libhardware: add audio defines and HAL headers

Change-Id: Iae523059834c252f0d39f7efc6e15832e6273101
Signed-off-by: Dima Zavin <dima@android.com>

4 files changed, 1181 insertions, 0 deletions

diff --git a/include/hardware/audio.h b/include/hardware/audio.h
new file mode 100644
index 0000000..8f2ac0c
--- /dev/null
+++ b/include/hardware/audio.h
@@ -0,0 +1,393 @@
+/*
+ * 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 <sys/cdefs.h>
+#include <sys/types.h>
+
+#include <cutils/bitops.h>
+
+__BEGIN_DECLS
+
+/* The enums were moved here mostly from
+ * frameworks/base/include/media/AudioSystem.h
+ */
+
+typedef int audio_io_handle_t;
+
+/* Audio stream types */
+typedef enum {
+    AUDIO_STREAM_DEFAULT          = -1,
+    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, /* Sounds that cannot be muted by user and must be routed to speaker */
+    AUDIO_STREAM_DTMF             = 8,
+    AUDIO_STREAM_TTS              = 9,
+
+    AUDIO_STREAM_CNT,
+    AUDIO_STREAM_MAX              = AUDIO_STREAM_CNT - 1,
+} audio_stream_type_t;
+
+/* Do not change these values without updating their counterparts
+ * in media/java/android/media/MediaRecorder.java!
+ */
+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_CNT,
+    AUDIO_SOURCE_MAX                 = AUDIO_SOURCE_CNT - 1,
+} audio_source_t;
+
+/* special audio session values
+ * (XXX: should this be living in the audio effects land?)
+ */
+typedef enum {
+    /* session for effects attached to a particular output stream
+     * (value must be less than 0)
+     */
+    AUDIO_SESSION_OUTPUT_STAGE = -1,
+
+    /* session for effects applied to output mix. These effects can
+     * be moved by audio policy manager to another output stream
+     * (value must be 0)
+     */
+    AUDIO_SESSION_OUTPUT_MIX = 0,
+} audio_session_t;
+
+/* Audio sub formats (see enum audio_format). */
+
+/* PCM sub formats */
+typedef enum {
+    AUDIO_FORMAT_PCM_SUB_16_BIT          = 0x1, /* DO NOT CHANGE */
+    AUDIO_FORMAT_PCM_SUB_8_BIT           = 0x2, /* DO NOT CHANGE */
+} audio_format_pcm_sub_fmt_t;
+
+/* MP3 sub format field definition : can use 11 LSBs in the same way as MP3
+ * frame header to specify bit rate, stereo mode, version...
+ */
+typedef enum {
+    AUDIO_FORMAT_MP3_SUB_NONE            = 0x0,
+} audio_format_mp3_sub_fmt_t;
+
+/* AMR NB/WB sub format field definition: specify frame block interleaving,
+ * bandwidth efficient or octet aligned, encoding mode for recording...
+ */
+typedef enum {
+    AUDIO_FORMAT_AMR_SUB_NONE            = 0x0,
+} audio_format_amr_sub_fmt_t;
+
+/* AAC sub format field definition: specify profile or bitrate for recording... */
+typedef enum {
+    AUDIO_FORMAT_AAC_SUB_NONE            = 0x0,
+} audio_format_aac_sub_fmt_t;
+
+/* VORBIS sub format field definition: specify quality for recording... */
+typedef enum {
+    AUDIO_FORMAT_VORBIS_SUB_NONE         = 0x0,
+} audio_format_vorbis_sub_fmt_t;
+
+/* Audio format consists in a main format field (upper 8 bits) and a sub format
+ * field (lower 24 bits).
+ *
+ * The main format indicates the main codec type. The sub format field
+ * indicates options and parameters for each format. The sub format is mainly
+ * used for record to indicate for instance the requested bitrate or profile.
+ * It can also be used for certain formats to give informations not present in
+ * the encoded audio stream (e.g. octet alignement for AMR).
