diff --git a/audio/common/2.0/default/Android.bp b/audio/common/2.0/default/Android.bp
index ac66479..123f8b3 100644
--- a/audio/common/2.0/default/Android.bp
+++ b/audio/common/2.0/default/Android.bp
@@ -16,10 +16,7 @@
 cc_library_shared {
     name: "android.hardware.audio.common@2.0-util",
     defaults: ["hidl_defaults"],
-    vendor_available: true,
-    vndk: {
-        enabled: true,
-    },
+    vendor: true,
     srcs: [
         "HidlUtils.cpp",
     ],
@@ -41,7 +38,7 @@
     ],
 
     header_libs: [
-        "libaudio_system_headers",
+        "android.hardware.audio.common.legacy@2.0",
         "libhardware_headers",
     ],
 }
diff --git a/audio/common/2.0/legacy/Android.bp b/audio/common/2.0/legacy/Android.bp
new file mode 100644
index 0000000..2888c96
--- /dev/null
+++ b/audio/common/2.0/legacy/Android.bp
@@ -0,0 +1,15 @@
+cc_library_headers {
+    name: "android.hardware.audio.common.legacy@2.0",
+    vendor: true,
+    header_libs: [
+        "libhardware_headers",
+        "android.hardware.audio.common.legacy@all-versions",
+    ],
+    export_header_lib_headers: [
+        "libhardware_headers",
+        "android.hardware.audio.common.legacy@all-versions",
+    ],
+
+    export_include_dirs: ["include"],
+}
+
diff --git a/audio/common/2.0/legacy/OWNERS b/audio/common/2.0/legacy/OWNERS
new file mode 100644
index 0000000..6fdc97c
--- /dev/null
+++ b/audio/common/2.0/legacy/OWNERS
@@ -0,0 +1,3 @@
+elaurent@google.com
+krocard@google.com
+mnaganov@google.com
diff --git a/audio/common/2.0/legacy/include/hardware/audio.h b/audio/common/2.0/legacy/include/hardware/audio.h
new file mode 100644
index 0000000..1ad3e0e
--- /dev/null
+++ b/audio/common/2.0/legacy/include/hardware/audio.h
@@ -0,0 +1,709 @@
+/*
+ * 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/audio_effect.h>
+#include <hardware/hardware.h>
+#include <system/audio.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
diff --git a/audio/common/2.0/legacy/include/system/audio-base.h b/audio/common/2.0/legacy/include/system/audio-base.h
new file mode 100644
index 0000000..53e524b
--- /dev/null
+++ b/audio/common/2.0/legacy/include/system/audio-base.h
@@ -0,0 +1,434 @@
+// This file is autogenerated by hidl-gen. Do not edit manually.
+// Source: android.hardware.audio.common@2.0
+// Root: android.hardware:hardware/interfaces
+
+#ifndef HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_COMMON_V2_0_EXPORTED_CONSTANTS_H_
+#define HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_COMMON_V2_0_EXPORTED_CONSTANTS_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+enum {
+    AUDIO_IO_HANDLE_NONE = 0,
+    AUDIO_MODULE_HANDLE_NONE = 0,
+    AUDIO_PORT_HANDLE_NONE = 0,
+    AUDIO_PATCH_HANDLE_NONE = 0,
+};
+
+typedef enum {
+    AUDIO_STREAM_DEFAULT = -1,  // (-1)
+    AUDIO_STREAM_MIN = 0,
+    AUDIO_STREAM_VOICE_CALL = 0,
+    AUDIO_STREAM_SYSTEM = 1,
+    AUDIO_STREAM_RING = 2,
+    AUDIO_STREAM_MUSIC = 3,
+    AUDIO_STREAM_ALARM = 4,
+    AUDIO_STREAM_NOTIFICATION = 5,
+    AUDIO_STREAM_BLUETOOTH_SCO = 6,
+    AUDIO_STREAM_ENFORCED_AUDIBLE = 7,
+    AUDIO_STREAM_DTMF = 8,
+    AUDIO_STREAM_TTS = 9,
+    AUDIO_STREAM_ACCESSIBILITY = 10,
+    AUDIO_STREAM_REROUTING = 11,
+    AUDIO_STREAM_PATCH = 12,
+    AUDIO_STREAM_PUBLIC_CNT = 11,      // (ACCESSIBILITY + 1)
+    AUDIO_STREAM_FOR_POLICY_CNT = 12,  // PATCH
+    AUDIO_STREAM_CNT = 13,             // (PATCH + 1)
+} audio_stream_type_t;
+
+typedef enum {
+    AUDIO_SOURCE_DEFAULT = 0,
+    AUDIO_SOURCE_MIC = 1,
+    AUDIO_SOURCE_VOICE_UPLINK = 2,
+    AUDIO_SOURCE_VOICE_DOWNLINK = 3,
+    AUDIO_SOURCE_VOICE_CALL = 4,
+    AUDIO_SOURCE_CAMCORDER = 5,
+    AUDIO_SOURCE_VOICE_RECOGNITION = 6,
+    AUDIO_SOURCE_VOICE_COMMUNICATION = 7,
+    AUDIO_SOURCE_REMOTE_SUBMIX = 8,
+    AUDIO_SOURCE_UNPROCESSED = 9,
+    AUDIO_SOURCE_CNT = 10,
+    AUDIO_SOURCE_MAX = 9,  // (CNT - 1)
+    AUDIO_SOURCE_FM_TUNER = 1998,
+    AUDIO_SOURCE_HOTWORD = 1999,
+} audio_source_t;
+
+typedef enum {
+    AUDIO_SESSION_OUTPUT_STAGE = -1,  // (-1)
+    AUDIO_SESSION_OUTPUT_MIX = 0,
+    AUDIO_SESSION_ALLOCATE = 0,
+    AUDIO_SESSION_NONE = 0,
+} audio_session_t;
+
+typedef enum {
+    AUDIO_FORMAT_INVALID = 4294967295u,           // 0xFFFFFFFFUL
+    AUDIO_FORMAT_DEFAULT = 0u,                    // 0
+    AUDIO_FORMAT_PCM = 0u,                        // 0x00000000UL
+    AUDIO_FORMAT_MP3 = 16777216u,                 // 0x01000000UL
+    AUDIO_FORMAT_AMR_NB = 33554432u,              // 0x02000000UL
+    AUDIO_FORMAT_AMR_WB = 50331648u,              // 0x03000000UL
+    AUDIO_FORMAT_AAC = 67108864u,                 // 0x04000000UL
+    AUDIO_FORMAT_HE_AAC_V1 = 83886080u,           // 0x05000000UL
+    AUDIO_FORMAT_HE_AAC_V2 = 100663296u,          // 0x06000000UL
+    AUDIO_FORMAT_VORBIS = 117440512u,             // 0x07000000UL
+    AUDIO_FORMAT_OPUS = 134217728u,               // 0x08000000UL
+    AUDIO_FORMAT_AC3 = 150994944u,                // 0x09000000UL
+    AUDIO_FORMAT_E_AC3 = 167772160u,              // 0x0A000000UL
+    AUDIO_FORMAT_DTS = 184549376u,                // 0x0B000000UL
+    AUDIO_FORMAT_DTS_HD = 201326592u,             // 0x0C000000UL
+    AUDIO_FORMAT_IEC61937 = 218103808u,           // 0x0D000000UL
+    AUDIO_FORMAT_DOLBY_TRUEHD = 234881024u,       // 0x0E000000UL
+    AUDIO_FORMAT_EVRC = 268435456u,               // 0x10000000UL
+    AUDIO_FORMAT_EVRCB = 285212672u,              // 0x11000000UL
+    AUDIO_FORMAT_EVRCWB = 301989888u,             // 0x12000000UL
+    AUDIO_FORMAT_EVRCNW = 318767104u,             // 0x13000000UL
+    AUDIO_FORMAT_AAC_ADIF = 335544320u,           // 0x14000000UL
+    AUDIO_FORMAT_WMA = 352321536u,                // 0x15000000UL
+    AUDIO_FORMAT_WMA_PRO = 369098752u,            // 0x16000000UL
+    AUDIO_FORMAT_AMR_WB_PLUS = 385875968u,        // 0x17000000UL
+    AUDIO_FORMAT_MP2 = 402653184u,                // 0x18000000UL
+    AUDIO_FORMAT_QCELP = 419430400u,              // 0x19000000UL
+    AUDIO_FORMAT_DSD = 436207616u,                // 0x1A000000UL
+    AUDIO_FORMAT_FLAC = 452984832u,               // 0x1B000000UL
+    AUDIO_FORMAT_ALAC = 469762048u,               // 0x1C000000UL
+    AUDIO_FORMAT_APE = 486539264u,                // 0x1D000000UL
+    AUDIO_FORMAT_AAC_ADTS = 503316480u,           // 0x1E000000UL
+    AUDIO_FORMAT_SBC = 520093696u,                // 0x1F000000UL
+    AUDIO_FORMAT_APTX = 536870912u,               // 0x20000000UL
+    AUDIO_FORMAT_APTX_HD = 553648128u,            // 0x21000000UL
+    AUDIO_FORMAT_AC4 = 570425344u,                // 0x22000000UL
+    AUDIO_FORMAT_LDAC = 587202560u,               // 0x23000000UL
+    AUDIO_FORMAT_MAIN_MASK = 4278190080u,         // 0xFF000000UL
+    AUDIO_FORMAT_SUB_MASK = 16777215u,            // 0x00FFFFFFUL
+    AUDIO_FORMAT_PCM_SUB_16_BIT = 1u,             // 0x1
+    AUDIO_FORMAT_PCM_SUB_8_BIT = 2u,              // 0x2
+    AUDIO_FORMAT_PCM_SUB_32_BIT = 3u,             // 0x3
+    AUDIO_FORMAT_PCM_SUB_8_24_BIT = 4u,           // 0x4
+    AUDIO_FORMAT_PCM_SUB_FLOAT = 5u,              // 0x5
+    AUDIO_FORMAT_PCM_SUB_24_BIT_PACKED = 6u,      // 0x6
+    AUDIO_FORMAT_MP3_SUB_NONE = 0u,               // 0x0
+    AUDIO_FORMAT_AMR_SUB_NONE = 0u,               // 0x0
+    AUDIO_FORMAT_AAC_SUB_MAIN = 1u,               // 0x1
+    AUDIO_FORMAT_AAC_SUB_LC = 2u,                 // 0x2
+    AUDIO_FORMAT_AAC_SUB_SSR = 4u,                // 0x4
+    AUDIO_FORMAT_AAC_SUB_LTP = 8u,                // 0x8
+    AUDIO_FORMAT_AAC_SUB_HE_V1 = 16u,             // 0x10
+    AUDIO_FORMAT_AAC_SUB_SCALABLE = 32u,          // 0x20
+    AUDIO_FORMAT_AAC_SUB_ERLC = 64u,              // 0x40
+    AUDIO_FORMAT_AAC_SUB_LD = 128u,               // 0x80
+    AUDIO_FORMAT_AAC_SUB_HE_V2 = 256u,            // 0x100
+    AUDIO_FORMAT_AAC_SUB_ELD = 512u,              // 0x200
+    AUDIO_FORMAT_VORBIS_SUB_NONE = 0u,            // 0x0
+    AUDIO_FORMAT_PCM_16_BIT = 1u,                 // (PCM | PCM_SUB_16_BIT)
+    AUDIO_FORMAT_PCM_8_BIT = 2u,                  // (PCM | PCM_SUB_8_BIT)
+    AUDIO_FORMAT_PCM_32_BIT = 3u,                 // (PCM | PCM_SUB_32_BIT)
+    AUDIO_FORMAT_PCM_8_24_BIT = 4u,               // (PCM | PCM_SUB_8_24_BIT)
+    AUDIO_FORMAT_PCM_FLOAT = 5u,                  // (PCM | PCM_SUB_FLOAT)
+    AUDIO_FORMAT_PCM_24_BIT_PACKED = 6u,          // (PCM | PCM_SUB_24_BIT_PACKED)
+    AUDIO_FORMAT_AAC_MAIN = 67108865u,            // (AAC | AAC_SUB_MAIN)
+    AUDIO_FORMAT_AAC_LC = 67108866u,              // (AAC | AAC_SUB_LC)
+    AUDIO_FORMAT_AAC_SSR = 67108868u,             // (AAC | AAC_SUB_SSR)
+    AUDIO_FORMAT_AAC_LTP = 67108872u,             // (AAC | AAC_SUB_LTP)
+    AUDIO_FORMAT_AAC_HE_V1 = 67108880u,           // (AAC | AAC_SUB_HE_V1)
+    AUDIO_FORMAT_AAC_SCALABLE = 67108896u,        // (AAC | AAC_SUB_SCALABLE)
+    AUDIO_FORMAT_AAC_ERLC = 67108928u,            // (AAC | AAC_SUB_ERLC)
+    AUDIO_FORMAT_AAC_LD = 67108992u,              // (AAC | AAC_SUB_LD)
+    AUDIO_FORMAT_AAC_HE_V2 = 67109120u,           // (AAC | AAC_SUB_HE_V2)
+    AUDIO_FORMAT_AAC_ELD = 67109376u,             // (AAC | AAC_SUB_ELD)
+    AUDIO_FORMAT_AAC_ADTS_MAIN = 503316481u,      // (AAC_ADTS | AAC_SUB_MAIN)
+    AUDIO_FORMAT_AAC_ADTS_LC = 503316482u,        // (AAC_ADTS | AAC_SUB_LC)
+    AUDIO_FORMAT_AAC_ADTS_SSR = 503316484u,       // (AAC_ADTS | AAC_SUB_SSR)
+    AUDIO_FORMAT_AAC_ADTS_LTP = 503316488u,       // (AAC_ADTS | AAC_SUB_LTP)
+    AUDIO_FORMAT_AAC_ADTS_HE_V1 = 503316496u,     // (AAC_ADTS | AAC_SUB_HE_V1)
+    AUDIO_FORMAT_AAC_ADTS_SCALABLE = 503316512u,  // (AAC_ADTS | AAC_SUB_SCALABLE)
+    AUDIO_FORMAT_AAC_ADTS_ERLC = 503316544u,      // (AAC_ADTS | AAC_SUB_ERLC)
+    AUDIO_FORMAT_AAC_ADTS_LD = 503316608u,        // (AAC_ADTS | AAC_SUB_LD)
+    AUDIO_FORMAT_AAC_ADTS_HE_V2 = 503316736u,     // (AAC_ADTS | AAC_SUB_HE_V2)
+    AUDIO_FORMAT_AAC_ADTS_ELD = 503316992u,       // (AAC_ADTS | AAC_SUB_ELD)
+} audio_format_t;
+
+enum {
+    FCC_2 = 2,
+    FCC_8 = 8,
+};
+
+enum {
+    AUDIO_CHANNEL_REPRESENTATION_POSITION = 0u,      // 0
+    AUDIO_CHANNEL_REPRESENTATION_INDEX = 2u,         // 2
+    AUDIO_CHANNEL_NONE = 0u,                         // 0x0
+    AUDIO_CHANNEL_INVALID = 3221225472u,             // 0xC0000000
+    AUDIO_CHANNEL_OUT_FRONT_LEFT = 1u,               // 0x1
+    AUDIO_CHANNEL_OUT_FRONT_RIGHT = 2u,              // 0x2
+    AUDIO_CHANNEL_OUT_FRONT_CENTER = 4u,             // 0x4
+    AUDIO_CHANNEL_OUT_LOW_FREQUENCY = 8u,            // 0x8
+    AUDIO_CHANNEL_OUT_BACK_LEFT = 16u,               // 0x10
+    AUDIO_CHANNEL_OUT_BACK_RIGHT = 32u,              // 0x20
+    AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER = 64u,    // 0x40
+    AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 128u,  // 0x80
+    AUDIO_CHANNEL_OUT_BACK_CENTER = 256u,            // 0x100
+    AUDIO_CHANNEL_OUT_SIDE_LEFT = 512u,              // 0x200
+    AUDIO_CHANNEL_OUT_SIDE_RIGHT = 1024u,            // 0x400
+    AUDIO_CHANNEL_OUT_TOP_CENTER = 2048u,            // 0x800
+    AUDIO_CHANNEL_OUT_TOP_FRONT_LEFT = 4096u,        // 0x1000
+    AUDIO_CHANNEL_OUT_TOP_FRONT_CENTER = 8192u,      // 0x2000
+    AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT = 16384u,      // 0x4000
+    AUDIO_CHANNEL_OUT_TOP_BACK_LEFT = 32768u,        // 0x8000
+    AUDIO_CHANNEL_OUT_TOP_BACK_CENTER = 65536u,      // 0x10000
+    AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT = 131072u,      // 0x20000
+    AUDIO_CHANNEL_OUT_MONO = 1u,                     // OUT_FRONT_LEFT
+    AUDIO_CHANNEL_OUT_STEREO = 3u,                   // (OUT_FRONT_LEFT | OUT_FRONT_RIGHT)
+    AUDIO_CHANNEL_OUT_2POINT1 = 11u,  // ((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_LOW_FREQUENCY)
+    AUDIO_CHANNEL_OUT_QUAD =
+        51u,  // (((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_BACK_LEFT) | OUT_BACK_RIGHT)
+    AUDIO_CHANNEL_OUT_QUAD_BACK = 51u,  // OUT_QUAD
+    AUDIO_CHANNEL_OUT_QUAD_SIDE =
+        1539u,  // (((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_SIDE_LEFT) | OUT_SIDE_RIGHT)
+    AUDIO_CHANNEL_OUT_SURROUND =
+        263u,  // (((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_FRONT_CENTER) | OUT_BACK_CENTER)
+    AUDIO_CHANNEL_OUT_PENTA = 55u,    // (OUT_QUAD | OUT_FRONT_CENTER)
+    AUDIO_CHANNEL_OUT_5POINT1 = 63u,  // (((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_FRONT_CENTER)
+                                      // | OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) | OUT_BACK_RIGHT)
+    AUDIO_CHANNEL_OUT_5POINT1_BACK = 63u,    // OUT_5POINT1
+    AUDIO_CHANNEL_OUT_5POINT1_SIDE = 1551u,  // (((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) |
+                                             // OUT_FRONT_CENTER) | OUT_LOW_FREQUENCY) |
+                                             // OUT_SIDE_LEFT) | OUT_SIDE_RIGHT)
+    AUDIO_CHANNEL_OUT_6POINT1 = 319u,        // ((((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) |
+                                       // OUT_FRONT_CENTER) | OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) |
+                                       // OUT_BACK_RIGHT) | OUT_BACK_CENTER)
+    AUDIO_CHANNEL_OUT_7POINT1 = 1599u,  // (((((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) |
+                                        // OUT_FRONT_CENTER) | OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) |
+                                        // OUT_BACK_RIGHT) | OUT_SIDE_LEFT) | OUT_SIDE_RIGHT)
