL version 3.2 and higher only.
     * @param sink_metadata Description of the audio that is recorded by the clients.
     */
    void (*update_sink_metadata_v7)(struct audio_stream_in *stream,
                                    const struct sink_metadata_v7* sink_metadata);
};
typedef struct audio_stream_in audio_stream_in_t;

/**
 * return the frame size (number of bytes per sample).
 *
 * Deprecated: use audio_stream_out_frame_size() or audio_stream_in_frame_size() instead.
 */
__attribute__((__deprecated__))
static inline size_t audio_stream_frame_size(const struct audio_stream *s)
{
    size_t chan_samp_sz;
    audio_format_t format = s->get_format(s);

    if (audio_has_proportional_frames(format)) {
        chan_samp_sz = audio_bytes_per_sample(format);
        return popcount(s->get_channels(s)) * chan_samp_sz;
    }

    return sizeof(int8_t);
}

/**
 * return the frame size (number of bytes per sample) of an output stream.
 */
static inline size_t audio_stream_out_frame_size(const struct audio_stream_out *s)
{
    size_t chan_samp_sz;
    audio_format_t format = s->common.get_format(&s->common);

    if (audio_has_proportional_frames(format)) {
        chan_samp_sz = audio_bytes_per_sample(format);
        return audio_channel_count_from_out_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
    }

    return sizeof(int8_t);
}

/**
 * return the frame size (number of bytes per sample) of an input stream.
 */
static inline size_t audio_stream_in_frame_size(const struct audio_stream_in *s)
{
    size_t chan_samp_sz;
    audio_format_t format = s->common.get_format(&s->common);

    if (audio_has_proportional_frames(format)) {
        chan_samp_sz = audio_bytes_per_sample(format);
        return audio_channel_count_from_in_mask(s->common.get_channels(&s->common)) * chan_samp_sz;
    }

    return sizeof(int8_t);
}

/**********************************************************************/

/**
 * Every hardware module must have a data structure named HAL_MODULE_INFO_SYM
 * and the fields of this data structure must begin with hw_module_t
 * followed by module specific information.
 */
struct audio_module {
    struct hw_module_t common;
};

struct audio_hw_device {
    /**
     * Common methods of the audio device.  This *must* be the first member of audio_hw_device
     * as users of this structure will cast a hw_device_t to audio_hw_device pointer in contexts
     * where it's known the hw_device_t references an audio_hw_device.
     */
    struct hw_device_t common;

    /**
     * used by audio flinger to enumerate what devices are supported by
     * each audio_hw_device implementation.
     *
     * Return value is a bitmask of 1 or more values of audio_devices_t
     *
     * NOTE: audio HAL implementations starting with
     * AUDIO_DEVICE_API_VERSION_2_0 do not implement this function.
     * All supported devices should be listed in audio_policy.conf
     * file and the audio policy manager must choose the appropriate
     * audio module based on information in this file.
     */
    uint32_t (*get_supported_devices)(const struct audio_hw_device *dev);

    /**
     * check to see if the audio hardware interface has been initialized.
     * returns 0 on success, -ENODEV on failure.
     */
    int (*init_check)(const struct audio_hw_device *dev);

    /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
    int (*set_voice_volume)(struct audio_hw_device *dev, float volume);

    /**
     * set the audio volume for all audio activities other than voice call.
     * Range between 0.0 and 1.0. If any value other than 0 is returned,
     * the software mixer will emulate this capability.
     */
    int (*set_master_volume)(struct audio_hw_device *dev, float volume);

    /**
     * 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.  HALs which do not support
     * this method may leave it set to NULL.
     */
    int (*get_master_volume)(struct audio_hw_device *dev, float *volume);

    /**
     * set_mode is called when the audio mode changes. AUDIO_MODE_NORMAL mode
     * is for standard audio playback, AUDIO_MODE_RINGTONE when a ringtone is
     * playing, and AUDIO_MODE_IN_CALL when a call is in progress.
     */
    int (*set_mode)(struct audio_hw_device *dev, audio_mode_t mode);

    /* mic mute */
    int (*set_mic_mute)(struct audio_hw_device *dev, bool state);
    int (*get_mic_mute)(const struct audio_hw_device *dev, bool *state);

    /* set/get global audio parameters */
    int (*set_parameters)(struct audio_hw_device *dev, const char *kv_pairs);

    /*
     * Returns a pointer to a heap allocated string. The caller is responsible
     * for freeing the memory for it using free().
     */
    char * (*get_parameters)(const struct audio_hw_device *dev,
                             const char *keys);

    /* Returns audio input buffer size according to parameters passed or
     * 0 if one of the parameters is not supported.
     * See also get_buffer_size which is for a particular stream.
     */
    size_t (*get_input_buffer_size)(const struct audio_hw_device *dev,
                                    const struct audio_config *config);

    /** This method creates and opens the audio hardware output stream.
     * The "address" parameter qualifies the "devices" audio device type if needed.
     * The format format depends on the device type:
     * - Bluetooth devices use the MAC address of the device in the form "00:11:22:AA:BB:CC"
     * - USB devices use the ALSA card and device numbers in the form  "card=X;device=Y"
     * - Other devices may use a number or any other string.
     */

    int (*open_output_stream)(struct audio_hw_device *dev,
                              audio_io_handle_t handle,
                              audio_devices_t devices,
                              audio_output_flags_t flags,
                              struct audio_config *config,
                              struct audio_stream_out **stream_out,
                              const char *address);

    void (*close_output_stream)(struct audio_hw_device *dev,
                                struct audio_stream_out* stream_out);