+ */
+typedef enum {
+    AUDIO_FORMAT_INVALID             = 0xFFFFFFFFUL,
+    AUDIO_FORMAT_DEFAULT             = 0,
+    AUDIO_FORMAT_PCM                 = 0x00000000UL, /* DO NOT CHANGE */
+    AUDIO_FORMAT_MP3                 = 0x01000000UL,
+    AUDIO_FORMAT_AMR_NB              = 0x02000000UL,
+    AUDIO_FORMAT_AMR_WB              = 0x03000000UL,
+    AUDIO_FORMAT_AAC                 = 0x04000000UL,
+    AUDIO_FORMAT_HE_AAC_V1           = 0x05000000UL,
+    AUDIO_FORMAT_HE_AAC_V2           = 0x06000000UL,
+    AUDIO_FORMAT_VORBIS              = 0x07000000UL,
+    AUDIO_FORMAT_MAIN_MASK           = 0xFF000000UL,
+    AUDIO_FORMAT_SUB_MASK            = 0x00FFFFFFUL,
+
+    /* Aliases */
+    AUDIO_FORMAT_PCM_16_BIT          = (AUDIO_FORMAT_PCM |
+                                        AUDIO_FORMAT_PCM_SUB_16_BIT),
+    AUDIO_FORMAT_PCM_8_BIT           = (AUDIO_FORMAT_PCM |
+                                        AUDIO_FORMAT_PCM_SUB_8_BIT),
+} audio_format_t;
+
+/* Channel mask definitions must be kept in sync with JAVA values in
+ * frameworks/base/media/java/android/media/AudioFormat.java */
+typedef enum {
+    /* output channels */
+    AUDIO_CHANNEL_OUT_FRONT_LEFT            = 0x4,
+    AUDIO_CHANNEL_OUT_FRONT_RIGHT           = 0x8,
+    AUDIO_CHANNEL_OUT_FRONT_CENTER          = 0x10,
+    AUDIO_CHANNEL_OUT_LOW_FREQUENCY         = 0x20,
+    AUDIO_CHANNEL_OUT_BACK_LEFT             = 0x40,
+    AUDIO_CHANNEL_OUT_BACK_RIGHT            = 0x80,
+    AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER  = 0x100,
+    AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER = 0x200,
+    AUDIO_CHANNEL_OUT_BACK_CENTER           = 0x400,
+
+    AUDIO_CHANNEL_OUT_MONO     = AUDIO_CHANNEL_OUT_FRONT_LEFT,
+    AUDIO_CHANNEL_OUT_STEREO   = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT),
+    AUDIO_CHANNEL_OUT_QUAD     = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+                                  AUDIO_CHANNEL_OUT_BACK_LEFT |
+                                  AUDIO_CHANNEL_OUT_BACK_RIGHT),
+    AUDIO_CHANNEL_OUT_SURROUND = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+                                  AUDIO_CHANNEL_OUT_FRONT_CENTER |
+                                  AUDIO_CHANNEL_OUT_BACK_CENTER),
+    AUDIO_CHANNEL_OUT_5POINT1  = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+                                  AUDIO_CHANNEL_OUT_FRONT_CENTER |
+                                  AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
+                                  AUDIO_CHANNEL_OUT_BACK_LEFT |
+                                  AUDIO_CHANNEL_OUT_BACK_RIGHT),
+    AUDIO_CHANNEL_OUT_7POINT1  = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+                                  AUDIO_CHANNEL_OUT_FRONT_CENTER |
+                                  AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
+                                  AUDIO_CHANNEL_OUT_BACK_LEFT |
+                                  AUDIO_CHANNEL_OUT_BACK_RIGHT |
+                                  AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER |
+                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER),
+    AUDIO_CHANNEL_OUT_ALL      = (AUDIO_CHANNEL_OUT_FRONT_LEFT |
+                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT |
+                                  AUDIO_CHANNEL_OUT_FRONT_CENTER |
+                                  AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
+                                  AUDIO_CHANNEL_OUT_BACK_LEFT |
+                                  AUDIO_CHANNEL_OUT_BACK_RIGHT |
+                                  AUDIO_CHANNEL_OUT_FRONT_LEFT_OF_CENTER |
+                                  AUDIO_CHANNEL_OUT_FRONT_RIGHT_OF_CENTER |
+                                  AUDIO_CHANNEL_OUT_BACK_CENTER),
+
+    /* input channels */