+    AUDIO_CHANNEL_OUT_ALL =
+        262143u,  // (((((((((((((((((OUT_FRONT_LEFT | OUT_FRONT_RIGHT) | OUT_FRONT_CENTER) |
+                  // OUT_LOW_FREQUENCY) | OUT_BACK_LEFT) | OUT_BACK_RIGHT) |
+                  // OUT_FRONT_LEFT_OF_CENTER) | OUT_FRONT_RIGHT_OF_CENTER) | OUT_BACK_CENTER) |
+                  // OUT_SIDE_LEFT) | OUT_SIDE_RIGHT) | OUT_TOP_CENTER) | OUT_TOP_FRONT_LEFT) |
+                  // OUT_TOP_FRONT_CENTER) | OUT_TOP_FRONT_RIGHT) | OUT_TOP_BACK_LEFT) |
+                  // OUT_TOP_BACK_CENTER) | OUT_TOP_BACK_RIGHT)
+    AUDIO_CHANNEL_IN_LEFT = 4u,               // 0x4
+    AUDIO_CHANNEL_IN_RIGHT = 8u,              // 0x8
+    AUDIO_CHANNEL_IN_FRONT = 16u,             // 0x10
+    AUDIO_CHANNEL_IN_BACK = 32u,              // 0x20
+    AUDIO_CHANNEL_IN_LEFT_PROCESSED = 64u,    // 0x40
+    AUDIO_CHANNEL_IN_RIGHT_PROCESSED = 128u,  // 0x80
+    AUDIO_CHANNEL_IN_FRONT_PROCESSED = 256u,  // 0x100
+    AUDIO_CHANNEL_IN_BACK_PROCESSED = 512u,   // 0x200
+    AUDIO_CHANNEL_IN_PRESSURE = 1024u,        // 0x400
+    AUDIO_CHANNEL_IN_X_AXIS = 2048u,          // 0x800
+    AUDIO_CHANNEL_IN_Y_AXIS = 4096u,          // 0x1000
+    AUDIO_CHANNEL_IN_Z_AXIS = 8192u,          // 0x2000
+    AUDIO_CHANNEL_IN_VOICE_UPLINK = 16384u,   // 0x4000
+    AUDIO_CHANNEL_IN_VOICE_DNLINK = 32768u,   // 0x8000
+    AUDIO_CHANNEL_IN_MONO = 16u,              // IN_FRONT
+    AUDIO_CHANNEL_IN_STEREO = 12u,            // (IN_LEFT | IN_RIGHT)
+    AUDIO_CHANNEL_IN_FRONT_BACK = 48u,        // (IN_FRONT | IN_BACK)
+    AUDIO_CHANNEL_IN_6 = 252u,                // (((((IN_LEFT | IN_RIGHT) | IN_FRONT) | IN_BACK) |
+                                              // IN_LEFT_PROCESSED) | IN_RIGHT_PROCESSED)
+    AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO = 16400u,  // (IN_VOICE_UPLINK | IN_MONO)
+    AUDIO_CHANNEL_IN_VOICE_DNLINK_MONO = 32784u,  // (IN_VOICE_DNLINK | IN_MONO)
+    AUDIO_CHANNEL_IN_VOICE_CALL_MONO = 49168u,    // (IN_VOICE_UPLINK_MONO | IN_VOICE_DNLINK_MONO)
+    AUDIO_CHANNEL_IN_ALL =
+        65532u,  // (((((((((((((IN_LEFT | IN_RIGHT) | IN_FRONT) | IN_BACK) | IN_LEFT_PROCESSED) |
+                 // IN_RIGHT_PROCESSED) | IN_FRONT_PROCESSED) | IN_BACK_PROCESSED) | IN_PRESSURE) |
+                 // IN_X_AXIS) | IN_Y_AXIS) | IN_Z_AXIS) | IN_VOICE_UPLINK) | IN_VOICE_DNLINK)
+    AUDIO_CHANNEL_COUNT_MAX = 30u,             // 30
+    AUDIO_CHANNEL_INDEX_HDR = 2147483648u,     // (REPRESENTATION_INDEX << COUNT_MAX)
+    AUDIO_CHANNEL_INDEX_MASK_1 = 2147483649u,  // (INDEX_HDR | ((1 << 1) - 1))
+    AUDIO_CHANNEL_INDEX_MASK_2 = 2147483651u,  // (INDEX_HDR | ((1 << 2) - 1))
+    AUDIO_CHANNEL_INDEX_MASK_3 = 2147483655u,  // (INDEX_HDR | ((1 << 3) - 1))
+    AUDIO_CHANNEL_INDEX_MASK_4 = 2147483663u,  // (INDEX_HDR | ((1 << 4) - 1))
+    AUDIO_CHANNEL_INDEX_MASK_5 = 2147483679u,  // (INDEX_HDR | ((1 << 5) - 1))
+    AUDIO_CHANNEL_INDEX_MASK_6 = 2147483711u,  // (INDEX_HDR | ((1 << 6) - 1))
+    AUDIO_CHANNEL_INDEX_MASK_7 = 2147483775u,  // (INDEX_HDR | ((1 << 7) - 1))
+    AUDIO_CHANNEL_INDEX_MASK_8 = 2147483903u,  // (INDEX_HDR | ((1 << 8) - 1))
+};
+
+enum {
+    AUDIO_INTERLEAVE_LEFT = 0,
+    AUDIO_INTERLEAVE_RIGHT = 1,
+};
+
+typedef enum {
+    AUDIO_MODE_INVALID = -2,  // (-2)
+    AUDIO_MODE_CURRENT = -1,  // (-1)
+    AUDIO_MODE_NORMAL = 0,
+    AUDIO_MODE_RINGTONE = 1,
+    AUDIO_MODE_IN_CALL = 2,
+    AUDIO_MODE_IN_COMMUNICATION = 3,
+    AUDIO_MODE_CNT = 4,
+    AUDIO_MODE_MAX = 3,  // (CNT - 1)
+} audio_mode_t;
+
+enum {
+    AUDIO_DEVICE_NONE = 0u,                             // 0x0
+    AUDIO_DEVICE_BIT_IN = 2147483648u,                  // 0x80000000
+    AUDIO_DEVICE_BIT_DEFAULT = 1073741824u,             // 0x40000000
+    AUDIO_DEVICE_OUT_EARPIECE = 1u,                     // 0x1
+    AUDIO_DEVICE_OUT_SPEAKER = 2u,                      // 0x2
+    AUDIO_DEVICE_OUT_WIRED_HEADSET = 4u,                // 0x4
+    AUDIO_DEVICE_OUT_WIRED_HEADPHONE = 8u,              // 0x8
+    AUDIO_DEVICE_OUT_BLUETOOTH_SCO = 16u,               // 0x10
+    AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET = 32u,       // 0x20
+    AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT = 64u,        // 0x40
+    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP = 128u,             // 0x80
+    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 256u,  // 0x100
+    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER = 512u,     // 0x200
+    AUDIO_DEVICE_OUT_AUX_DIGITAL = 1024u,               // 0x400
+    AUDIO_DEVICE_OUT_HDMI = 1024u,                      // OUT_AUX_DIGITAL
+    AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET = 2048u,         // 0x800
+    AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET = 4096u,         // 0x1000
+    AUDIO_DEVICE_OUT_USB_ACCESSORY = 8192u,             // 0x2000
+    AUDIO_DEVICE_OUT_USB_DEVICE = 16384u,               // 0x4000
+    AUDIO_DEVICE_OUT_REMOTE_SUBMIX = 32768u,            // 0x8000
+    AUDIO_DEVICE_OUT_TELEPHONY_TX = 65536u,             // 0x10000
+    AUDIO_DEVICE_OUT_LINE = 131072u,                    // 0x20000
+    AUDIO_DEVICE_OUT_HDMI_ARC = 262144u,                // 0x40000
+    AUDIO_DEVICE_OUT_SPDIF = 524288u,                   // 0x80000
+    AUDIO_DEVICE_OUT_FM = 1048576u,                     // 0x100000
+    AUDIO_DEVICE_OUT_AUX_LINE = 2097152u,               // 0x200000
+    AUDIO_DEVICE_OUT_SPEAKER_SAFE = 4194304u,           // 0x400000
+    AUDIO_DEVICE_OUT_IP = 8388608u,                     // 0x800000
+    AUDIO_DEVICE_OUT_BUS = 16777216u,                   // 0x1000000
+    AUDIO_DEVICE_OUT_PROXY = 33554432u,                 // 0x2000000
+    AUDIO_DEVICE_OUT_USB_HEADSET = 67108864u,           // 0x4000000
+    AUDIO_DEVICE_OUT_DEFAULT = 1073741824u,             // BIT_DEFAULT
+    AUDIO_DEVICE_OUT_ALL =
+        1207959551u,  // (((((((((((((((((((((((((((OUT_EARPIECE | OUT_SPEAKER) | OUT_WIRED_HEADSET)
+                      // | OUT_WIRED_HEADPHONE) | OUT_BLUETOOTH_SCO) | OUT_BLUETOOTH_SCO_HEADSET) |
+                      // OUT_BLUETOOTH_SCO_CARKIT) | OUT_BLUETOOTH_A2DP) |
+                      // OUT_BLUETOOTH_A2DP_HEADPHONES) | OUT_BLUETOOTH_A2DP_SPEAKER) | OUT_HDMI) |
+                      // OUT_ANLG_DOCK_HEADSET) | OUT_DGTL_DOCK_HEADSET) | OUT_USB_ACCESSORY) |
+                      // OUT_USB_DEVICE) | OUT_REMOTE_SUBMIX) | OUT_TELEPHONY_TX) | OUT_LINE) |
+                      // OUT_HDMI_ARC) | OUT_SPDIF) | OUT_FM) | OUT_AUX_LINE) | OUT_SPEAKER_SAFE) |
+                      // OUT_IP) | OUT_BUS) | OUT_PROXY) | OUT_USB_HEADSET) | OUT_DEFAULT)
+    AUDIO_DEVICE_OUT_ALL_A2DP = 896u,  // ((OUT_BLUETOOTH_A2DP | OUT_BLUETOOTH_A2DP_HEADPHONES) |
+                                       // OUT_BLUETOOTH_A2DP_SPEAKER)
+    AUDIO_DEVICE_OUT_ALL_SCO =
+        112u,  // ((OUT_BLUETOOTH_SCO | OUT_BLUETOOTH_SCO_HEADSET) | OUT_BLUETOOTH_SCO_CARKIT)
+    AUDIO_DEVICE_OUT_ALL_USB =
+        67133440u,  // ((OUT_USB_ACCESSORY | OUT_USB_DEVICE) | OUT_USB_HEADSET)
+    AUDIO_DEVICE_IN_COMMUNICATION = 2147483649u,          // (BIT_IN | 0x1)
+    AUDIO_DEVICE_IN_AMBIENT = 2147483650u,                // (BIT_IN | 0x2)
+    AUDIO_DEVICE_IN_BUILTIN_MIC = 2147483652u,            // (BIT_IN | 0x4)
+    AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET = 2147483656u,  // (BIT_IN | 0x8)
+    AUDIO_DEVICE_IN_WIRED_HEADSET = 2147483664u,          // (BIT_IN | 0x10)
+    AUDIO_DEVICE_IN_AUX_DIGITAL = 2147483680u,            // (BIT_IN | 0x20)
+    AUDIO_DEVICE_IN_HDMI = 2147483680u,                   // IN_AUX_DIGITAL
+    AUDIO_DEVICE_IN_VOICE_CALL = 2147483712u,             // (BIT_IN | 0x40)
+    AUDIO_DEVICE_IN_TELEPHONY_RX = 2147483712u,           // IN_VOICE_CALL
+    AUDIO_DEVICE_IN_BACK_MIC = 2147483776u,               // (BIT_IN | 0x80)
+    AUDIO_DEVICE_IN_REMOTE_SUBMIX = 2147483904u,          // (BIT_IN | 0x100)
+    AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET = 2147484160u,      // (BIT_IN | 0x200)
+    AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET = 2147484672u,      // (BIT_IN | 0x400)
+    AUDIO_DEVICE_IN_USB_ACCESSORY = 2147485696u,          // (BIT_IN | 0x800)
+    AUDIO_DEVICE_IN_USB_DEVICE = 2147487744u,             // (BIT_IN | 0x1000)
+    AUDIO_DEVICE_IN_FM_TUNER = 2147491840u,               // (BIT_IN | 0x2000)
+    AUDIO_DEVICE_IN_TV_TUNER = 2147500032u,               // (BIT_IN | 0x4000)
+    AUDIO_DEVICE_IN_LINE = 2147516416u,                   // (BIT_IN | 0x8000)
+    AUDIO_DEVICE_IN_SPDIF = 2147549184u,                  // (BIT_IN | 0x10000)
+    AUDIO_DEVICE_IN_BLUETOOTH_A2DP = 2147614720u,         // (BIT_IN | 0x20000)
+    AUDIO_DEVICE_IN_LOOPBACK = 2147745792u,               // (BIT_IN | 0x40000)
+    AUDIO_DEVICE_IN_IP = 2148007936u,                     // (BIT_IN | 0x80000)
+    AUDIO_DEVICE_IN_BUS = 2148532224u,                    // (BIT_IN | 0x100000)
+    AUDIO_DEVICE_IN_PROXY = 2164260864u,                  // (BIT_IN | 0x1000000)
+    AUDIO_DEVICE_IN_USB_HEADSET = 2181038080u,            // (BIT_IN | 0x2000000)
+    AUDIO_DEVICE_IN_DEFAULT = 3221225472u,                // (BIT_IN | BIT_DEFAULT)
+    AUDIO_DEVICE_IN_ALL =
+        3273654271u,  // (((((((((((((((((((((((IN_COMMUNICATION | IN_AMBIENT) | IN_BUILTIN_MIC) |
+                      // IN_BLUETOOTH_SCO_HEADSET) | IN_WIRED_HEADSET) | IN_HDMI) | IN_TELEPHONY_RX)
+                      // | IN_BACK_MIC) | IN_REMOTE_SUBMIX) | IN_ANLG_DOCK_HEADSET) |
+                      // IN_DGTL_DOCK_HEADSET) | IN_USB_ACCESSORY) | IN_USB_DEVICE) | IN_FM_TUNER) |
+                      // IN_TV_TUNER) | IN_LINE) | IN_SPDIF) | IN_BLUETOOTH_A2DP) | IN_LOOPBACK) |
+                      // IN_IP) | IN_BUS) | IN_PROXY) | IN_USB_HEADSET) | IN_DEFAULT)
+    AUDIO_DEVICE_IN_ALL_SCO = 2147483656u,  // IN_BLUETOOTH_SCO_HEADSET
+    AUDIO_DEVICE_IN_ALL_USB = 2181044224u,  // ((IN_USB_ACCESSORY | IN_USB_DEVICE) | IN_USB_HEADSET)
+};
+
+typedef enum {
+    AUDIO_OUTPUT_FLAG_NONE = 0,                // 0x0
+    AUDIO_OUTPUT_FLAG_DIRECT = 1,              // 0x1
+    AUDIO_OUTPUT_FLAG_PRIMARY = 2,             // 0x2
+    AUDIO_OUTPUT_FLAG_FAST = 4,                // 0x4
+    AUDIO_OUTPUT_FLAG_DEEP_BUFFER = 8,         // 0x8
+    AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD = 16,   // 0x10
+    AUDIO_OUTPUT_FLAG_NON_BLOCKING = 32,       // 0x20
+    AUDIO_OUTPUT_FLAG_HW_AV_SYNC = 64,         // 0x40
+    AUDIO_OUTPUT_FLAG_TTS = 128,               // 0x80
+    AUDIO_OUTPUT_FLAG_RAW = 256,               // 0x100
+    AUDIO_OUTPUT_FLAG_SYNC = 512,              // 0x200
+    AUDIO_OUTPUT_FLAG_IEC958_NONAUDIO = 1024,  // 0x400
+    AUDIO_OUTPUT_FLAG_DIRECT_PCM = 8192,       // 0x2000
+    AUDIO_OUTPUT_FLAG_MMAP_NOIRQ = 16384,      // 0x4000
+    AUDIO_OUTPUT_FLAG_VOIP_RX = 32768,         // 0x8000
+} audio_output_flags_t;
+
+typedef enum {
+    AUDIO_INPUT_FLAG_NONE = 0,         // 0x0
+    AUDIO_INPUT_FLAG_FAST = 1,         // 0x1
+    AUDIO_INPUT_FLAG_HW_HOTWORD = 2,   // 0x2
+    AUDIO_INPUT_FLAG_RAW = 4,          // 0x4
+    AUDIO_INPUT_FLAG_SYNC = 8,         // 0x8
+    AUDIO_INPUT_FLAG_MMAP_NOIRQ = 16,  // 0x10
+    AUDIO_INPUT_FLAG_VOIP_TX = 32,     // 0x20
+} audio_input_flags_t;
+
+typedef enum {
+    AUDIO_USAGE_UNKNOWN = 0,
+    AUDIO_USAGE_MEDIA = 1,
+    AUDIO_USAGE_VOICE_COMMUNICATION = 2,
+    AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING = 3,
+    AUDIO_USAGE_ALARM = 4,
+    AUDIO_USAGE_NOTIFICATION = 5,
+    AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE = 6,
+    AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST = 7,
+    AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT = 8,
+    AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED = 9,
+    AUDIO_USAGE_NOTIFICATION_EVENT = 10,
+    AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY = 11,
+    AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE = 12,
+    AUDIO_USAGE_ASSISTANCE_SONIFICATION = 13,
+    AUDIO_USAGE_GAME = 14,
+    AUDIO_USAGE_VIRTUAL_SOURCE = 15,
+    AUDIO_USAGE_ASSISTANT = 16,
+    AUDIO_USAGE_CNT = 17,
+    AUDIO_USAGE_MAX = 16,  // (CNT - 1)
+} audio_usage_t;
+
+enum {
+    AUDIO_GAIN_MODE_JOINT = 1u,     // 0x1
+    AUDIO_GAIN_MODE_CHANNELS = 2u,  // 0x2
+    AUDIO_GAIN_MODE_RAMP = 4u,      // 0x4
+};
+
+typedef enum {
+    AUDIO_PORT_ROLE_NONE = 0,
+    AUDIO_PORT_ROLE_SOURCE = 1,
+    AUDIO_PORT_ROLE_SINK = 2,
+} audio_port_role_t;
+
+typedef enum {
+    AUDIO_PORT_TYPE_NONE = 0,
+    AUDIO_PORT_TYPE_DEVICE = 1,
+    AUDIO_PORT_TYPE_MIX = 2,
+    AUDIO_PORT_TYPE_SESSION = 3,
+} audio_port_type_t;
+
+enum {
+    AUDIO_PORT_CONFIG_SAMPLE_RATE = 1u,   // 0x1
+    AUDIO_PORT_CONFIG_CHANNEL_MASK = 2u,  // 0x2
+    AUDIO_PORT_CONFIG_FORMAT = 4u,        // 0x4
+    AUDIO_PORT_CONFIG_GAIN = 8u,          // 0x8
+    AUDIO_PORT_CONFIG_ALL = 15u,          // (((SAMPLE_RATE | CHANNEL_MASK) | FORMAT) | GAIN)
+};
+
+typedef enum {
+    AUDIO_LATENCY_LOW = 0,
+    AUDIO_LATENCY_NORMAL = 1,
+} audio_mix_latency_class_t;
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_COMMON_V2_0_EXPORTED_CONSTANTS_H_
diff --git a/audio/common/2.0/legacy/include/system/audio_effect-base.h b/audio/common/2.0/legacy/include/system/audio_effect-base.h
new file mode 100644
index 0000000..cd17f55
--- /dev/null
+++ b/audio/common/2.0/legacy/include/system/audio_effect-base.h
@@ -0,0 +1,101 @@
+// This file is autogenerated by hidl-gen. Do not edit manually.
