diff --git a/include/hardware/audio.h b/include/hardware/audio.h
new file mode 100644
index 0000000..2d6eb30
--- /dev/null
+++ b/include/hardware/audio.h
@@ -0,0 +1,745 @@
+/*
+ * 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_CURRENT AUDIO_DEVICE_API_VERSION_3_0
+/* 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"
+
+/**
+ *  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 int (*stream_callback_t)(stream_callback_event_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;
+
+/**
+ * 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);
+};
+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);
+};
+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);
+
+    /** 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);
+
+};
+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