+    AUDIO_CHANNEL_IN_LEFT            = 0x4,
+    AUDIO_CHANNEL_IN_RIGHT           = 0x8,
+    AUDIO_CHANNEL_IN_FRONT           = 0x10,
+    AUDIO_CHANNEL_IN_BACK            = 0x20,
+    AUDIO_CHANNEL_IN_LEFT_PROCESSED  = 0x40,
+    AUDIO_CHANNEL_IN_RIGHT_PROCESSED = 0x80,
+    AUDIO_CHANNEL_IN_FRONT_PROCESSED = 0x100,
+    AUDIO_CHANNEL_IN_BACK_PROCESSED  = 0x200,
+    AUDIO_CHANNEL_IN_PRESSURE        = 0x400,
+    AUDIO_CHANNEL_IN_X_AXIS          = 0x800,
+    AUDIO_CHANNEL_IN_Y_AXIS          = 0x1000,
+    AUDIO_CHANNEL_IN_Z_AXIS          = 0x2000,
+    AUDIO_CHANNEL_IN_VOICE_UPLINK    = 0x4000,
+    AUDIO_CHANNEL_IN_VOICE_DNLINK    = 0x8000,
+
+    AUDIO_CHANNEL_IN_MONO   = AUDIO_CHANNEL_IN_FRONT,
+    AUDIO_CHANNEL_IN_STEREO = (AUDIO_CHANNEL_IN_LEFT | AUDIO_CHANNEL_IN_RIGHT),
+    AUDIO_CHANNEL_IN_ALL    = (AUDIO_CHANNEL_IN_LEFT |
+                               AUDIO_CHANNEL_IN_RIGHT |
+                               AUDIO_CHANNEL_IN_FRONT |
+                               AUDIO_CHANNEL_IN_BACK|
+                               AUDIO_CHANNEL_IN_LEFT_PROCESSED |
+                               AUDIO_CHANNEL_IN_RIGHT_PROCESSED |
+                               AUDIO_CHANNEL_IN_FRONT_PROCESSED |
+                               AUDIO_CHANNEL_IN_BACK_PROCESSED|
+                               AUDIO_CHANNEL_IN_PRESSURE |
+                               AUDIO_CHANNEL_IN_X_AXIS |
+                               AUDIO_CHANNEL_IN_Y_AXIS |
+                               AUDIO_CHANNEL_IN_Z_AXIS |
+                               AUDIO_CHANNEL_IN_VOICE_UPLINK |
+                               AUDIO_CHANNEL_IN_VOICE_DNLINK),
+} audio_channels_t;
+
+typedef enum {
+    AUDIO_MODE_INVALID          = -2,
+    AUDIO_MODE_CURRENT          = -1,
+    AUDIO_MODE_NORMAL           = 0,
+    AUDIO_MODE_RINGTONE         = 1,
+    AUDIO_MODE_IN_CALL          = 2,
+    AUDIO_MODE_IN_COMMUNICATION = 3,
+
+    AUDIO_MODE_CNT,
+    AUDIO_MODE_MAX              = AUDIO_MODE_CNT - 1,
+} audio_mode_t;
+
+typedef enum {
+    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 enum {
+    /* output devices */
+    AUDIO_DEVICE_OUT_EARPIECE                  = 0x1,
+    AUDIO_DEVICE_OUT_SPEAKER                   = 0x2,
+    AUDIO_DEVICE_OUT_WIRED_HEADSET             = 0x4,
+    AUDIO_DEVICE_OUT_WIRED_HEADPHONE           = 0x8,
+    AUDIO_DEVICE_OUT_BLUETOOTH_SCO             = 0x10,
+    AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET     = 0x20,
+    AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT      = 0x40,
+    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP            = 0x80,
+    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES = 0x100,
+    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER    = 0x200,
+    AUDIO_DEVICE_OUT_AUX_DIGITAL               = 0x400,
+    AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET         = 0x800,
+    AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET         = 0x1000,
+    AUDIO_DEVICE_OUT_DEFAULT                   = 0x8000,
+    AUDIO_DEVICE_OUT_ALL      = (AUDIO_DEVICE_OUT_EARPIECE |
+                                 AUDIO_DEVICE_OUT_SPEAKER |
+                                 AUDIO_DEVICE_OUT_WIRED_HEADSET |
+                                 AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_SCO |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER |
+                                 AUDIO_DEVICE_OUT_AUX_DIGITAL |
+                                 AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
+                                 AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET |
+                                 AUDIO_DEVICE_OUT_DEFAULT),
+    AUDIO_DEVICE_OUT_ALL_A2DP = (AUDIO_DEVICE_OUT_BLUETOOTH_A2DP |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER),
+    AUDIO_DEVICE_OUT_ALL_SCO  = (AUDIO_DEVICE_OUT_BLUETOOTH_SCO |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET |
+                                 AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT),
+
+    /* input devices */
+    AUDIO_DEVICE_IN_COMMUNICATION         = 0x10000,
+    AUDIO_DEVICE_IN_AMBIENT               = 0x20000,
+    AUDIO_DEVICE_IN_BUILTIN_MIC           = 0x40000,
+    AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET = 0x80000,
+    AUDIO_DEVICE_IN_WIRED_HEADSET         = 0x100000,
+    AUDIO_DEVICE_IN_AUX_DIGITAL           = 0x200000,
+    AUDIO_DEVICE_IN_VOICE_CALL            = 0x400000,
+    AUDIO_DEVICE_IN_BACK_MIC              = 0x800000,
+    AUDIO_DEVICE_IN_DEFAULT               = 0x80000000,
+
+    AUDIO_DEVICE_IN_ALL     = (AUDIO_DEVICE_IN_COMMUNICATION |
+                               AUDIO_DEVICE_IN_AMBIENT |
+                               AUDIO_DEVICE_IN_BUILTIN_MIC |
+                               AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET |
+                               AUDIO_DEVICE_IN_WIRED_HEADSET |
+                               AUDIO_DEVICE_IN_AUX_DIGITAL |
+                               AUDIO_DEVICE_IN_VOICE_CALL |
+                               AUDIO_DEVICE_IN_BACK_MIC |
+                               AUDIO_DEVICE_IN_DEFAULT),
+    AUDIO_DEVICE_IN_ALL_SCO = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET,
+} audio_devices_t;
+
+static inline bool audio_is_output_device(audio_devices_t device)
+{
+    if ((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 ((popcount(device) == 1) && ((device & ~AUDIO_DEVICE_IN_ALL) == 0))
+        return true;
+    else
+        return false;
+}
+
+static inline bool audio_is_a2dp_device(audio_devices_t device)
+{
+    if ((popcount(device) == 1) && (device & AUDIO_DEVICE_OUT_ALL_A2DP))
+        return true;
+    else
+        return false;
+}
+
+static inline bool audio_is_bluetooth_sco_device(audio_devices_t device)
+{
+    if ((popcount(device) == 1) && (device & (AUDIO_DEVICE_OUT_ALL_SCO |
+                   AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)))
+        return true;
+    else
+        return false;
+}
+
+static inline bool audio_is_input_channel(uint32_t channel)
+{
+    if ((channel & ~AUDIO_CHANNEL_IN_ALL) == 0)
+        return true;
+    else
+        return false;
+}
+
+static inline bool audio_is_output_channel(uint32_t channel)
+{
+    if ((channel & ~AUDIO_CHANNEL_OUT_ALL) == 0)
+        return true;
+    else
+        return false;
+}
+
+static inline bool audio_is_valid_format(uint32_t format)
+{
+    switch (format & AUDIO_FORMAT_MAIN_MASK) {
+    case AUDIO_FORMAT_PCM:
+    case AUDIO_FORMAT_MP3:
+    case AUDIO_FORMAT_AMR_NB:
+    case AUDIO_FORMAT_AMR_WB:
+    case AUDIO_FORMAT_AAC:
+    case AUDIO_FORMAT_HE_AAC_V1:
+    case AUDIO_FORMAT_HE_AAC_V2:
+    case AUDIO_FORMAT_VORBIS:
+        return true;
+    default:
+        return false;
+    }
+}
+
+static inline bool audio_is_linear_pcm(uint32_t format)
+{
+    switch (format) {
+    case AUDIO_FORMAT_PCM_16_BIT:
+    case AUDIO_FORMAT_PCM_8_BIT:
+        return true;
+    default:
+        return false;
+    }
+}
+
+
+__END_DECLS
+
+#endif  // ANDROID_AUDIO_CORE_H
diff --git a/include/hardware/audio_hal.h b/include/hardware/audio_hal.h
new file mode 100644
index 0000000..50675bd
--- /dev/null
+++ b/include/hardware/audio_hal.h
@@ -0,0 +1,298 @@
+/*
+ * 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 <cutils/bitops.h>
+
+#include <hardware/hardware.h>
+#include <hardware/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"
+
+/**************************************/
+
+/* common audio stream parameters and operations */
+struct audio_stream {
+
+    /**
+     * sampling rate is in Hz - eg. 44100
+     */
+    uint32_t (*get_sample_rate)(const struct audio_stream *stream);
+    int (*set_sample_rate)(struct audio_stream *stream, uint32_t rate);
+
+    /**