+// Source: android.hardware.audio.effect@2.0
+// Root: android.hardware:hardware/interfaces
+
+#ifndef HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_EFFECT_V2_0_EXPORTED_CONSTANTS_H_
+#define HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_EFFECT_V2_0_EXPORTED_CONSTANTS_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+enum {
+    EFFECT_FLAG_TYPE_SHIFT = 0,
+    EFFECT_FLAG_TYPE_SIZE = 3,
+    EFFECT_FLAG_TYPE_MASK = 7,       // (((1 << TYPE_SIZE) - 1) << TYPE_SHIFT)
+    EFFECT_FLAG_TYPE_INSERT = 0,     // (0 << TYPE_SHIFT)
+    EFFECT_FLAG_TYPE_AUXILIARY = 1,  // (1 << TYPE_SHIFT)
+    EFFECT_FLAG_TYPE_REPLACE = 2,    // (2 << TYPE_SHIFT)
+    EFFECT_FLAG_TYPE_PRE_PROC = 3,   // (3 << TYPE_SHIFT)
+    EFFECT_FLAG_TYPE_POST_PROC = 4,  // (4 << TYPE_SHIFT)
+    EFFECT_FLAG_INSERT_SHIFT = 3,    // (TYPE_SHIFT + TYPE_SIZE)
+    EFFECT_FLAG_INSERT_SIZE = 3,
+    EFFECT_FLAG_INSERT_MASK = 56,       // (((1 << INSERT_SIZE) - 1) << INSERT_SHIFT)
+    EFFECT_FLAG_INSERT_ANY = 0,         // (0 << INSERT_SHIFT)
+    EFFECT_FLAG_INSERT_FIRST = 8,       // (1 << INSERT_SHIFT)
+    EFFECT_FLAG_INSERT_LAST = 16,       // (2 << INSERT_SHIFT)
+    EFFECT_FLAG_INSERT_EXCLUSIVE = 24,  // (3 << INSERT_SHIFT)
+    EFFECT_FLAG_VOLUME_SHIFT = 6,       // (INSERT_SHIFT + INSERT_SIZE)
+    EFFECT_FLAG_VOLUME_SIZE = 3,
+    EFFECT_FLAG_VOLUME_MASK = 448,  // (((1 << VOLUME_SIZE) - 1) << VOLUME_SHIFT)
+    EFFECT_FLAG_VOLUME_CTRL = 64,   // (1 << VOLUME_SHIFT)
+    EFFECT_FLAG_VOLUME_IND = 128,   // (2 << VOLUME_SHIFT)
+    EFFECT_FLAG_VOLUME_NONE = 0,    // (0 << VOLUME_SHIFT)
+    EFFECT_FLAG_DEVICE_SHIFT = 9,   // (VOLUME_SHIFT + VOLUME_SIZE)
+    EFFECT_FLAG_DEVICE_SIZE = 3,
+    EFFECT_FLAG_DEVICE_MASK = 3584,  // (((1 << DEVICE_SIZE) - 1) << DEVICE_SHIFT)
+    EFFECT_FLAG_DEVICE_IND = 512,    // (1 << DEVICE_SHIFT)
+    EFFECT_FLAG_DEVICE_NONE = 0,     // (0 << DEVICE_SHIFT)
+    EFFECT_FLAG_INPUT_SHIFT = 12,    // (DEVICE_SHIFT + DEVICE_SIZE)
+    EFFECT_FLAG_INPUT_SIZE = 2,
+    EFFECT_FLAG_INPUT_MASK = 12288,     // (((1 << INPUT_SIZE) - 1) << INPUT_SHIFT)
+    EFFECT_FLAG_INPUT_DIRECT = 4096,    // (1 << INPUT_SHIFT)
+    EFFECT_FLAG_INPUT_PROVIDER = 8192,  // (2 << INPUT_SHIFT)
+    EFFECT_FLAG_INPUT_BOTH = 12288,     // (3 << INPUT_SHIFT)
+    EFFECT_FLAG_OUTPUT_SHIFT = 14,      // (INPUT_SHIFT + INPUT_SIZE)
+    EFFECT_FLAG_OUTPUT_SIZE = 2,
+    EFFECT_FLAG_OUTPUT_MASK = 49152,      // (((1 << OUTPUT_SIZE) - 1) << OUTPUT_SHIFT)
+    EFFECT_FLAG_OUTPUT_DIRECT = 16384,    // (1 << OUTPUT_SHIFT)
+    EFFECT_FLAG_OUTPUT_PROVIDER = 32768,  // (2 << OUTPUT_SHIFT)
+    EFFECT_FLAG_OUTPUT_BOTH = 49152,      // (3 << OUTPUT_SHIFT)
+    EFFECT_FLAG_HW_ACC_SHIFT = 16,        // (OUTPUT_SHIFT + OUTPUT_SIZE)
+    EFFECT_FLAG_HW_ACC_SIZE = 2,
+    EFFECT_FLAG_HW_ACC_MASK = 196608,    // (((1 << HW_ACC_SIZE) - 1) << HW_ACC_SHIFT)
+    EFFECT_FLAG_HW_ACC_SIMPLE = 65536,   // (1 << HW_ACC_SHIFT)
+    EFFECT_FLAG_HW_ACC_TUNNEL = 131072,  // (2 << HW_ACC_SHIFT)
+    EFFECT_FLAG_AUDIO_MODE_SHIFT = 18,   // (HW_ACC_SHIFT + HW_ACC_SIZE)
+    EFFECT_FLAG_AUDIO_MODE_SIZE = 2,
+    EFFECT_FLAG_AUDIO_MODE_MASK = 786432,  // (((1 << AUDIO_MODE_SIZE) - 1) << AUDIO_MODE_SHIFT)
+    EFFECT_FLAG_AUDIO_MODE_IND = 262144,   // (1 << AUDIO_MODE_SHIFT)
+    EFFECT_FLAG_AUDIO_MODE_NONE = 0,       // (0 << AUDIO_MODE_SHIFT)
+    EFFECT_FLAG_AUDIO_SOURCE_SHIFT = 20,   // (AUDIO_MODE_SHIFT + AUDIO_MODE_SIZE)
+    EFFECT_FLAG_AUDIO_SOURCE_SIZE = 2,
+    EFFECT_FLAG_AUDIO_SOURCE_MASK =
+        3145728,  // (((1 << AUDIO_SOURCE_SIZE) - 1) << AUDIO_SOURCE_SHIFT)
+    EFFECT_FLAG_AUDIO_SOURCE_IND = 1048576,  // (1 << AUDIO_SOURCE_SHIFT)
+    EFFECT_FLAG_AUDIO_SOURCE_NONE = 0,       // (0 << AUDIO_SOURCE_SHIFT)
+    EFFECT_FLAG_OFFLOAD_SHIFT = 22,          // (AUDIO_SOURCE_SHIFT + AUDIO_SOURCE_SIZE)
+    EFFECT_FLAG_OFFLOAD_SIZE = 1,
+    EFFECT_FLAG_OFFLOAD_MASK = 4194304,       // (((1 << OFFLOAD_SIZE) - 1) << OFFLOAD_SHIFT)
+    EFFECT_FLAG_OFFLOAD_SUPPORTED = 4194304,  // (1 << OFFLOAD_SHIFT)
+    EFFECT_FLAG_NO_PROCESS_SHIFT = 23,        // (OFFLOAD_SHIFT + OFFLOAD_SIZE)
+    EFFECT_FLAG_NO_PROCESS_SIZE = 1,
+    EFFECT_FLAG_NO_PROCESS_MASK = 8388608,  // (((1 << NO_PROCESS_SIZE) - 1) << NO_PROCESS_SHIFT)
+    EFFECT_FLAG_NO_PROCESS = 8388608,       // (1 << NO_PROCESS_SHIFT)
+};
+
+typedef enum {
+    EFFECT_BUFFER_ACCESS_WRITE = 0,
+    EFFECT_BUFFER_ACCESS_READ = 1,
+    EFFECT_BUFFER_ACCESS_ACCUMULATE = 2,
+} effect_buffer_access_e;
+
+enum {
+    EFFECT_CONFIG_BUFFER = 1,     // 0x0001
+    EFFECT_CONFIG_SMP_RATE = 2,   // 0x0002
+    EFFECT_CONFIG_CHANNELS = 4,   // 0x0004
+    EFFECT_CONFIG_FORMAT = 8,     // 0x0008
+    EFFECT_CONFIG_ACC_MODE = 16,  // 0x0010
+    EFFECT_CONFIG_ALL = 31,       // ((((BUFFER | SMP_RATE) | CHANNELS) | FORMAT) | ACC_MODE)
+};
+
+typedef enum {
+    EFFECT_FEATURE_AUX_CHANNELS = 0,
+    EFFECT_FEATURE_CNT = 1,
+} effect_feature_e;
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif  // HIDL_GENERATED_ANDROID_HARDWARE_AUDIO_EFFECT_V2_0_EXPORTED_CONSTANTS_H_
diff --git a/audio/common/all-versions/default/Android.bp b/audio/common/all-versions/default/Android.bp
index 8f6b74c..9b82f05 100644
--- a/audio/common/all-versions/default/Android.bp
+++ b/audio/common/all-versions/default/Android.bp
@@ -16,10 +16,7 @@
 cc_library_shared {
     name: "android.hardware.audio.common-util",
     defaults: ["hidl_defaults"],
-    vendor_available: true,
-    vndk: {
-        enabled: true,
-    },
+    vendor: true,
     srcs: [
         "EffectMap.cpp",
     ],
@@ -33,7 +30,7 @@
     ],
 
     header_libs: [
-        "libaudio_system_headers",
+        "android.hardware.audio.common.legacy@2.0",
         "libhardware_headers",
     ],
 }
diff --git a/audio/common/all-versions/legacy/Android.bp b/audio/common/all-versions/legacy/Android.bp
new file mode 100644
index 0000000..2fb01dd
--- /dev/null
+++ b/audio/common/all-versions/legacy/Android.bp
@@ -0,0 +1,8 @@
+cc_library_headers {
+    name: "android.hardware.audio.common.legacy@all-versions",
+    vendor: true,
+    export_include_dirs: ["include"],
+    header_libs: ["libcutils_headers"],
+    export_header_lib_headers: ["libcutils_headers"],
+}
+
diff --git a/audio/common/all-versions/legacy/OWNERS b/audio/common/all-versions/legacy/OWNERS
new file mode 100644
index 0000000..6fdc97c
--- /dev/null
+++ b/audio/common/all-versions/legacy/OWNERS
@@ -0,0 +1,3 @@
+elaurent@google.com
+krocard@google.com
+mnaganov@google.com
diff --git a/audio/common/all-versions/legacy/include/hardware/audio.h b/audio/common/all-versions/legacy/include/hardware/audio.h
new file mode 100644
index 0000000..1ad3e0e
--- /dev/null
+++ b/audio/common/all-versions/legacy/include/hardware/audio.h
@@ -0,0 +1,709 @@
+/*
+ * 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/audio_effect.h>
+#include <hardware/hardware.h>
+#include <system/audio.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
diff --git a/audio/common/all-versions/legacy/include/hardware/audio_alsaops.h b/audio/common/all-versions/legacy/include/hardware/audio_alsaops.h
new file mode 100644
index 0000000..aa16654
--- /dev/null
+++ b/audio/common/all-versions/legacy/include/hardware/audio_alsaops.h
@@ -0,0 +1,101 @@
+/*
+ * Copyright (C) 2014 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.
+ */
+
+/* This file contains shared utility functions to handle the tinyalsa
+ * implementation for Android internal audio, generally in the hardware layer.
+ * Some routines may log a fatal error on failure, as noted.
+ */
+
+#ifndef ANDROID_AUDIO_ALSAOPS_H
+#define ANDROID_AUDIO_ALSAOPS_H
+
+#include <log/log.h>
+
+#include <system/audio.h>
+#include <tinyalsa/asoundlib.h>
+
+__BEGIN_DECLS
+
+/* Converts audio_format to pcm_format.
+ * Parameters:
+ *  format  the audio_format_t to convert
+ *
+ * Logs a fatal error if format is not a valid convertible audio_format_t.
+ */
+static inline enum pcm_format pcm_format_from_audio_format(audio_format_t format) {
+    switch (format) {
+#if HAVE_BIG_ENDIAN
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return PCM_FORMAT_S16_BE;
+        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+            return PCM_FORMAT_S24_3BE;
+        case AUDIO_FORMAT_PCM_32_BIT:
+            return PCM_FORMAT_S32_BE;
+        case AUDIO_FORMAT_PCM_8_24_BIT:
+            return PCM_FORMAT_S24_BE;
+#else
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return PCM_FORMAT_S16_LE;
+        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+            return PCM_FORMAT_S24_3LE;
+        case AUDIO_FORMAT_PCM_32_BIT:
+            return PCM_FORMAT_S32_LE;
+        case AUDIO_FORMAT_PCM_8_24_BIT:
+            return PCM_FORMAT_S24_LE;
+#endif
+        case AUDIO_FORMAT_PCM_FLOAT: /* there is no equivalent for float */
+        default:
+            LOG_ALWAYS_FATAL("pcm_format_from_audio_format: invalid audio format %#x", format);
+            return 0;
+    }
+}
+
+/* Converts pcm_format to audio_format.
+ * Parameters:
+ *  format  the pcm_format to convert
+ *
+ * Logs a fatal error if format is not a valid convertible pcm_format.
+ */
+static inline audio_format_t audio_format_from_pcm_format(enum pcm_format format) {
+    switch (format) {
+#if HAVE_BIG_ENDIAN
+        case PCM_FORMAT_S16_BE:
+            return AUDIO_FORMAT_PCM_16_BIT;
+        case PCM_FORMAT_S24_3BE:
+            return AUDIO_FORMAT_PCM_24_BIT_PACKED;
+        case PCM_FORMAT_S24_BE:
+            return AUDIO_FORMAT_PCM_8_24_BIT;
+        case PCM_FORMAT_S32_BE:
+            return AUDIO_FORMAT_PCM_32_BIT;
+#else
+        case PCM_FORMAT_S16_LE:
+            return AUDIO_FORMAT_PCM_16_BIT;
+        case PCM_FORMAT_S24_3LE:
+            return AUDIO_FORMAT_PCM_24_BIT_PACKED;
+        case PCM_FORMAT_S24_LE:
+            return AUDIO_FORMAT_PCM_8_24_BIT;
+        case PCM_FORMAT_S32_LE:
+            return AUDIO_FORMAT_PCM_32_BIT;
+#endif
+        default:
+            LOG_ALWAYS_FATAL("audio_format_from_pcm_format: invalid pcm format %#x", format);
+            return 0;
+    }
+}
+
+__END_DECLS
+
+#endif /* ANDROID_AUDIO_ALSAOPS_H */
diff --git a/audio/common/all-versions/legacy/include/hardware/audio_effect.h b/audio/common/all-versions/legacy/include/hardware/audio_effect.h
new file mode 100644
index 0000000..b91c60a
--- /dev/null
+++ b/audio/common/all-versions/legacy/include/hardware/audio_effect.h
@@ -0,0 +1,295 @@
+/*
+ * 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_EFFECT_H
+#define ANDROID_AUDIO_EFFECT_H
+
+#include <errno.h>
+#include <stdint.h>
+#include <strings.h>
+#include <sys/cdefs.h>
+#include <sys/types.h>
+
+#include <cutils/bitops.h>
+
+#include <system/audio_effect.h>
+
+__BEGIN_DECLS
+
+/////////////////////////////////////////////////
+//      Common Definitions
+/////////////////////////////////////////////////
+
+#define EFFECT_MAKE_API_VERSION(M, m) (((M) << 16) | ((m)&0xFFFF))
+#define EFFECT_API_VERSION_MAJOR(v) ((v) >> 16)
+#define EFFECT_API_VERSION_MINOR(v) ((m)&0xFFFF)
+
+/////////////////////////////////////////////////
+//      Effect control interface
+/////////////////////////////////////////////////
+
+// Effect control interface version 2.0
+#define EFFECT_CONTROL_API_VERSION EFFECT_MAKE_API_VERSION(2, 0)
+
+// Effect control interface structure: effect_interface_s
+// The effect control interface is exposed by each effect engine implementation. It consists of
+// a set of functions controlling the configuration, activation and process of the engine.