    /** This method creates and opens the audio hardware input stream */
    int (*open_input_stream)(struct audio_hw_device *dev,
                             audio_io_handle_t handle,
                             audio_devices_t devices,
                             struct audio_config *config,
                             struct audio_stream_in **stream_in,
                             audio_input_flags_t flags,
                             const char *address,
                             audio_source_t source);

    void (*close_input_stream)(struct audio_hw_device *dev,
                               struct audio_stream_in *stream_in);

    /**
     * Called by the framework to read available microphones characteristics.
     *
     * \param[in] dev the hw_device object.
     * \param[out] mic_array Pointer to first element on array with microphone info
     * \param[out] mic_count When called, this holds the value of the max number of elements
     *                       allowed in the mic_array. The actual number of elements written
     *                       is returned here.
     *                       if mic_count is passed as zero, mic_array will not be populated,
     *                       and mic_count will return the actual number of microphones in the
     *                       system.
     *
     * \return 0 if the microphone array is successfully filled.
     *         -ENOSYS if there is an error filling the data
     */
    int (*get_microphones)(const struct audio_hw_device *dev,
                           struct audio_microphone_characteristic_t *mic_array,
                           size_t *mic_count);

    /** This method dumps the state of the audio hardware */
    int (*dump)(const struct audio_hw_device *dev, int fd);

    /**
     * set the audio mute status for all audio activities.  If any value other
     * than 0 is returned, the software mixer will emulate this capability.
     */
    int (*set_master_mute)(struct audio_hw_device *dev, bool mute);

    /**
     * Get the current master mute status for the HAL, if the HAL supports
     * master mute control.  AudioFlinger will query this value from the primary
     * audio HAL when the service starts and use the value for setting the
     * initial master mute across all HALs.  HALs which do not support this
     * method may leave it set to NULL.
     */
    int (*get_master_mute)(struct audio_hw_device *dev, bool *mute);

    /**
     * Routing control
     */

    /* Creates an audio patch between several source and sink ports.
     * The handle is allocated by the HAL and should be unique for this
     * audio HAL module. */
    int (*create_audio_patch)(struct audio_hw_device *dev,
                               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);

    /* Release an audio patch */
    int (*release_audio_patch)(struct audio_hw_device *dev,
                               audio_patch_handle_t handle);

    /* Fills the list of supported attributes for a given audio port.
     * As input, "port" contains the information (type, role, address etc...)
     * needed by the HAL to identify the port.
     * As output, "port" contains possible attributes (sampling rates, formats,
     * channel masks, gain controllers...) for this port.
     */
    int (*get_audio_port)(struct audio_hw_device *dev,
                          struct audio_port *port);

    /* Set audio port configuration */
    int (*set_audio_port_config)(struct audio_hw_device *dev,
                         const struct audio_port_config *config);

    /**
     * Applies an audio effect to an audio device.
     *
     * @param dev the audio HAL device context.
     * @param device identifies the sink or source device the effect must be applied to.
     *               "device" is the audio_port_handle_t indicated for the device when
     *               the audio patch connecting that device was created.
     * @param effect effect interface handle corresponding to the effect being added.
     * @return retval operation completion status.
     */
    int (*add_device_effect)(struct audio_hw_device *dev,
                        audio_port_handle_t device, effect_handle_t effect);

    /**
     * Stops applying an audio effect to an audio device.
     *
     * @param dev the audio HAL device context.
     * @param device identifies the sink or source device this effect was applied to.
     *               "device" is the audio_port_handle_t indicated for the device when
     *               the audio patch is created.
     * @param effect effect interface handle corresponding to the effect being removed.
     * @return retval operation completion status.
     */
    int (*remove_device_effect)(struct audio_hw_device *dev,
                        audio_port_handle_t device, effect_handle_t effect);

    /**
     * Fills the list of supported attributes for a given audio port.
     * As input, "port" contains the information (type, role, address etc...)
     * needed by the HAL to identify the port.
     * As output, "port" contains possible attributes (sampling rates, formats,
     * channel masks, gain controllers...) for this port. The possible attributes
     * are saved as audio profiles, which contains audio format and the supported
     * sampling rates and channel masks.
     */
    int (*get_audio_port_v7)(struct audio_hw_device *dev,
                             struct audio_port_v7 *port);
};
typedef struct audio_hw_device audio_hw_device_t;

/** convenience API for opening and closing a supported device */

static inline int audio_hw_device_open(const struct hw_module_t* module,
                                       struct audio_hw_device** device)
{
    return module->methods->open(module, AUDIO_HARDWARE_INTERFACE,
                                 TO_HW_DEVICE_T_OPEN(device));
}

static inline int audio_hw_device_close(struct audio_hw_device* device)
{
    return device->common.close(&device->common);
}


__END_DECLS

#endif  // ANDROID_AUDIO_INTERFACE_H