+     * size of output buffer in bytes - eg. 4800
+     */
+    size_t (*get_buffer_size)(const struct audio_stream *stream);
+
+    /**
+     * the channel mask -
+     *  e.g. AUDIO_CHANNEL_OUT_STEREO or AUDIO_CHANNEL_IN_STEREO
+     */
+    uint32_t (*get_channels)(const struct audio_stream *stream);
+
+    /**
+     * audio format - eg. AUDIO_FORMAT_PCM_16_BIT
+     */
+    int (*get_format)(const struct audio_stream *stream);
+    int (*set_format)(struct audio_stream *stream, int format);
+
+    /**
+     * Put the audio hardware input/output into standby mode.
+     * 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);
+
+    audio_devices_t (*get_device)(const struct audio_stream *stream);
+    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.
+     */
+    char * (*get_parameters)(const struct audio_stream *stream,
+                             const char *keys);
+};
+typedef struct audio_stream audio_stream_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 {
+    struct audio_stream common;
+
+    /**
+     * return the audio hardware driver latency in milli seconds.
+     */
+    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
+     */
+    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);
+};
+typedef struct audio_stream_out audio_stream_out_t;
+
+struct 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 */
+    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);
+};
+typedef struct audio_stream_in audio_stream_in_t;
+
+/**
+ * return the frame size (number of bytes per sample).
+ */
+static inline uint32_t audio_stream_frame_size(struct audio_stream *s)
+{
+    int chan_samp_sz;
+
+    switch (s->get_format(s)) {
+    case AUDIO_FORMAT_PCM_16_BIT:
+        chan_samp_sz = sizeof(int16_t);
+        break;
+    case AUDIO_FORMAT_PCM_8_BIT:
+    default:
+        chan_samp_sz = sizeof(int8_t);
+        break;
+    }
+
+    return popcount(s->get_channels(s)) * chan_samp_sz;
+}
+
+
+/**********************************************************************/
+
+/**
+ * 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 {
+    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
+     */
+    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);
+
+    /**
+     * setMode 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, int 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.
+     */
+    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
+     */
+    size_t (*get_input_buffer_size)(const struct audio_hw_device *dev,
+                                    uint32_t sample_rate, int format,
+                                    int channel_count);
+
+    /** This method creates and opens the audio hardware output stream */
+    int (*open_output_stream)(struct audio_hw_device *dev, uint32_t devices,
+                              int *format, uint32_t *channels,
+                              uint32_t *sample_rate,
+                              struct audio_stream_out **out);
+
+    void (*close_output_stream)(struct audio_hw_device *dev,
+                                struct audio_stream_out* out);
+
+    /** This method creates and opens the audio hardware input stream */
+    int (*open_input_stream)(struct audio_hw_device *dev, uint32_t devices,
+                             int *format, uint32_t *channels,
+                             uint32_t *sample_rate,
+                             audio_in_acoustics_t acoustics,
+                             struct audio_stream_in **stream_in);
+
+    void (*close_input_stream)(struct audio_hw_device *dev,
+                               struct audio_stream_in *in);
+
+    /** This method dumps the state of the audio hardware */
+    int (*dump)(const struct audio_hw_device *dev, int fd);
+};
+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,