+// The functions are grouped in a structure of type effect_interface_s.
+//
+// Effect control interface handle: effect_handle_t
+// The effect_handle_t serves two purposes regarding the implementation of the effect engine:
+// - 1 it is the address of a pointer to an effect_interface_s structure where the functions
+// of the effect control API for a particular effect are located.
+// - 2 it is the address of the context of a particular effect instance.
+// A typical implementation in the effect library would define a structure as follows:
+// struct effect_module_s {
+//        const struct effect_interface_s *itfe;
+//        effect_config_t config;
+//        effect_context_t context;
+// }
+// The implementation of EffectCreate() function would then allocate a structure of this
+// type and return its address as effect_handle_t
+typedef struct effect_interface_s** effect_handle_t;
+
+// Effect control interface definition
+struct effect_interface_s {
+    ////////////////////////////////////////////////////////////////////////////////
+    //
+    //    Function:       process
+    //
+    //    Description:    Effect process function. Takes input samples as specified
+    //          (count and location) in input buffer descriptor and output processed
+    //          samples as specified in output buffer descriptor. If the buffer descriptor
+    //          is not specified the function must use either the buffer or the
+    //          buffer provider function installed by the EFFECT_CMD_SET_CONFIG command.
+    //          The effect framework will call the process() function after the EFFECT_CMD_ENABLE
+    //          command is received and until the EFFECT_CMD_DISABLE is received. When the engine
+    //          receives the EFFECT_CMD_DISABLE command it should turn off the effect gracefully
+    //          and when done indicate that it is OK to stop calling the process() function by
+    //          returning the -ENODATA status.
+    //
+    //    NOTE: the process() function implementation should be "real-time safe" that is
+    //      it should not perform blocking calls: malloc/free, sleep, read/write/open/close,
+    //      pthread_cond_wait/pthread_mutex_lock...
+    //
+    //    Input:
+    //          self:       handle to the effect interface this function
+    //              is called on.
+    //          inBuffer:   buffer descriptor indicating where to read samples to process.
+    //              If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG command.
+    //
+    //          outBuffer:   buffer descriptor indicating where to write processed samples.
+    //              If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG command.
+    //
+    //    Output:
+    //        returned value:    0 successful operation
+    //                          -ENODATA the engine has finished the disable phase and the framework
+    //                                  can stop calling process()
+    //                          -EINVAL invalid interface handle or
+    //                                  invalid input/output buffer description
+    ////////////////////////////////////////////////////////////////////////////////
+    int32_t (*process)(effect_handle_t self, audio_buffer_t* inBuffer, audio_buffer_t* outBuffer);
+    ////////////////////////////////////////////////////////////////////////////////
+    //
+    //    Function:       command
+    //
+    //    Description:    Send a command and receive a response to/from effect engine.
+    //
+    //    Input:
+    //          self:       handle to the effect interface this function
+    //              is called on.
+    //          cmdCode:    command code: the command can be a standardized command defined in
+    //              effect_command_e (see below) or a proprietary command.
+    //          cmdSize:    size of command in bytes
+    //          pCmdData:   pointer to command data
+    //          pReplyData: pointer to reply data
+    //
+    //    Input/Output:
+    //          replySize: maximum size of reply data as input
+    //                      actual size of reply data as output
+    //
+    //    Output:
+    //          returned value: 0       successful operation
+    //                          -EINVAL invalid interface handle or
+    //                                  invalid command/reply size or format according to
+    //                                  command code
+    //              The return code should be restricted to indicate problems related to this API
+    //              specification. Status related to the execution of a particular command should be
+    //              indicated as part of the reply field.
+    //
+    //          *pReplyData updated with command response
+    //
+    ////////////////////////////////////////////////////////////////////////////////
+    int32_t (*command)(effect_handle_t self, uint32_t cmdCode, uint32_t cmdSize, void* pCmdData,
+                       uint32_t* replySize, void* pReplyData);
+    ////////////////////////////////////////////////////////////////////////////////
+    //
+    //    Function:        get_descriptor
+    //
+    //    Description:    Returns the effect descriptor
+    //
+    //    Input:
+    //          self:       handle to the effect interface this function
+    //              is called on.
+    //
+    //    Input/Output:
+    //          pDescriptor:    address where to return the effect descriptor.
+    //
+    //    Output:
+    //        returned value:    0          successful operation.
+    //                          -EINVAL     invalid interface handle or invalid pDescriptor
+    //        *pDescriptor:     updated with the effect descriptor.
+    //
+    ////////////////////////////////////////////////////////////////////////////////
+    int32_t (*get_descriptor)(effect_handle_t self, effect_descriptor_t* pDescriptor);
+    ////////////////////////////////////////////////////////////////////////////////
+    //
+    //    Function:       process_reverse
+    //
+    //    Description:    Process reverse stream function. This function is used to pass
+    //          a reference stream to the effect engine. If the engine does not need a reference
+    //          stream, this function pointer can be set to NULL.
+    //          This function would typically implemented by an Echo Canceler.
+    //
+    //    Input:
+    //          self:       handle to the effect interface this function
+    //              is called on.
+    //          inBuffer:   buffer descriptor indicating where to read samples to process.
+    //              If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG_REVERSE command.
+    //
+    //          outBuffer:   buffer descriptor indicating where to write processed samples.
+    //              If NULL, use the configuration passed by EFFECT_CMD_SET_CONFIG_REVERSE command.
+    //              If the buffer and buffer provider in the configuration received by
+    //              EFFECT_CMD_SET_CONFIG_REVERSE are also NULL, do not return modified reverse
+    //              stream data
+    //
+    //    Output:
+    //        returned value:    0 successful operation
+    //                          -ENODATA the engine has finished the disable phase and the framework
+    //                                  can stop calling process_reverse()
+    //                          -EINVAL invalid interface handle or
+    //                                  invalid input/output buffer description
+    ////////////////////////////////////////////////////////////////////////////////
+    int32_t (*process_reverse)(effect_handle_t self, audio_buffer_t* inBuffer,
+                               audio_buffer_t* outBuffer);
+};
+
+/////////////////////////////////////////////////
+//      Effect library interface
+/////////////////////////////////////////////////
+
+// Effect library interface version 3.0
+// Note that EffectsFactory.c only checks the major version component, so changes to the minor
+// number can only be used for fully backwards compatible changes
+#define EFFECT_LIBRARY_API_VERSION EFFECT_MAKE_API_VERSION(3, 0)
+
+#define AUDIO_EFFECT_LIBRARY_TAG ((('A') << 24) | (('E') << 16) | (('L') << 8) | ('T'))
+
+// Every effect library must have a data structure named AUDIO_EFFECT_LIBRARY_INFO_SYM
+// and the fields of this data structure must begin with audio_effect_library_t
+
+typedef struct audio_effect_library_s {
+    // tag must be initialized to AUDIO_EFFECT_LIBRARY_TAG
+    uint32_t tag;
+    // Version of the effect library API : 0xMMMMmmmm MMMM: Major, mmmm: minor
+    uint32_t version;
+    // Name of this library
+    const char* name;
+    // Author/owner/implementor of the library
+    const char* implementor;
+
+    ////////////////////////////////////////////////////////////////////////////////
+    //
+    //    Function:        create_effect
+    //
+    //    Description:    Creates an effect engine of the specified implementation uuid and
+    //          returns an effect control interface on this engine. The function will allocate the
+    //          resources for an instance of the requested effect engine and return
+    //          a handle on the effect control interface.
+    //
+    //    Input:
+    //          uuid:    pointer to the effect uuid.
+    //          sessionId:  audio session to which this effect instance will be attached.
+    //              All effects created with the same session ID are connected in series and process
+    //              the same signal stream. Knowing that two effects are part of the same effect
+    //              chain can help the library implement some kind of optimizations.
+    //          ioId:   identifies the output or input stream this effect is directed to in
+    //              audio HAL.
+    //              For future use especially with tunneled HW accelerated effects
+    //
+    //    Input/Output:
+    //          pHandle:        address where to return the effect interface handle.
+    //
+    //    Output:
+    //        returned value:    0          successful operation.
+    //                          -ENODEV     library failed to initialize
+    //                          -EINVAL     invalid pEffectUuid or pHandle
+    //                          -ENOENT     no effect with this uuid found
+    //        *pHandle:         updated with the effect interface handle.
+    //
+    ////////////////////////////////////////////////////////////////////////////////
+    int32_t (*create_effect)(const effect_uuid_t* uuid, int32_t sessionId, int32_t ioId,
+                             effect_handle_t* pHandle);
+
+    ////////////////////////////////////////////////////////////////////////////////
+    //
+    //    Function:        release_effect
+    //
+    //    Description:    Releases the effect engine whose handle is given as argument.
+    //          All resources allocated to this particular instance of the effect are
+    //          released.
+    //
+    //    Input:
+    //          handle:         handle on the effect interface to be released.
+    //
+    //    Output:
+    //        returned value:    0          successful operation.
+    //                          -ENODEV     library failed to initialize
+    //                          -EINVAL     invalid interface handle
+    //
+    ////////////////////////////////////////////////////////////////////////////////
+    int32_t (*release_effect)(effect_handle_t handle);
+
+    ////////////////////////////////////////////////////////////////////////////////
+    //
+    //    Function:        get_descriptor
+    //
+    //    Description:    Returns the descriptor of the effect engine which implementation UUID is
+    //          given as argument.
+    //
+    //    Input/Output:
+    //          uuid:           pointer to the effect uuid.
+    //          pDescriptor:    address where to return the effect descriptor.
+    //
+    //    Output:
+    //        returned value:    0          successful operation.
+    //                          -ENODEV     library failed to initialize
+    //                          -EINVAL     invalid pDescriptor or uuid
+    //        *pDescriptor:     updated with the effect descriptor.
+    //
+    ////////////////////////////////////////////////////////////////////////////////
+    int32_t (*get_descriptor)(const effect_uuid_t* uuid, effect_descriptor_t* pDescriptor);
+} audio_effect_library_t;
+
+// Name of the hal_module_info
+#define AUDIO_EFFECT_LIBRARY_INFO_SYM AELI
+
+// Name of the hal_module_info as a string
+#define AUDIO_EFFECT_LIBRARY_INFO_SYM_AS_STR "AELI"
+
+__END_DECLS
+
+#endif  // ANDROID_AUDIO_EFFECT_H
diff --git a/audio/common/all-versions/legacy/include/hardware/audio_policy.h b/audio/common/all-versions/legacy/include/hardware/audio_policy.h
new file mode 100644
index 0000000..8cc79df
--- /dev/null
+++ b/audio/common/all-versions/legacy/include/hardware/audio_policy.h
@@ -0,0 +1,391 @@
+/*
+ * 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_POLICY_INTERFACE_H
+#define ANDROID_AUDIO_POLICY_INTERFACE_H
+
+#include <stdint.h>
+#include <sys/cdefs.h>
+#include <sys/types.h>
+
+#include <hardware/hardware.h>
+
+#include <system/audio.h>
+#include <system/audio_policy.h>
+
+__BEGIN_DECLS
+
+/**
+ * The id of this module
+ */
+#define AUDIO_POLICY_HARDWARE_MODULE_ID "audio_policy"
+
+/**
+ * Name of the audio devices to open
+ */
+#define AUDIO_POLICY_INTERFACE "policy"
+
+/* ---------------------------------------------------------------------------- */
+
+/*
+ * The audio_policy and audio_policy_service_ops structs define the
+ * communication interfaces between the platform specific audio policy manager
+ * and Android generic audio policy manager.
+ * The platform specific audio policy manager must implement methods of the
+ * audio_policy struct.
+ * This implementation makes use of the audio_policy_service_ops to control
+ * the activity and configuration of audio input and output streams.
+ *
+ * The platform specific audio policy manager is in charge of the audio
+ * routing and volume control policies for a given platform.
+ * The main roles of this module are:
+ *   - keep track of current system state (removable device connections, phone
+ *     state, user requests...).
+ *   System state changes and user actions are notified to audio policy
+ *   manager with methods of the audio_policy.
+ *
+ *   - process get_output() queries received when AudioTrack objects are
+ *     created: Those queries return a handler on an output that has been
+ *     selected, configured and opened by the audio policy manager and that
+ *     must be used by the AudioTrack when registering to the AudioFlinger
+ *     with the createTrack() method.
+ *   When the AudioTrack object is released, a release_output() query
+ *   is received and the audio policy manager can decide to close or
+ *   reconfigure the output depending on other streams using this output and
+ *   current system state.
+ *
+ *   - similarly process get_input() and release_input() queries received from
+ *     AudioRecord objects and configure audio inputs.
+ *   - process volume control requests: the stream volume is converted from
+ *     an index value (received from UI) to a float value applicable to each
+ *     output as a function of platform specific settings and current output
+ *     route (destination device). It also make sure that streams are not
+ *     muted if not allowed (e.g. camera shutter sound in some countries).
+ */
+
+/* XXX: this should be defined OUTSIDE of frameworks/base */
+struct effect_descriptor_s;
+
+struct audio_policy {
+    /*
+     * configuration functions
+     */
+
+    /* indicate a change in device connection status */
+    int (*set_device_connection_state)(struct audio_policy* pol, audio_devices_t device,
+                                       audio_policy_dev_state_t state, const char* device_address);
+
+    /* retrieve a device connection status */
+    audio_policy_dev_state_t (*get_device_connection_state)(const struct audio_policy* pol,
+                                                            audio_devices_t device,
+                                                            const char* device_address);
+
+    /* indicate a change in phone state. Valid phones states are defined
+     * by audio_mode_t */
+    void (*set_phone_state)(struct audio_policy* pol, audio_mode_t state);
+
+    /* deprecated, never called (was "indicate a change in ringer mode") */
+    void (*set_ringer_mode)(struct audio_policy* pol, uint32_t mode, uint32_t mask);
+
+    /* force using a specific device category for the specified usage */
+    void (*set_force_use)(struct audio_policy* pol, audio_policy_force_use_t usage,
+                          audio_policy_forced_cfg_t config);
+
+    /* retrieve current device category forced for a given usage */
+    audio_policy_forced_cfg_t (*get_force_use)(const struct audio_policy* pol,
+                                               audio_policy_force_use_t usage);
+
+    /* if can_mute is true, then audio streams that are marked ENFORCED_AUDIBLE
+     * can still be muted. */
+    void (*set_can_mute_enforced_audible)(struct audio_policy* pol, bool can_mute);
+
+    /* check proper initialization */
+    int (*init_check)(const struct audio_policy* pol);
+
+    /*
+     * Audio routing query functions
+     */
+
+    /* request an output appropriate for playback of the supplied stream type and
+     * parameters */
+    audio_io_handle_t (*get_output)(struct audio_policy* pol, audio_stream_type_t stream,
+                                    uint32_t samplingRate, audio_format_t format,
+                                    audio_channel_mask_t channelMask, audio_output_flags_t flags,
+                                    const audio_offload_info_t* offloadInfo);
+
+    /* indicates to the audio policy manager that the output starts being used
+     * by corresponding stream. */
+    int (*start_output)(struct audio_policy* pol, audio_io_handle_t output,
+                        audio_stream_type_t stream, audio_session_t session);
+
+    /* indicates to the audio policy manager that the output stops being used
+     * by corresponding stream. */
+    int (*stop_output)(struct audio_policy* pol, audio_io_handle_t output,
+                       audio_stream_type_t stream, audio_session_t session);
+
+    /* releases the output. */
+    void (*release_output)(struct audio_policy* pol, audio_io_handle_t output);
+
+    /* request an input appropriate for record from the supplied device with
+     * supplied parameters. */
+    audio_io_handle_t (*get_input)(struct audio_policy* pol, audio_source_t inputSource,
+                                   uint32_t samplingRate, audio_format_t format,
+                                   audio_channel_mask_t channelMask,
+                                   audio_in_acoustics_t acoustics);
+
+    /* indicates to the audio policy manager that the input starts being used */
+    int (*start_input)(struct audio_policy* pol, audio_io_handle_t input);
+
+    /* indicates to the audio policy manager that the input stops being used. */
+    int (*stop_input)(struct audio_policy* pol, audio_io_handle_t input);
+
+    /* releases the input. */
+    void (*release_input)(struct audio_policy* pol, audio_io_handle_t input);
+
+    /*
+     * volume control functions
+     */
+
+    /* initialises stream volume conversion parameters by specifying volume
+     * index range. The index range for each stream is defined by AudioService. */
+    void (*init_stream_volume)(struct audio_policy* pol, audio_stream_type_t stream, int index_min,
+                               int index_max);
+
+    /* sets the new stream volume at a level corresponding to the supplied
+     * index. The index is within the range specified by init_stream_volume() */
+    int (*set_stream_volume_index)(struct audio_policy* pol, audio_stream_type_t stream, int index);
+
+    /* retrieve current volume index for the specified stream */
+    int (*get_stream_volume_index)(const struct audio_policy* pol, audio_stream_type_t stream,
+                                   int* index);
+
+    /* sets the new stream volume at a level corresponding to the supplied
+     * index for the specified device.
+     * The index is within the range specified by init_stream_volume() */
+    int (*set_stream_volume_index_for_device)(struct audio_policy* pol, audio_stream_type_t stream,
+                                              int index, audio_devices_t device);
+
+    /* retrieve current volume index for the specified stream for the specified device */
+    int (*get_stream_volume_index_for_device)(const struct audio_policy* pol,
+                                              audio_stream_type_t stream, int* index,
+                                              audio_devices_t device);
+
+    /* return the strategy corresponding to a given stream type */
+    uint32_t (*get_strategy_for_stream)(const struct audio_policy* pol, audio_stream_type_t stream);
+
+    /* return the enabled output devices for the given stream type */
+    audio_devices_t (*get_devices_for_stream)(const struct audio_policy* pol,
+                                              audio_stream_type_t stream);
+
+    /* Audio effect management */
+    audio_io_handle_t (*get_output_for_effect)(struct audio_policy* pol,
+                                               const struct effect_descriptor_s* desc);
+
+    int (*register_effect)(struct audio_policy* pol, const struct effect_descriptor_s* desc,
+                           audio_io_handle_t output, uint32_t strategy, audio_session_t session,
+                           int id);
+
+    int (*unregister_effect)(struct audio_policy* pol, int id);
+
+    int (*set_effect_enabled)(struct audio_policy* pol, int id, bool enabled);
+
+    bool (*is_stream_active)(const struct audio_policy* pol, audio_stream_type_t stream,
+                             uint32_t in_past_ms);
+
+    bool (*is_stream_active_remotely)(const struct audio_policy* pol, audio_stream_type_t stream,
+                                      uint32_t in_past_ms);
+
+    bool (*is_source_active)(const struct audio_policy* pol, audio_source_t source);
+
+    /* dump state */
+    int (*dump)(const struct audio_policy* pol, int fd);
+
+    /* check if offload is possible for given sample rate, bitrate, duration, ... */
+    bool (*is_offload_supported)(const struct audio_policy* pol, const audio_offload_info_t* info);
+};
+
+struct audio_policy_service_ops {
+    /*
+     * Audio output Control functions
+     */
+
+    /* Opens an audio output with the requested parameters.
+     *
+     * The parameter values can indicate to use the default values in case the
+     * audio policy manager has no specific requirements for the output being
+     * opened.
+     *
+     * When the function returns, the parameter values reflect the actual
+     * values used by the audio hardware output stream.
+     *
+     * The audio policy manager can check if the proposed parameters are
+     * suitable or not and act accordingly.
+     */
+    audio_io_handle_t (*open_output)(void* service, audio_devices_t* pDevices,
+                                     uint32_t* pSamplingRate, audio_format_t* pFormat,
+                                     audio_channel_mask_t* pChannelMask, uint32_t* pLatencyMs,
+                                     audio_output_flags_t flags);
+
+    /* creates a special output that is duplicated to the two outputs passed as
+     * arguments. The duplication is performed by
+     * a special mixer thread in the AudioFlinger.
+     */
+    audio_io_handle_t (*open_duplicate_output)(void* service, audio_io_handle_t output1,
+                                               audio_io_handle_t output2);
+
+    /* closes the output stream */
+    int (*close_output)(void* service, audio_io_handle_t output);
+
+    /* suspends the output.
+     *
+     * When an output is suspended, the corresponding audio hardware output
+     * stream is placed in standby and the AudioTracks attached to the mixer
+     * thread are still processed but the output mix is discarded.
+     */
+    int (*suspend_output)(void* service, audio_io_handle_t output);
+
+    /* restores a suspended output. */
+    int (*restore_output)(void* service, audio_io_handle_t output);
+
+    /* */
+    /* Audio input Control functions */
+    /* */
+
+    /* opens an audio input
+     * deprecated - new implementations should use open_input_on_module,
+     * and the acoustics parameter is ignored
+     */
+    audio_io_handle_t (*open_input)(void* service, audio_devices_t* pDevices,
+                                    uint32_t* pSamplingRate, audio_format_t* pFormat,
+                                    audio_channel_mask_t* pChannelMask,
+                                    audio_in_acoustics_t acoustics);
+
+    /* closes an audio input */
+    int (*close_input)(void* service, audio_io_handle_t input);
+
+    /* */
+    /* misc control functions */
+    /* */
+
+    /* set a stream volume for a particular output.
+     *
+     * For the same user setting, a given stream type can have different
+     * volumes for each output (destination device) it is attached to.
+     */
+    int (*set_stream_volume)(void* service, audio_stream_type_t stream, float volume,
+                             audio_io_handle_t output, int delay_ms);
+
+    /* invalidate a stream type, causing a reroute to an unspecified new output */
+    int (*invalidate_stream)(void* service, audio_stream_type_t stream);
+
+    /* function enabling to send proprietary informations directly from audio
+     * policy manager to audio hardware interface. */
+    void (*set_parameters)(void* service, audio_io_handle_t io_handle, const char* kv_pairs,
+                           int delay_ms);
+
+    /* function enabling to receive proprietary informations directly from
+     * audio hardware interface to audio policy manager.
+     *
+     * Returns a pointer to a heap allocated string. The caller is responsible
+     * for freeing the memory for it using free().
+     */
+
+    char* (*get_parameters)(void* service, audio_io_handle_t io_handle, const char* keys);
+
+    /* request the playback of a tone on the specified stream.
+     * used for instance to replace notification sounds when playing over a
+     * telephony device during a phone call.
+     */
+    int (*start_tone)(void* service, audio_policy_tone_t tone, audio_stream_type_t stream);
+
+    int (*stop_tone)(void* service);
+
+    /* set down link audio volume. */
+    int (*set_voice_volume)(void* service, float volume, int delay_ms);
+
+    /* move effect to the specified output */
+    int (*move_effects)(void* service, audio_session_t session, audio_io_handle_t src_output,
+                        audio_io_handle_t dst_output);
+
+    /* loads an audio hw module.
+     *
+     * The module name passed is the base name of the HW module library, e.g "primary" or "a2dp".
+     * The function returns a handle on the module that will be used to specify a particular
+     * module when calling open_output_on_module() or open_input_on_module()
+     */
+    audio_module_handle_t (*load_hw_module)(void* service, const char* name);
+
+    /* Opens an audio output on a particular HW module.
+     *
+     * Same as open_output() but specifying a specific HW module on which the output must be opened.
+     */
+    audio_io_handle_t (*open_output_on_module)(void* service, audio_module_handle_t module,
+                                               audio_devices_t* pDevices, uint32_t* pSamplingRate,
+                                               audio_format_t* pFormat,
+                                               audio_channel_mask_t* pChannelMask,
+                                               uint32_t* pLatencyMs, audio_output_flags_t flags,
+                                               const audio_offload_info_t* offloadInfo);
+
+    /* Opens an audio input on a particular HW module.
+     *
+     * Same as open_input() but specifying a specific HW module on which the input must be opened.
+     * Also removed deprecated acoustics parameter
+     */
+    audio_io_handle_t (*open_input_on_module)(void* service, audio_module_handle_t module,
+                                              audio_devices_t* pDevices, uint32_t* pSamplingRate,
+                                              audio_format_t* pFormat,
+                                              audio_channel_mask_t* pChannelMask);
+};
+
+/**********************************************************************/
+
+/**
+ * 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.
+ */
+typedef struct audio_policy_module { struct hw_module_t common; } audio_policy_module_t;
+
+struct audio_policy_device {
+    /**
+     * Common methods of the audio policy device.  This *must* be the first member of
+     * audio_policy_device as users of this structure will cast a hw_device_t to
+     * audio_policy_device pointer in contexts where it's known the hw_device_t references an
+     * audio_policy_device.
+     */
+    struct hw_device_t common;
+
+    int (*create_audio_policy)(const struct audio_policy_device* device,
+                               struct audio_policy_service_ops* aps_ops, void* service,
+                               struct audio_policy** ap);
+
+    int (*destroy_audio_policy)(const struct audio_policy_device* device, struct audio_policy* ap);
+};
+
+/** convenience API for opening and closing a supported device */
+
+static inline int audio_policy_dev_open(const hw_module_t* module,
+                                        struct audio_policy_device** device) {
+    return module->methods->open(module, AUDIO_POLICY_INTERFACE, (hw_device_t**)device);
+}
+
+static inline int audio_policy_dev_close(struct audio_policy_device* device) {
+    return device->common.close(&device->common);
+}
+
+__END_DECLS
+
+#endif  // ANDROID_AUDIO_POLICY_INTERFACE_H
diff --git a/audio/common/all-versions/legacy/include/system/audio.h b/audio/common/all-versions/legacy/include/system/audio.h
new file mode 100644
index 0000000..7afa6c4
--- /dev/null
+++ b/audio/common/all-versions/legacy/include/system/audio.h
@@ -0,0 +1,1038 @@
+/*
+ * 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_CORE_H
+#define ANDROID_AUDIO_CORE_H
+
+#include <stdbool.h>
+#include <stdint.h>
+#include <stdio.h>
+#include <sys/cdefs.h>
+#include <sys/types.h>
+
+#include <cutils/bitops.h>
+
+#include "system/audio-base.h"
+
+__BEGIN_DECLS
+
+/* The enums were moved here mostly from
+ * frameworks/base/include/media/AudioSystem.h
+ */
+
+/* represents an invalid uid for tracks; the calling or client uid is often substituted. */
+#define AUDIO_UID_INVALID ((uid_t)-1)
+
+/* device address used to refer to the standard remote submix */
+#define AUDIO_REMOTE_SUBMIX_DEVICE_ADDRESS "0"
+
+/* AudioFlinger and AudioPolicy services use I/O handles to identify audio sources and sinks */
+typedef int audio_io_handle_t;
+
+/* Do not change these values without updating their counterparts
+ * in frameworks/base/media/java/android/media/AudioAttributes.java
+ */
+typedef enum {
+    AUDIO_CONTENT_TYPE_UNKNOWN = 0,
+    AUDIO_CONTENT_TYPE_SPEECH = 1,
+    AUDIO_CONTENT_TYPE_MUSIC = 2,
+    AUDIO_CONTENT_TYPE_MOVIE = 3,
+    AUDIO_CONTENT_TYPE_SONIFICATION = 4,
+
+    AUDIO_CONTENT_TYPE_CNT,
+    AUDIO_CONTENT_TYPE_MAX = AUDIO_CONTENT_TYPE_CNT - 1,
+} audio_content_type_t;
+
+typedef uint32_t audio_flags_mask_t;
+
+/* Do not change these values without updating their counterparts
+ * in frameworks/base/media/java/android/media/AudioAttributes.java
+ */
+enum {
+    AUDIO_FLAG_NONE = 0x0,
+    AUDIO_FLAG_AUDIBILITY_ENFORCED = 0x1,
+    AUDIO_FLAG_SECURE = 0x2,
+    AUDIO_FLAG_SCO = 0x4,
+    AUDIO_FLAG_BEACON = 0x8,
+    AUDIO_FLAG_HW_AV_SYNC = 0x10,
+    AUDIO_FLAG_HW_HOTWORD = 0x20,
+    AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY = 0x40,
+    AUDIO_FLAG_BYPASS_MUTE = 0x80,
+    AUDIO_FLAG_LOW_LATENCY = 0x100,
+    AUDIO_FLAG_DEEP_BUFFER = 0x200,
+};
+
+/* Audio attributes */
+#define AUDIO_ATTRIBUTES_TAGS_MAX_SIZE 256
+typedef struct {
+    audio_content_type_t content_type;
+    audio_usage_t usage;
+    audio_source_t source;
+    audio_flags_mask_t flags;
+    char tags[AUDIO_ATTRIBUTES_TAGS_MAX_SIZE]; /* UTF8 */
+} __attribute__((packed)) audio_attributes_t;  // sent through Binder;
+
+/* a unique ID allocated by AudioFlinger for use as an audio_io_handle_t, audio_session_t,
+ * effect ID (int), audio_module_handle_t, and audio_patch_handle_t.
+ * Audio port IDs (audio_port_handle_t) are allocated by AudioPolicy
+ * in a different namespace than AudioFlinger unique IDs.
+ */
+typedef int audio_unique_id_t;
+
+/* Possible uses for an audio_unique_id_t */
+typedef enum {
+    AUDIO_UNIQUE_ID_USE_UNSPECIFIED = 0,
+    AUDIO_UNIQUE_ID_USE_SESSION = 1,  // for allocated sessions, not special AUDIO_SESSION_*
+    AUDIO_UNIQUE_ID_USE_MODULE = 2,
+    AUDIO_UNIQUE_ID_USE_EFFECT = 3,
+    AUDIO_UNIQUE_ID_USE_PATCH = 4,
+    AUDIO_UNIQUE_ID_USE_OUTPUT = 5,
+    AUDIO_UNIQUE_ID_USE_INPUT = 6,
+    AUDIO_UNIQUE_ID_USE_PLAYER = 7,
+    AUDIO_UNIQUE_ID_USE_MAX = 8,  // must be a power-of-two
+    AUDIO_UNIQUE_ID_USE_MASK = AUDIO_UNIQUE_ID_USE_MAX - 1
+} audio_unique_id_use_t;
+
+/* Return the use of an audio_unique_id_t */
+static inline audio_unique_id_use_t audio_unique_id_get_use(audio_unique_id_t id) {
+    return (audio_unique_id_use_t)(id & AUDIO_UNIQUE_ID_USE_MASK);
+}
+
+/* Reserved audio_unique_id_t values.  FIXME: not a complete list. */
+#define AUDIO_UNIQUE_ID_ALLOCATE AUDIO_SESSION_ALLOCATE
+
+/* A channel mask per se only defines the presence or absence of a channel, not the order.
+ * But see AUDIO_INTERLEAVE_* below for the platform convention of order.
+ *
+ * audio_channel_mask_t is an opaque type and its internal layout should not
+ * be assumed as it may change in the future.
+ * Instead, always use the functions declared in this header to examine.
+ *
+ * These are the current representations:
+ *
+ *   AUDIO_CHANNEL_REPRESENTATION_POSITION
+ *     is a channel mask representation for position assignment.
+ *     Each low-order bit corresponds to the spatial position of a transducer (output),
+ *     or interpretation of channel (input).
+ *     The user of a channel mask needs to know the context of whether it is for output or input.
+ *     The constants AUDIO_CHANNEL_OUT_* or AUDIO_CHANNEL_IN_* apply to the bits portion.
+ *     It is not permitted for no bits to be set.
+ *
+ *   AUDIO_CHANNEL_REPRESENTATION_INDEX
+ *     is a channel mask representation for index assignment.
+ *     Each low-order bit corresponds to a selected channel.
+ *     There is no platform interpretation of the various bits.
+ *     There is no concept of output or input.
+ *     It is not permitted for no bits to be set.
+ *
+ * All other representations are reserved for future use.
+ *
+ * Warning: current representation distinguishes between input and output, but this will not the be
+ * case in future revisions of the platform. Wherever there is an ambiguity between input and output
+ * that is currently resolved by checking the channel mask, the implementer should look for ways to
+ * fix it with additional information outside of the mask.
+ */
+typedef uint32_t audio_channel_mask_t;
+
+/* log(2) of maximum number of representations, not part of public API */
+#define AUDIO_CHANNEL_REPRESENTATION_LOG2 2
+
+/* The return value is undefined if the channel mask is invalid. */
+static inline uint32_t audio_channel_mask_get_bits(audio_channel_mask_t channel) {
+    return channel & ((1 << AUDIO_CHANNEL_COUNT_MAX) - 1);
+}
+
+typedef uint32_t audio_channel_representation_t;
+
+/* The return value is undefined if the channel mask is invalid. */
+static inline audio_channel_representation_t audio_channel_mask_get_representation(
+    audio_channel_mask_t channel) {
+    // The right shift should be sufficient, but also "and" for safety in case mask is not 32 bits
+    return (audio_channel_representation_t)((channel >> AUDIO_CHANNEL_COUNT_MAX) &
+                                            ((1 << AUDIO_CHANNEL_REPRESENTATION_LOG2) - 1));
+}
+
+/* Returns true if the channel mask is valid,
+ * or returns false for AUDIO_CHANNEL_NONE, AUDIO_CHANNEL_INVALID, and other invalid values.
+ * This function is unable to determine whether a channel mask for position assignment
+ * is invalid because an output mask has an invalid output bit set,
+ * or because an input mask has an invalid input bit set.
+ * All other APIs that take a channel mask assume that it is valid.
+ */
+static inline bool audio_channel_mask_is_valid(audio_channel_mask_t channel) {
+    uint32_t bits = audio_channel_mask_get_bits(channel);
+    audio_channel_representation_t representation = audio_channel_mask_get_representation(channel);
+    switch (representation) {
+        case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+        case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+            break;
+        default:
+            bits = 0;
+            break;
+    }
+    return bits != 0;
+}
+
+/* Not part of public API */
+static inline audio_channel_mask_t audio_channel_mask_from_representation_and_bits(
+    audio_channel_representation_t representation, uint32_t bits) {
+    return (audio_channel_mask_t)((representation << AUDIO_CHANNEL_COUNT_MAX) | bits);
+}
+
+/* This enum is deprecated */
+typedef enum {
+    AUDIO_IN_ACOUSTICS_NONE = 0,
+    AUDIO_IN_ACOUSTICS_AGC_ENABLE = 0x0001,
+    AUDIO_IN_ACOUSTICS_AGC_DISABLE = 0,
+    AUDIO_IN_ACOUSTICS_NS_ENABLE = 0x0002,
+    AUDIO_IN_ACOUSTICS_NS_DISABLE = 0,
+    AUDIO_IN_ACOUSTICS_TX_IIR_ENABLE = 0x0004,
+    AUDIO_IN_ACOUSTICS_TX_DISABLE = 0,
+} audio_in_acoustics_t;
+
+typedef uint32_t audio_devices_t;
+/**
+ * Stub audio output device. Used in policy configuration file on platforms without audio outputs.
+ * This alias value to AUDIO_DEVICE_OUT_DEFAULT is only used in the audio policy context.
+ */
+#define AUDIO_DEVICE_OUT_STUB AUDIO_DEVICE_OUT_DEFAULT
+/**
+ * Stub audio input device. Used in policy configuration file on platforms without audio inputs.
+ * This alias value to AUDIO_DEVICE_IN_DEFAULT is only used in the audio policy context.
+ */
+#define AUDIO_DEVICE_IN_STUB AUDIO_DEVICE_IN_DEFAULT
+
+/* Additional information about compressed streams offloaded to
+ * hardware playback
+ * The version and size fields must be initialized by the caller by using
+ * one of the constants defined here.
+ * Must be aligned to transmit as raw memory through Binder.
+ */
+typedef struct {
+    uint16_t version;                   // version of the info structure
+    uint16_t size;                      // total size of the structure including version and size
+    uint32_t sample_rate;               // sample rate in Hz
+    audio_channel_mask_t channel_mask;  // channel mask
+    audio_format_t format;              // audio format
+    audio_stream_type_t stream_type;    // stream type
+    uint32_t bit_rate;                  // bit rate in bits per second
+    int64_t duration_us;                // duration in microseconds, -1 if unknown
+    bool has_video;                     // true if stream is tied to a video stream
+    bool is_streaming;                  // true if streaming, false if local playback
+    uint32_t bit_width;
+    uint32_t offload_buffer_size;  // offload fragment size
+    audio_usage_t usage;
+} __attribute__((aligned(8))) audio_offload_info_t;
+
+#define AUDIO_MAKE_OFFLOAD_INFO_VERSION(maj, min) ((((maj)&0xff) << 8) | ((min)&0xff))
+
+#define AUDIO_OFFLOAD_INFO_VERSION_0_1 AUDIO_MAKE_OFFLOAD_INFO_VERSION(0, 1)
+#define AUDIO_OFFLOAD_INFO_VERSION_CURRENT AUDIO_OFFLOAD_INFO_VERSION_0_1
+
+static const audio_offload_info_t AUDIO_INFO_INITIALIZER = {
+    /* .version = */ AUDIO_OFFLOAD_INFO_VERSION_CURRENT,
+    /* .size = */ sizeof(audio_offload_info_t),
+    /* .sample_rate = */ 0,
+    /* .channel_mask = */ 0,
+    /* .format = */ AUDIO_FORMAT_DEFAULT,
+    /* .stream_type = */ AUDIO_STREAM_VOICE_CALL,
+    /* .bit_rate = */ 0,
+    /* .duration_us = */ 0,
+    /* .has_video = */ false,
+    /* .is_streaming = */ false,
+    /* .bit_width = */ 16,
+    /* .offload_buffer_size = */ 0,
+    /* .usage = */ AUDIO_USAGE_UNKNOWN};
+
+/* common audio stream configuration parameters
+ * You should memset() the entire structure to zero before use to
+ * ensure forward compatibility
+ * Must be aligned to transmit as raw memory through Binder.
+ */
+struct __attribute__((aligned(8))) audio_config {
+    uint32_t sample_rate;
+    audio_channel_mask_t channel_mask;
+    audio_format_t format;
+    audio_offload_info_t offload_info;
+    uint32_t frame_count;
+};
+typedef struct audio_config audio_config_t;
+
+static const audio_config_t AUDIO_CONFIG_INITIALIZER = {
+    /* .sample_rate = */ 0,
+    /* .channel_mask = */ AUDIO_CHANNEL_NONE,
+    /* .format = */ AUDIO_FORMAT_DEFAULT,
+    /* .offload_info = */
+    {/* .version = */ AUDIO_OFFLOAD_INFO_VERSION_CURRENT,
+     /* .size = */ sizeof(audio_offload_info_t),
+     /* .sample_rate = */ 0,
+     /* .channel_mask = */ 0,
+     /* .format = */ AUDIO_FORMAT_DEFAULT,
+     /* .stream_type = */ AUDIO_STREAM_VOICE_CALL,
+     /* .bit_rate = */ 0,
+     /* .duration_us = */ 0,
+     /* .has_video = */ false,
+     /* .is_streaming = */ false,
+     /* .bit_width = */ 16,
+     /* .offload_buffer_size = */ 0,
+     /* .usage = */ AUDIO_USAGE_UNKNOWN},
+    /* .frame_count = */ 0,
+};
+
+struct audio_config_base {
+    uint32_t sample_rate;
+    audio_channel_mask_t channel_mask;
+    audio_format_t format;
+};
+
+typedef struct audio_config_base audio_config_base_t;
+
+static const audio_config_base_t AUDIO_CONFIG_BASE_INITIALIZER = {
+    /* .sample_rate = */ 0,
+    /* .channel_mask = */ AUDIO_CHANNEL_NONE,
+    /* .format = */ AUDIO_FORMAT_DEFAULT};
+
+/* audio hw module handle functions or structures referencing a module */
+typedef int audio_module_handle_t;
+
+/******************************
+ *  Volume control
+ *****************************/
+
+/** 3 dB headroom are allowed on float samples (3db = 10^(3/20) = 1.412538).
+ * See: https://developer.android.com/reference/android/media/AudioTrack.html#write(float[], int,
+ * int, int)
+ */
+#define FLOAT_NOMINAL_RANGE_HEADROOM 1.412538
+
+/* If the audio hardware supports gain control on some audio paths,
+ * the platform can expose them in the audio_policy.conf file. The audio HAL
+ * will then implement gain control functions that will use the following data
+ * structures. */
+
+typedef uint32_t audio_gain_mode_t;
+
+/* An audio_gain struct is a representation of a gain stage.
+ * A gain stage is always attached to an audio port. */
+struct audio_gain {
+    audio_gain_mode_t mode;            /* e.g. AUDIO_GAIN_MODE_JOINT */
+    audio_channel_mask_t channel_mask; /* channels which gain an be controlled.
+                                          N/A if AUDIO_GAIN_MODE_CHANNELS is not supported */
+    int min_value;                     /* minimum gain value in millibels */
+    int max_value;                     /* maximum gain value in millibels */
+    int default_value;                 /* default gain value in millibels */
+    unsigned int step_value;           /* gain step in millibels */
+    unsigned int min_ramp_ms;          /* minimum ramp duration in ms */
+    unsigned int max_ramp_ms;          /* maximum ramp duration in ms */
+};
+
+/* The gain configuration structure is used to get or set the gain values of a
+ * given port */
+struct audio_gain_config {
+    int index;                         /* index of the corresponding audio_gain in the
+                                          audio_port gains[] table */
+    audio_gain_mode_t mode;            /* mode requested for this command */
+    audio_channel_mask_t channel_mask; /* channels which gain value follows.
+                                          N/A in joint mode */
+
+    // note this "8" is not FCC_8, so it won't need to be changed for > 8 channels
+    int values[sizeof(audio_channel_mask_t) * 8]; /* gain values in millibels
+                              for each channel ordered from LSb to MSb in
+                              channel mask. The number of values is 1 in joint
+                              mode or popcount(channel_mask) */
+    unsigned int ramp_duration_ms;                /* ramp duration in ms */
+};
+
+/******************************
+ *  Routing control
+ *****************************/
+
+/* Types defined here are used to describe an audio source or sink at internal
+ * framework interfaces (audio policy, patch panel) or at the audio HAL.
+ * Sink and sources are grouped in a concept of “audio port” representing an
+ * audio end point at the edge of the system managed by the module exposing
+ * the interface. */
+
+/* Each port has a unique ID or handle allocated by policy manager */
+typedef int audio_port_handle_t;
+
+/* the maximum length for the human-readable device name */
+#define AUDIO_PORT_MAX_NAME_LEN 128
+
+/* maximum audio device address length */
+#define AUDIO_DEVICE_MAX_ADDRESS_LEN 32
+
+/* extension for audio port configuration structure when the audio port is a
+ * hardware device */
+struct audio_port_config_device_ext {
+    audio_module_handle_t hw_module;            /* module the device is attached to */
+    audio_devices_t type;                       /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */
+    char address[AUDIO_DEVICE_MAX_ADDRESS_LEN]; /* device address. "" if N/A */
+};
+
+/* extension for audio port configuration structure when the audio port is a
+ * sub mix */
+struct audio_port_config_mix_ext {
+    audio_module_handle_t hw_module; /* module the stream is attached to */
+    audio_io_handle_t handle;        /* I/O handle of the input/output stream */
+    union {
+        // TODO: change use case for output streams: use strategy and mixer attributes
+        audio_stream_type_t stream;
+        audio_source_t source;
+    } usecase;
+};
+
+/* extension for audio port configuration structure when the audio port is an
+ * audio session */
+struct audio_port_config_session_ext {
+    audio_session_t session; /* audio session */
+};
+
+/* audio port configuration structure used to specify a particular configuration of
+ * an audio port */
+struct audio_port_config {
+    audio_port_handle_t id;            /* port unique ID */
+    audio_port_role_t role;            /* sink or source */
+    audio_port_type_t type;            /* device, mix ... */
+    unsigned int config_mask;          /* e.g AUDIO_PORT_CONFIG_ALL */
+    unsigned int sample_rate;          /* sampling rate in Hz */
+    audio_channel_mask_t channel_mask; /* channel mask if applicable */
+    audio_format_t format;             /* format if applicable */
+    struct audio_gain_config gain;     /* gain to apply if applicable */
+    union {
+        struct audio_port_config_device_ext device;   /* device specific info */
+        struct audio_port_config_mix_ext mix;         /* mix specific info */
+        struct audio_port_config_session_ext session; /* session specific info */
+    } ext;
+};
+
+/* max number of sampling rates in audio port */
+#define AUDIO_PORT_MAX_SAMPLING_RATES 32
+/* max number of channel masks in audio port */
+#define AUDIO_PORT_MAX_CHANNEL_MASKS 32
+/* max number of audio formats in audio port */
+#define AUDIO_PORT_MAX_FORMATS 32
+/* max number of gain controls in audio port */
+#define AUDIO_PORT_MAX_GAINS 16
+
+/* extension for audio port structure when the audio port is a hardware device */
+struct audio_port_device_ext {
+    audio_module_handle_t hw_module; /* module the device is attached to */
+    audio_devices_t type;            /* device type (e.g AUDIO_DEVICE_OUT_SPEAKER) */
+    char address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
+};
+
+/* extension for audio port structure when the audio port is a sub mix */
+struct audio_port_mix_ext {
+    audio_module_handle_t hw_module;         /* module the stream is attached to */
+    audio_io_handle_t handle;                /* I/O handle of the input.output stream */
+    audio_mix_latency_class_t latency_class; /* latency class */
+    // other attributes: routing strategies
+};
+
+/* extension for audio port structure when the audio port is an audio session */
+struct audio_port_session_ext {
+    audio_session_t session; /* audio session */
+};
+
+struct audio_port {
+    audio_port_handle_t id; /* port unique ID */
+    audio_port_role_t role; /* sink or source */
+    audio_port_type_t type; /* device, mix ... */
+    char name[AUDIO_PORT_MAX_NAME_LEN];
+    unsigned int num_sample_rates; /* number of sampling rates in following array */
+    unsigned int sample_rates[AUDIO_PORT_MAX_SAMPLING_RATES];
+    unsigned int num_channel_masks; /* number of channel masks in following array */
+    audio_channel_mask_t channel_masks[AUDIO_PORT_MAX_CHANNEL_MASKS];
+    unsigned int num_formats; /* number of formats in following array */
+    audio_format_t formats[AUDIO_PORT_MAX_FORMATS];
+    unsigned int num_gains; /* number of gains in following array */
+    struct audio_gain gains[AUDIO_PORT_MAX_GAINS];
+    struct audio_port_config active_config; /* current audio port configuration */
+    union {
+        struct audio_port_device_ext device;
+        struct audio_port_mix_ext mix;
+        struct audio_port_session_ext session;
+    } ext;
+};
+
+/* An audio patch represents a connection between one or more source ports and
+ * one or more sink ports. Patches are connected and disconnected by audio policy manager or by
+ * applications via framework APIs.
+ * Each patch is identified by a handle at the interface used to create that patch. For instance,
+ * when a patch is created by the audio HAL, the HAL allocates and returns a handle.
+ * This handle is unique to a given audio HAL hardware module.
+ * But the same patch receives another system wide unique handle allocated by the framework.
+ * This unique handle is used for all transactions inside the framework.
+ */
+typedef int audio_patch_handle_t;
+
+#define AUDIO_PATCH_PORTS_MAX 16
+
+struct audio_patch {
+    audio_patch_handle_t id;  /* patch unique ID */
+    unsigned int num_sources; /* number of sources in following array */
+    struct audio_port_config sources[AUDIO_PATCH_PORTS_MAX];
+    unsigned int num_sinks; /* number of sinks in following array */
+    struct audio_port_config sinks[AUDIO_PATCH_PORTS_MAX];
+};
+
+/* a HW synchronization source returned by the audio HAL */
+typedef uint32_t audio_hw_sync_t;
+
+/* an invalid HW synchronization source indicating an error */
+#define AUDIO_HW_SYNC_INVALID 0
+
+/**
+ * Mmap buffer descriptor returned by audio_stream->create_mmap_buffer().
+ * note\ Used by streams opened in mmap mode.
+ */
+struct audio_mmap_buffer_info {
+    void* shared_memory_address; /**< base address of mmap memory buffer.
+                                      For use by local process only */
+    int32_t shared_memory_fd;    /**< FD for mmap memory buffer */
+    int32_t buffer_size_frames;  /**< total buffer size in frames */
+    int32_t burst_size_frames;   /**< transfer size granularity in frames */
+};
+
+/**
+ * Mmap buffer read/write position returned by audio_stream->get_mmap_position().
+ * note\ Used by streams opened in mmap mode.
+ */
+struct audio_mmap_position {
+    int64_t time_nanoseconds; /**< timestamp in ns, CLOCK_MONOTONIC */
+    int32_t position_frames;  /**< increasing 32 bit frame count reset when stream->stop()
+                                   is called */
+};
+
+static inline bool audio_is_output_device(audio_devices_t device) {
+    if (((device & AUDIO_DEVICE_BIT_IN) == 0) && (popcount(device) == 1) &&
+        ((device & ~AUDIO_DEVICE_OUT_ALL) == 0))
+        return true;
+    else
+        return false;
+}
+
+static inline bool audio_is_input_device(audio_devices_t device) {
+    if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+        device &= ~AUDIO_DEVICE_BIT_IN;
+        if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_IN_ALL) == 0)) return true;
+    }
+    return false;
+}
+
+static inline bool audio_is_output_devices(audio_devices_t device) {
+    return (device & AUDIO_DEVICE_BIT_IN) == 0;
+}
+
+static inline bool audio_is_a2dp_in_device(audio_devices_t device) {
+    if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+        device &= ~AUDIO_DEVICE_BIT_IN;
+        if ((popcount(device) == 1) && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP)) return true;
+    }
+    return false;
+}
+
+static inline bool audio_is_a2dp_out_device(audio_devices_t device) {
+    if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_A2DP))
+        return true;
+    else
+        return false;
+}
+
+// Deprecated - use audio_is_a2dp_out_device() instead
+static inline bool audio_is_a2dp_device(audio_devices_t device) {
+    return audio_is_a2dp_out_device(device);
+}
+
+static inline bool audio_is_bluetooth_sco_device(audio_devices_t device) {
+    if ((device & AUDIO_DEVICE_BIT_IN) == 0) {
+        if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_OUT_ALL_SCO) == 0)) return true;
+    } else {
+        device &= ~AUDIO_DEVICE_BIT_IN;
+        if ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) == 0))
+            return true;
+    }
+
+    return false;
+}
+
+static inline bool audio_is_usb_out_device(audio_devices_t device) {
+    return ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_USB));
+}
+
+static inline bool audio_is_usb_in_device(audio_devices_t device) {
+    if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+        device &= ~AUDIO_DEVICE_BIT_IN;
+        if (popcount(device) == 1 && (device & AUDIO_DEVICE_IN_ALL_USB) != 0) return true;
+    }
+    return false;
+}
+
+/* OBSOLETE - use audio_is_usb_out_device() instead. */
+static inline bool audio_is_usb_device(audio_devices_t device) {
+    return audio_is_usb_out_device(device);
+}
+
+static inline bool audio_is_remote_submix_device(audio_devices_t device) {
+    if ((audio_is_output_devices(device) &&
+         (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX) == AUDIO_DEVICE_OUT_REMOTE_SUBMIX) ||
+        (!audio_is_output_devices(device) &&
+         (device & AUDIO_DEVICE_IN_REMOTE_SUBMIX) == AUDIO_DEVICE_IN_REMOTE_SUBMIX))
+        return true;
+    else
+        return false;
+}
+
+/* Returns true if:
+ *  representation is valid, and
+ *  there is at least one channel bit set which _could_ correspond to an input channel, and
+ *  there are no channel bits set which could _not_ correspond to an input channel.
+ * Otherwise returns false.
+ */
+static inline bool audio_is_input_channel(audio_channel_mask_t channel) {
+    uint32_t bits = audio_channel_mask_get_bits(channel);
+    switch (audio_channel_mask_get_representation(channel)) {
+        case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+            if (bits & ~AUDIO_CHANNEL_IN_ALL) {
+                bits = 0;
+            }
+        // fall through
+        case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+            return bits != 0;
+        default:
+            return false;
+    }
+}
+
+/* Returns true if:
+ *  representation is valid, and
+ *  there is at least one channel bit set which _could_ correspond to an output channel, and
+ *  there are no channel bits set which could _not_ correspond to an output channel.
+ * Otherwise returns false.
+ */
+static inline bool audio_is_output_channel(audio_channel_mask_t channel) {
+    uint32_t bits = audio_channel_mask_get_bits(channel);
+    switch (audio_channel_mask_get_representation(channel)) {
+        case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+            if (bits & ~AUDIO_CHANNEL_OUT_ALL) {
+                bits = 0;
+            }
+        // fall through
+        case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+            return bits != 0;
+        default:
+            return false;
+    }
+}
+
+/* Returns the number of channels from an input channel mask,
+ * used in the context of audio input or recording.
+ * If a channel bit is set which could _not_ correspond to an input channel,
+ * it is excluded from the count.
+ * Returns zero if the representation is invalid.
+ */
+static inline uint32_t audio_channel_count_from_in_mask(audio_channel_mask_t channel) {
+    uint32_t bits = audio_channel_mask_get_bits(channel);
+    switch (audio_channel_mask_get_representation(channel)) {
+        case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+            // TODO: We can now merge with from_out_mask and remove anding
+            bits &= AUDIO_CHANNEL_IN_ALL;
+        // fall through
+        case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+            return popcount(bits);
+        default:
+            return 0;
+    }
+}
+
+/* Returns the number of channels from an output channel mask,
+ * used in the context of audio output or playback.
+ * If a channel bit is set which could _not_ correspond to an output channel,
+ * it is excluded from the count.
+ * Returns zero if the representation is invalid.
+ */
+static inline uint32_t audio_channel_count_from_out_mask(audio_channel_mask_t channel) {
+    uint32_t bits = audio_channel_mask_get_bits(channel);
+    switch (audio_channel_mask_get_representation(channel)) {
+        case AUDIO_CHANNEL_REPRESENTATION_POSITION:
+            // TODO: We can now merge with from_in_mask and remove anding
+            bits &= AUDIO_CHANNEL_OUT_ALL;
+        // fall through
+        case AUDIO_CHANNEL_REPRESENTATION_INDEX:
+            return popcount(bits);
+        default:
+            return 0;
+    }
+}
+
+/* Derive a channel mask for index assignment from a channel count.
+ * Returns the matching channel mask,
+ * or AUDIO_CHANNEL_NONE if the channel count is zero,
+ * or AUDIO_CHANNEL_INVALID if the channel count exceeds AUDIO_CHANNEL_COUNT_MAX.
+ */
+static inline audio_channel_mask_t audio_channel_mask_for_index_assignment_from_count(
+    uint32_t channel_count) {
+    if (channel_count == 0) {
+        return AUDIO_CHANNEL_NONE;
+    }
+    if (channel_count > AUDIO_CHANNEL_COUNT_MAX) {
+        return AUDIO_CHANNEL_INVALID;
+    }
+    uint32_t bits = (1 << channel_count) - 1;
+    return audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_INDEX,
+                                                           bits);
+}
+
+/* Derive an output channel mask for position assignment from a channel count.
+ * This is to be used when the content channel mask is unknown. The 1, 2, 4, 5, 6, 7 and 8 channel
+ * cases are mapped to the standard game/home-theater layouts, but note that 4 is mapped to quad,
+ * and not stereo + FC + mono surround. A channel count of 3 is arbitrarily mapped to stereo + FC
+ * for continuity with stereo.
+ * Returns the matching channel mask,
+ * or AUDIO_CHANNEL_NONE if the channel count is zero,
+ * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the
+ * configurations for which a default output channel mask is defined.
+ */
+static inline audio_channel_mask_t audio_channel_out_mask_from_count(uint32_t channel_count) {
+    uint32_t bits;
+    switch (channel_count) {
+        case 0:
+            return AUDIO_CHANNEL_NONE;
+        case 1:
+            bits = AUDIO_CHANNEL_OUT_MONO;
+            break;
+        case 2:
+            bits = AUDIO_CHANNEL_OUT_STEREO;
+            break;
+        case 3:
+            bits = AUDIO_CHANNEL_OUT_STEREO | AUDIO_CHANNEL_OUT_FRONT_CENTER;
+            break;
+        case 4:  // 4.0
+            bits = AUDIO_CHANNEL_OUT_QUAD;
+            break;
+        case 5:  // 5.0
+            bits = AUDIO_CHANNEL_OUT_QUAD | AUDIO_CHANNEL_OUT_FRONT_CENTER;
+            break;
+        case 6:  // 5.1
+            bits = AUDIO_CHANNEL_OUT_5POINT1;
+            break;
+        case 7:  // 6.1
+            bits = AUDIO_CHANNEL_OUT_5POINT1 | AUDIO_CHANNEL_OUT_BACK_CENTER;
+            break;
+        case 8:
+            bits = AUDIO_CHANNEL_OUT_7POINT1;
+            break;
+        // FIXME FCC_8
+        default:
+            return AUDIO_CHANNEL_INVALID;
+    }
+    return audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_POSITION,
+                                                           bits);
+}
+
+/* Derive a default input channel mask from a channel count.
+ * Assumes a position mask for mono and stereo, or an index mask for channel counts > 2.
+ * Returns the matching channel mask,
+ * or AUDIO_CHANNEL_NONE if the channel count is zero,
+ * or AUDIO_CHANNEL_INVALID if the channel count exceeds that of the
+ * configurations for which a default input channel mask is defined.
+ */
+static inline audio_channel_mask_t audio_channel_in_mask_from_count(uint32_t channel_count) {
+    uint32_t bits;
+    switch (channel_count) {
+        case 0:
+            return AUDIO_CHANNEL_NONE;
+        case 1:
+            bits = AUDIO_CHANNEL_IN_MONO;
+            break;
+        case 2:
+            bits = AUDIO_CHANNEL_IN_STEREO;
+            break;
+        case 3:
+        case 4:
+        case 5:
+        case 6:
+        case 7:
+        case 8:
+            // FIXME FCC_8
+            return audio_channel_mask_for_index_assignment_from_count(channel_count);
+        default:
+            return AUDIO_CHANNEL_INVALID;
+    }
+    return audio_channel_mask_from_representation_and_bits(AUDIO_CHANNEL_REPRESENTATION_POSITION,
+                                                           bits);
+}
+
+static inline bool audio_is_valid_format(audio_format_t format) {
+    switch (format & AUDIO_FORMAT_MAIN_MASK) {
+        case AUDIO_FORMAT_PCM:
+            switch (format) {
+                case AUDIO_FORMAT_PCM_16_BIT:
+                case AUDIO_FORMAT_PCM_8_BIT:
+                case AUDIO_FORMAT_PCM_32_BIT:
+                case AUDIO_FORMAT_PCM_8_24_BIT:
+                case AUDIO_FORMAT_PCM_FLOAT:
+                case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+                    return true;
+                default:
+                    return false;
+            }
+        /* not reached */
+        case AUDIO_FORMAT_MP3:
+        case AUDIO_FORMAT_AMR_NB:
+        case AUDIO_FORMAT_AMR_WB:
+        case AUDIO_FORMAT_AAC:
+        case AUDIO_FORMAT_AAC_ADTS:
+        case AUDIO_FORMAT_HE_AAC_V1:
+        case AUDIO_FORMAT_HE_AAC_V2:
+        case AUDIO_FORMAT_VORBIS:
+        case AUDIO_FORMAT_OPUS:
+        case AUDIO_FORMAT_AC3:
+        case AUDIO_FORMAT_E_AC3:
+        case AUDIO_FORMAT_DTS:
+        case AUDIO_FORMAT_DTS_HD:
+        case AUDIO_FORMAT_IEC61937:
+        case AUDIO_FORMAT_DOLBY_TRUEHD:
+        case AUDIO_FORMAT_QCELP:
+        case AUDIO_FORMAT_EVRC:
+        case AUDIO_FORMAT_EVRCB:
+        case AUDIO_FORMAT_EVRCWB:
+        case AUDIO_FORMAT_AAC_ADIF:
+        case AUDIO_FORMAT_AMR_WB_PLUS:
+        case AUDIO_FORMAT_MP2:
+        case AUDIO_FORMAT_EVRCNW:
+        case AUDIO_FORMAT_FLAC:
+        case AUDIO_FORMAT_ALAC:
+        case AUDIO_FORMAT_APE:
+        case AUDIO_FORMAT_WMA:
+        case AUDIO_FORMAT_WMA_PRO:
+        case AUDIO_FORMAT_DSD:
+        case AUDIO_FORMAT_AC4:
+        case AUDIO_FORMAT_LDAC:
+            return true;
+        default:
+            return false;
+    }
+}
+
+/**
+ * Extract the primary format, eg. PCM, AC3, etc.
+ */
+static inline audio_format_t audio_get_main_format(audio_format_t format) {
+    return (audio_format_t)(format & AUDIO_FORMAT_MAIN_MASK);
+}
+
+/**
+ * Is the data plain PCM samples that can be scaled and mixed?
+ */
+static inline bool audio_is_linear_pcm(audio_format_t format) {
+    return (audio_get_main_format(format) == AUDIO_FORMAT_PCM);
+}
+
+/**
+ * For this format, is the number of PCM audio frames directly proportional
+ * to the number of data bytes?
+ *
+ * In other words, is the format transported as PCM audio samples,
+ * but not necessarily scalable or mixable.
+ * This returns true for real PCM, but also for AUDIO_FORMAT_IEC61937,
+ * which is transported as 16 bit PCM audio, but where the encoded data
+ * cannot be mixed or scaled.
+ */
+static inline bool audio_has_proportional_frames(audio_format_t format) {
+    audio_format_t mainFormat = audio_get_main_format(format);
+    return (mainFormat == AUDIO_FORMAT_PCM || mainFormat == AUDIO_FORMAT_IEC61937);
+}
+
+static inline size_t audio_bytes_per_sample(audio_format_t format) {
+    size_t size = 0;
+
+    switch (format) {
+        case AUDIO_FORMAT_PCM_32_BIT:
+        case AUDIO_FORMAT_PCM_8_24_BIT:
+            size = sizeof(int32_t);
+            break;
+        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+            size = sizeof(uint8_t) * 3;
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+        case AUDIO_FORMAT_IEC61937:
+            size = sizeof(int16_t);
+            break;
+        case AUDIO_FORMAT_PCM_8_BIT:
+            size = sizeof(uint8_t);
+            break;
+        case AUDIO_FORMAT_PCM_FLOAT:
+            size = sizeof(float);
+            break;
+        default:
+            break;
+    }
+    return size;
+}
+
+/* converts device address to string sent to audio HAL via set_parameters */
+static inline char* audio_device_address_to_parameter(audio_devices_t device, const char* address) {
+    const size_t kSize = AUDIO_DEVICE_MAX_ADDRESS_LEN + sizeof("a2dp_sink_address=");
+    char param[kSize];
+
+    if (device & AUDIO_DEVICE_OUT_ALL_A2DP)
+        snprintf(param, kSize, "%s=%s", "a2dp_sink_address", address);
+    else if (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)
+        snprintf(param, kSize, "%s=%s", "mix", address);
+    else
+        snprintf(param, kSize, "%s", address);
+
+    return strdup(param);
+}
+
+static inline bool audio_device_is_digital(audio_devices_t device) {
+    if ((device & AUDIO_DEVICE_BIT_IN) != 0) {
+        // input
+        return (~AUDIO_DEVICE_BIT_IN & device &
+                (AUDIO_DEVICE_IN_ALL_USB | AUDIO_DEVICE_IN_HDMI | AUDIO_DEVICE_IN_SPDIF |
+                 AUDIO_DEVICE_IN_IP | AUDIO_DEVICE_IN_BUS)) != 0;
+    } else {
+        // output
+        return (device &
+                (AUDIO_DEVICE_OUT_ALL_USB | AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_HDMI_ARC |
+                 AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_IP | AUDIO_DEVICE_OUT_BUS)) != 0;
+    }
+}
+
+// Unique effect ID (can be generated from the following site:
+//  http://www.itu.int/ITU-T/asn1/uuid.html)
+// This struct is used for effects identification and in soundtrigger.
+typedef struct audio_uuid_s {
+    uint32_t timeLow;
+    uint16_t timeMid;
+    uint16_t timeHiAndVersion;
+    uint16_t clockSeq;
+    uint8_t node[6];
+} audio_uuid_t;
+
+__END_DECLS
+
+/**
+ * List of known audio HAL modules. This is the base name of the audio HAL
+ * library composed of the "audio." prefix, one of the base names below and
+ * a suffix specific to the device.
+ * e.g: audio.primary.goldfish.so or audio.a2dp.default.so
+ *
+ * The same module names are used in audio policy configuration files.
+ */
+
+#define AUDIO_HARDWARE_MODULE_ID_PRIMARY "primary"
+#define AUDIO_HARDWARE_MODULE_ID_A2DP "a2dp"
+#define AUDIO_HARDWARE_MODULE_ID_USB "usb"
+#define AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX "r_submix"
+#define AUDIO_HARDWARE_MODULE_ID_CODEC_OFFLOAD "codec_offload"
+#define AUDIO_HARDWARE_MODULE_ID_STUB "stub"
+
+/**
+ * Multi-Stream Decoder (MSD) HAL service name. MSD HAL is used to mix
+ * encoded streams together with PCM streams, producing re-encoded
+ * streams or PCM streams.
+ *
+ * The service must register itself using this name, and audioserver
+ * tries to instantiate a device factory using this name as well.
+ * Note that the HIDL implementation library file name *must* have the
+ * suffix "msd" in order to be picked up by HIDL that is:
+ *
+ *   android.hardware.audio@x.x-implmsd.so
+ */
+#define AUDIO_HAL_SERVICE_NAME_MSD "msd"
+
+/**
+ * Parameter definitions.
+ * Note that in the framework code it's recommended to use AudioParameter.h
+ * instead of these preprocessor defines, and for sure avoid just copying
+ * the constant values.
+ */
+
+#define AUDIO_PARAMETER_VALUE_ON "on"
+#define AUDIO_PARAMETER_VALUE_OFF "off"
+
+/**
+ *  audio device parameters
+ */
+
+/* BT SCO Noise Reduction + Echo Cancellation parameters */
+#define AUDIO_PARAMETER_KEY_BT_NREC "bt_headset_nrec"
+
+/* Get a new HW synchronization source identifier.
+ * Return a valid source (positive integer) or AUDIO_HW_SYNC_INVALID if an error occurs
+ * or no HW sync is available. */
+#define AUDIO_PARAMETER_HW_AV_SYNC "hw_av_sync"
+
+/* Screen state */
+#define AUDIO_PARAMETER_KEY_SCREEN_STATE "screen_state"
+
+/**
+ *  audio stream parameters
+ */
+
+#define AUDIO_PARAMETER_STREAM_ROUTING "routing"             /* audio_devices_t */
+#define AUDIO_PARAMETER_STREAM_FORMAT "format"               /* audio_format_t */
+#define AUDIO_PARAMETER_STREAM_CHANNELS "channels"           /* audio_channel_mask_t */
+#define AUDIO_PARAMETER_STREAM_FRAME_COUNT "frame_count"     /* size_t */
+#define AUDIO_PARAMETER_STREAM_INPUT_SOURCE "input_source"   /* audio_source_t */
+#define AUDIO_PARAMETER_STREAM_SAMPLING_RATE "sampling_rate" /* uint32_t */
+
+#define AUDIO_PARAMETER_DEVICE_CONNECT "connect"       /* audio_devices_t */
+#define AUDIO_PARAMETER_DEVICE_DISCONNECT "disconnect" /* audio_devices_t */
+
+/* Enable mono audio playback if 1, else should be 0. */
+#define AUDIO_PARAMETER_MONO_OUTPUT "mono_output"
+
+/* Set the HW synchronization source for an output stream. */
+#define AUDIO_PARAMETER_STREAM_HW_AV_SYNC "hw_av_sync"
+
+/* Query supported formats. The response is a '|' separated list of strings from
+ * audio_format_t enum e.g: "sup_formats=AUDIO_FORMAT_PCM_16_BIT" */
+#define AUDIO_PARAMETER_STREAM_SUP_FORMATS "sup_formats"
+/* Query supported channel masks. The response is a '|' separated list of strings from
+ * audio_channel_mask_t enum e.g: "sup_channels=AUDIO_CHANNEL_OUT_STEREO|AUDIO_CHANNEL_OUT_MONO" */
+#define AUDIO_PARAMETER_STREAM_SUP_CHANNELS "sup_channels"
+/* Query supported sampling rates. The response is a '|' separated list of integer values e.g:
+ * "sup_sampling_rates=44100|48000" */
+#define AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES "sup_sampling_rates"
+
+#define AUDIO_PARAMETER_VALUE_LIST_SEPARATOR "|"
+
+/**
+ * audio codec parameters
+ */
+
+#define AUDIO_OFFLOAD_CODEC_PARAMS "music_offload_codec_param"
+#define AUDIO_OFFLOAD_CODEC_BIT_PER_SAMPLE "music_offload_bit_per_sample"
+#define AUDIO_OFFLOAD_CODEC_BIT_RATE "music_offload_bit_rate"
+#define AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE "music_offload_avg_bit_rate"
+#define AUDIO_OFFLOAD_CODEC_ID "music_offload_codec_id"
+#define AUDIO_OFFLOAD_CODEC_BLOCK_ALIGN "music_offload_block_align"
+#define AUDIO_OFFLOAD_CODEC_SAMPLE_RATE "music_offload_sample_rate"
+#define AUDIO_OFFLOAD_CODEC_ENCODE_OPTION "music_offload_encode_option"
+#define AUDIO_OFFLOAD_CODEC_NUM_CHANNEL "music_offload_num_channels"
+#define AUDIO_OFFLOAD_CODEC_DOWN_SAMPLING "music_offload_down_sampling"
+#define AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES "delay_samples"
+#define AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES "padding_samples"
+
+// FIXME: a temporary declaration for the incall music flag, will be removed when
+// declared in types.hal for audio HAL V4.0 and auto imported to audio-base.h
+#define AUDIO_OUTPUT_FLAG_INCALL_MUSIC 0x10000
+
+#endif  // ANDROID_AUDIO_CORE_H
diff --git a/audio/common/all-versions/legacy/include/system/audio_effect.h b/audio/common/all-versions/legacy/include/system/audio_effect.h
new file mode 100644
index 0000000..f99f604
--- /dev/null
+++ b/audio/common/all-versions/legacy/include/system/audio_effect.h
@@ -0,0 +1,528 @@
+/*
+ * Copyright (C) 2016 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_EFFECT_CORE_H
+#define ANDROID_AUDIO_EFFECT_CORE_H
+
+#include "system/audio.h"
+#include "system/audio_effect-base.h"
+
+__BEGIN_DECLS
+
+/////////////////////////////////////////////////
+//      Common Definitions
+/////////////////////////////////////////////////
+
+//
+//--- Effect descriptor structure effect_descriptor_t
+//
+
+// This format is used for both "type" and "uuid" fields of the effect descriptor structure.
+// - When used for effect type and the engine is implementing and effect corresponding to a standard
+// OpenSL ES interface, this ID must be the one defined in OpenSLES_IID.h for that interface.
+// - When used as uuid, it should be a unique UUID for this particular implementation.
+typedef audio_uuid_t effect_uuid_t;
+
+// Maximum length of character strings in structures defines by this API.
+#define EFFECT_STRING_LEN_MAX 64
+
+// NULL UUID definition (matches SL_IID_NULL_)
+#define EFFECT_UUID_INITIALIZER                                                    \
+    {                                                                              \
+        0xec7178ec, 0xe5e1, 0x4432, 0xa3f4, { 0x46, 0x57, 0xe6, 0x79, 0x52, 0x10 } \
+    }
+static const effect_uuid_t EFFECT_UUID_NULL_ = EFFECT_UUID_INITIALIZER;
+static const effect_uuid_t* const EFFECT_UUID_NULL = &EFFECT_UUID_NULL_;
+static const char* const EFFECT_UUID_NULL_STR = "ec7178ec-e5e1-4432-a3f4-4657e6795210";
+
+// The effect descriptor contains necessary information to facilitate the enumeration of the effect
+// engines present in a library.
+typedef struct effect_descriptor_s {
+    effect_uuid_t type;    // UUID of to the OpenSL ES interface implemented by this effect
+    effect_uuid_t uuid;    // UUID for this particular implementation
+    uint32_t apiVersion;   // Version of the effect control API implemented
+    uint32_t flags;        // effect engine capabilities/requirements flags (see below)
+    uint16_t cpuLoad;      // CPU load indication (see below)
+    uint16_t memoryUsage;  // Data Memory usage (see below)
+    char name[EFFECT_STRING_LEN_MAX];         // human readable effect name
+    char implementor[EFFECT_STRING_LEN_MAX];  // human readable effect implementor name
+} effect_descriptor_t;
+
+/////////////////////////////////////////////////
+//      Effect control interface
+/////////////////////////////////////////////////
+
+//
+//--- Standardized command codes for command() function
+//
+enum effect_command_e {
+    EFFECT_CMD_INIT,                // initialize effect engine
+    EFFECT_CMD_SET_CONFIG,          // configure effect engine (see effect_config_t)
+    EFFECT_CMD_RESET,               // reset effect engine
+    EFFECT_CMD_ENABLE,              // enable effect process
+    EFFECT_CMD_DISABLE,             // disable effect process
+    EFFECT_CMD_SET_PARAM,           // set parameter immediately (see effect_param_t)
+    EFFECT_CMD_SET_PARAM_DEFERRED,  // set parameter deferred
+    EFFECT_CMD_SET_PARAM_COMMIT,    // commit previous set parameter deferred
+    EFFECT_CMD_GET_PARAM,           // get parameter
+    EFFECT_CMD_SET_DEVICE,          // set audio device (see audio.h, audio_devices_t)
+    EFFECT_CMD_SET_VOLUME,          // set volume
+    EFFECT_CMD_SET_AUDIO_MODE,      // set the audio mode (normal, ring, ...)
+    EFFECT_CMD_SET_CONFIG_REVERSE,  // configure effect engine reverse stream(see effect_config_t)
+    EFFECT_CMD_SET_INPUT_DEVICE,    // set capture device (see audio.h, audio_devices_t)
+    EFFECT_CMD_GET_CONFIG,          // read effect engine configuration
+    EFFECT_CMD_GET_CONFIG_REVERSE,  // read configure effect engine reverse stream configuration
+    EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS,  // get all supported configurations for a feature.
+    EFFECT_CMD_GET_FEATURE_CONFIG,             // get current feature configuration
+    EFFECT_CMD_SET_FEATURE_CONFIG,             // set current feature configuration
+    EFFECT_CMD_SET_AUDIO_SOURCE,               // set the audio source (see audio.h, audio_source_t)
+    EFFECT_CMD_OFFLOAD,                        // set if effect thread is an offload one,
+                                               // send the ioHandle of the effect thread
+    EFFECT_CMD_FIRST_PROPRIETARY = 0x10000     // first proprietary command code
+};
+
+//==================================================================================================
+// command: EFFECT_CMD_INIT
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Initialize effect engine: All configurations return to default
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: 0
+//  data: N/A
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(int)
+//  data: status
+//==================================================================================================
+// command: EFFECT_CMD_SET_CONFIG
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Apply new audio parameters configurations for input and output buffers
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(effect_config_t)
+//  data: effect_config_t
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(int)
+//  data: status
+//==================================================================================================
+// command: EFFECT_CMD_RESET
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Reset the effect engine. Keep configuration but resets state and buffer content
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: 0
+//  data: N/A
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: 0
+//  data: N/A
+//==================================================================================================
+// command: EFFECT_CMD_ENABLE
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Enable the process. Called by the framework before the first call to process()
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: 0
+//  data: N/A
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(int)
+//  data: status
+//==================================================================================================
+// command: EFFECT_CMD_DISABLE
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Disable the process. Called by the framework after the last call to process()
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: 0
+//  data: N/A
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(int)
+//  data: status
+//==================================================================================================
+// command: EFFECT_CMD_SET_PARAM
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Set a parameter and apply it immediately
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(effect_param_t) + size of param and value
+//  data: effect_param_t + param + value. See effect_param_t definition below for value offset
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(int)
+//  data: status
+//==================================================================================================
+// command: EFFECT_CMD_SET_PARAM_DEFERRED
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Set a parameter but apply it only when receiving EFFECT_CMD_SET_PARAM_COMMIT command
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(effect_param_t) + size of param and value
+//  data: effect_param_t + param + value. See effect_param_t definition below for value offset
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: 0
+//  data: N/A
+//==================================================================================================
+// command: EFFECT_CMD_SET_PARAM_COMMIT
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Apply all previously received EFFECT_CMD_SET_PARAM_DEFERRED commands
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: 0
+//  data: N/A
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(int)
+//  data: status
+//==================================================================================================
+// command: EFFECT_CMD_GET_PARAM
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Get a parameter value
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(effect_param_t) + size of param
+//  data: effect_param_t + param
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(effect_param_t) + size of param and value
+//  data: effect_param_t + param + value. See effect_param_t definition below for value offset
+//==================================================================================================
+// command: EFFECT_CMD_SET_DEVICE
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Set the rendering device the audio output path is connected to. See audio.h, audio_devices_t
+//  for device values.
+//  The effect implementation must set EFFECT_FLAG_DEVICE_IND flag in its descriptor to receive this
+//  command when the device changes
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(uint32_t)
+//  data: uint32_t
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: 0
+//  data: N/A
+//==================================================================================================
+// command: EFFECT_CMD_SET_VOLUME
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Set and get volume. Used by audio framework to delegate volume control to effect engine.
+//  The effect implementation must set EFFECT_FLAG_VOLUME_IND or EFFECT_FLAG_VOLUME_CTRL flag in
+//  its descriptor to receive this command before every call to process() function
+//  If EFFECT_FLAG_VOLUME_CTRL flag is set in the effect descriptor, the effect engine must return
+//  the volume that should be applied before the effect is processed. The overall volume (the volume
+//  actually applied by the effect engine multiplied by the returned value) should match the value
+//  indicated in the command.
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: n * sizeof(uint32_t)
+//  data: volume for each channel defined in effect_config_t for output buffer expressed in
+//      8.24 fixed point format
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: n * sizeof(uint32_t) / 0
+//  data: - if EFFECT_FLAG_VOLUME_CTRL is set in effect descriptor:
+//              volume for each channel defined in effect_config_t for output buffer expressed in
+//              8.24 fixed point format
+//        - if EFFECT_FLAG_VOLUME_CTRL is not set in effect descriptor:
+//              N/A
+//  It is legal to receive a null pointer as pReplyData in which case the effect framework has
+//  delegated volume control to another effect
+//==================================================================================================
+// command: EFFECT_CMD_SET_AUDIO_MODE
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Set the audio mode. The effect implementation must set EFFECT_FLAG_AUDIO_MODE_IND flag in its
+//  descriptor to receive this command when the audio mode changes.
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(uint32_t)
+//  data: audio_mode_t
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: 0
+//  data: N/A
+//==================================================================================================
+// command: EFFECT_CMD_SET_CONFIG_REVERSE
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Apply new audio parameters configurations for input and output buffers of reverse stream.
+//  An example of reverse stream is the echo reference supplied to an Acoustic Echo Canceler.
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(effect_config_t)
+//  data: effect_config_t
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(int)
+//  data: status
+//==================================================================================================
+// command: EFFECT_CMD_SET_INPUT_DEVICE
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Set the capture device the audio input path is connected to. See audio.h, audio_devices_t
+//  for device values.
+//  The effect implementation must set EFFECT_FLAG_DEVICE_IND flag in its descriptor to receive this
+//  command when the device changes
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(uint32_t)
+//  data: uint32_t
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: 0
+//  data: N/A
+//==================================================================================================
+// command: EFFECT_CMD_GET_CONFIG
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Read audio parameters configurations for input and output buffers
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: 0
+//  data: N/A
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(effect_config_t)
+//  data: effect_config_t
+//==================================================================================================
+// command: EFFECT_CMD_GET_CONFIG_REVERSE
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Read audio parameters configurations for input and output buffers of reverse stream
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: 0
+//  data: N/A
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(effect_config_t)
+//  data: effect_config_t
+//==================================================================================================
+// command: EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Queries for supported configurations for a particular feature (e.g. get the supported
+// combinations of main and auxiliary channels for a noise suppressor).
+// The command parameter is the feature identifier (See effect_feature_e for a list of defined
+// features) followed by the maximum number of configuration descriptor to return.
+// The reply is composed of:
+//  - status (uint32_t):
+//          - 0 if feature is supported
+//          - -ENOSYS if the feature is not supported,
+//          - -ENOMEM if the feature is supported but the total number of supported configurations
+//          exceeds the maximum number indicated by the caller.
+//  - total number of supported configurations (uint32_t)
+//  - an array of configuration descriptors.
+// The actual number of descriptors returned must not exceed the maximum number indicated by
+// the caller.
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: 2 x sizeof(uint32_t)
+//  data: effect_feature_e + maximum number of configurations to return
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: 2 x sizeof(uint32_t) + n x sizeof (<config descriptor>)
+//  data: status + total number of configurations supported + array of n config descriptors
+//==================================================================================================
+// command: EFFECT_CMD_GET_FEATURE_CONFIG
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Retrieves current configuration for a given feature.
+// The reply status is:
+//      - 0 if feature is supported
+//      - -ENOSYS if the feature is not supported,
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(uint32_t)
+//  data: effect_feature_e
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(uint32_t) + sizeof (<config descriptor>)
+//  data: status + config descriptor
+//==================================================================================================
+// command: EFFECT_CMD_SET_FEATURE_CONFIG
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Sets current configuration for a given feature.
+// The reply status is:
+//      - 0 if feature is supported
+//      - -ENOSYS if the feature is not supported,
+//      - -EINVAL if the configuration is invalid
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(uint32_t) + sizeof (<config descriptor>)
+//  data: effect_feature_e + config descriptor
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(uint32_t)
+//  data: status
+//==================================================================================================
+// command: EFFECT_CMD_SET_AUDIO_SOURCE
+//--------------------------------------------------------------------------------------------------
+// description:
+//  Set the audio source the capture path is configured for (Camcorder, voice recognition...).
+//  See audio.h, audio_source_t for values.
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(uint32_t)
+//  data: uint32_t
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: 0
+//  data: N/A
+//==================================================================================================
+// command: EFFECT_CMD_OFFLOAD
+//--------------------------------------------------------------------------------------------------
+// description:
+//  1.indicate if the playback thread the effect is attached to is offloaded or not
+//  2.update the io handle of the playback thread the effect is attached to
+//--------------------------------------------------------------------------------------------------
+// command format:
+//  size: sizeof(effect_offload_param_t)
+//  data: effect_offload_param_t
+//--------------------------------------------------------------------------------------------------
+// reply format:
+//  size: sizeof(uint32_t)
+//  data: uint32_t
+//--------------------------------------------------------------------------------------------------
+// command: EFFECT_CMD_FIRST_PROPRIETARY
+//--------------------------------------------------------------------------------------------------
+// description:
+//  All proprietary effect commands must use command codes above this value. The size and format of
+//  command and response fields is free in this case
+//==================================================================================================
+
+// Audio buffer descriptor used by process(), bufferProvider() functions and buffer_config_t
+// structure. Multi-channel audio is always interleaved. The channel order is from LSB to MSB with
+// regard to the channel mask definition in audio.h, audio_channel_mask_t e.g :
+// Stereo: left, right
+// 5 point 1: front left, front right, front center, low frequency, back left, back right
+// The buffer size is expressed in frame count, a frame being composed of samples for all
+// channels at a given time. Frame size for unspecified format (AUDIO_FORMAT_OTHER) is 8 bit by
+// definition
+typedef struct audio_buffer_s {
+    size_t frameCount;  // number of frames in buffer
+    union {
+        void* raw;     // raw pointer to start of buffer
+        float* f32;    // pointer to float 32 bit data at start of buffer
+        int32_t* s32;  // pointer to signed 32 bit data at start of buffer
+        int16_t* s16;  // pointer to signed 16 bit data at start of buffer
+        uint8_t* u8;   // pointer to unsigned 8 bit data at start of buffer
+    };
+} audio_buffer_t;
+
+// The buffer_provider_s structure contains functions that can be used
+// by the effect engine process() function to query and release input
+// or output audio buffer.
+// The getBuffer() function is called to retrieve a buffer where data
+// should read from or written to by process() function.
+// The releaseBuffer() function MUST be called when the buffer retrieved
+// with getBuffer() is not needed anymore.
+// The process function should use the buffer provider mechanism to retrieve
+// input or output buffer if the inBuffer or outBuffer passed as argument is NULL
+// and the buffer configuration (buffer_config_t) given by the EFFECT_CMD_SET_CONFIG
+// command did not specify an audio buffer.
+
+typedef int32_t (*buffer_function_t)(void* cookie, audio_buffer_t* buffer);
+
+typedef struct buffer_provider_s {
+    buffer_function_t getBuffer;      // retrieve next buffer
+    buffer_function_t releaseBuffer;  // release used buffer
+    void* cookie;                     // for use by client of buffer provider functions
+} buffer_provider_t;
+
+// The buffer_config_s structure specifies the input or output audio format
+// to be used by the effect engine.
+typedef struct buffer_config_s {
+    audio_buffer_t buffer;  // buffer for use by process() function if not passed explicitly
+    uint32_t samplingRate;  // sampling rate
+    uint32_t channels;      // channel mask (see audio_channel_mask_t in audio.h)
+    buffer_provider_t bufferProvider;  // buffer provider
+    uint8_t format;                    // Audio format (see audio_format_t in audio.h)
+    uint8_t accessMode;  // read/write or accumulate in buffer (effect_buffer_access_e)
+    uint16_t mask;       // indicates which of the above fields is valid
+} buffer_config_t;
+
+// EFFECT_FEATURE_AUX_CHANNELS feature configuration descriptor. Describe a combination
+// of main and auxiliary channels supported
+typedef struct channel_config_s {
+    audio_channel_mask_t main_channels;  // channel mask for main channels
+    audio_channel_mask_t aux_channels;   // channel mask for auxiliary channels
+} channel_config_t;
+
+// effect_config_s structure is used to configure audio parameters and buffers for effect engine
+// input and output.
+typedef struct effect_config_s {
+    buffer_config_t inputCfg;
+    buffer_config_t outputCfg;
+} effect_config_t;
+
+// effect_param_s structure describes the format of the pCmdData argument of EFFECT_CMD_SET_PARAM
+// command and pCmdData and pReplyData of EFFECT_CMD_GET_PARAM command.
+// psize and vsize represent the actual size of parameter and value.
+//
+// NOTE: the start of value field inside the data field is always on a 32 bit boundary:
+//
+//  +-----------+
+//  | status    | sizeof(int)
+//  +-----------+
+//  | psize     | sizeof(int)
+//  +-----------+
+//  | vsize     | sizeof(int)
+//  +-----------+
+//  |           |   |           |
+//  ~ parameter ~   > psize     |
+//  |           |   |           >  ((psize - 1)/sizeof(int) + 1) * sizeof(int)
+//  +-----------+               |
+//  | padding   |               |
+//  +-----------+
+//  |           |   |
+//  ~ value     ~   > vsize
+//  |           |   |
+//  +-----------+
+
+typedef struct effect_param_s {
+    int32_t status;  // Transaction status (unused for command, used for reply)
+    uint32_t psize;  // Parameter size
+    uint32_t vsize;  // Value size
+    char data[];     // Start of Parameter + Value data
+} effect_param_t;
+
+// Maximum effect_param_t size
+#define EFFECT_PARAM_SIZE_MAX 65536
+
+// structure used by EFFECT_CMD_OFFLOAD command
+typedef struct effect_offload_param_s {
+    bool isOffload;  // true if the playback thread the effect is attached to is offloaded
+    int ioHandle;    // io handle of the playback thread the effect is attached to
+} effect_offload_param_t;
+
+__END_DECLS
+
+#endif  // ANDROID_AUDIO_EFFECT_CORE_H
diff --git a/audio/common/all-versions/util/Android.bp b/audio/common/all-versions/util/Android.bp
index 5d33a3a..7132667 100644
--- a/audio/common/all-versions/util/Android.bp
+++ b/audio/common/all-versions/util/Android.bp
@@ -1,10 +1,7 @@
 cc_library_headers {
     name: "android.hardware.audio.common.util@all-versions",
     defaults: ["hidl_defaults"],
-    vendor_available: true,
-    vndk: {
-        enabled: true,
-    },
+    vendor: true,
 
     export_include_dirs: ["include"],
 }