+                                 (struct hw_device_t**)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/include/hardware/audio_policy.h b/include/hardware/audio_policy.h
new file mode 100644
index 0000000..1e0af7d
--- /dev/null
+++ b/include/hardware/audio_policy.h
@@ -0,0 +1,106 @@
+/*
+ * 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_CORE_H
+#define ANDROID_AUDIO_POLICY_CORE_H
+
+#include <stdint.h>
+#include <sys/cdefs.h>
+#include <sys/types.h>
+
+#include <cutils/bitops.h>
+
+__BEGIN_DECLS
+
+/* The enums were moved here mostly from
+ * frameworks/base/include/media/AudioSystem.h
+ */
+
+/* request to open a direct output with get_output() (by opposition to
+ * sharing an output with other AudioTracks)
+ */
+typedef enum {
+    AUDIO_POLICY_OUTPUT_FLAG_INDIRECT = 0x0,
+    AUDIO_POLICY_OUTPUT_FLAG_DIRECT = 0x1
+} audio_policy_output_flags_t;
+
+/* device categories used for audio_policy->set_force_use() */
+typedef enum {
+    AUDIO_POLICY_FORCE_NONE,
+    AUDIO_POLICY_FORCE_SPEAKER,
+    AUDIO_POLICY_FORCE_HEADPHONES,
+    AUDIO_POLICY_FORCE_BT_SCO,
+    AUDIO_POLICY_FORCE_BT_A2DP,
+    AUDIO_POLICY_FORCE_WIRED_ACCESSORY,
+    AUDIO_POLICY_FORCE_BT_CAR_DOCK,
+    AUDIO_POLICY_FORCE_BT_DESK_DOCK,
+    AUDIO_POLICY_FORCE_ANALOG_DOCK,
+    AUDIO_POLICY_FORCE_DIGITAL_DOCK,
+
+    AUDIO_POLICY_FORCE_CFG_CNT,
+    AUDIO_POLICY_FORCE_CFG_MAX = AUDIO_POLICY_FORCE_CFG_CNT - 1,
+
+    AUDIO_POLICY_FORCE_DEFAULT = AUDIO_POLICY_FORCE_NONE,
+} audio_policy_forced_cfg_t;
+
+/* usages used for audio_policy->set_force_use() */
+typedef enum {
+    AUDIO_POLICY_FORCE_FOR_COMMUNICATION,
+    AUDIO_POLICY_FORCE_FOR_MEDIA,
+    AUDIO_POLICY_FORCE_FOR_RECORD,
+    AUDIO_POLICY_FORCE_FOR_DOCK,
+
+    AUDIO_POLICY_FORCE_USE_CNT,
+    AUDIO_POLICY_FORCE_USE_MAX = AUDIO_POLICY_FORCE_USE_CNT - 1,
+} audio_policy_force_use_t;
+
+/* device connection states used for audio_policy->set_device_connection_state()
+ */
+typedef enum {
+    AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+    AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+
+    AUDIO_POLICY_DEVICE_STATE_CNT,
+    AUDIO_POLICY_DEVICE_STATE_MAX = AUDIO_POLICY_DEVICE_STATE_CNT - 1,
+} audio_policy_dev_state_t;
+
+typedef enum {
+    /* Used to generate a tone to notify the user of a
+     * notification/alarm/ringtone while they are in a call. */
+    AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION = 0,
+
+    AUDIO_POLICY_TONE_CNT,
+    AUDIO_POLICY_TONE_MAX                  = AUDIO_POLICY_TONE_CNT - 1,
+} audio_policy_tone_t;
+
+
+static inline bool audio_is_low_visibility(audio_stream_type_t stream)
+{
+    switch (stream) {
+    case AUDIO_STREAM_SYSTEM:
+    case AUDIO_STREAM_NOTIFICATION:
+    case AUDIO_STREAM_RING:
+        return true;
+    default:
+        return false;
+    }
+}
+
+
+__END_DECLS
+
+#endif  // ANDROID_AUDIO_POLICY_CORE_H
diff --git a/include/hardware/audio_policy_hal.h b/include/hardware/audio_policy_hal.h
new file mode 100644
index 0000000..37be011
--- /dev/null
+++ b/include/hardware/audio_policy_hal.h
@@ -0,0 +1,384 @@
+/*
+ * 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 <hardware/audio.h>
+#include <hardware/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);
+
+    /* retreive 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, int state);
+
+    /* 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);
+
+    /* retreive 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 appriate 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,
+                                    uint32_t format,
+                                    uint32_t channels,
+                                    audio_policy_output_flags_t flags);
+
+    /* 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,
+                        int 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,
+                       int session);
+
+    /* releases the output. */
+    void (*release_output)(struct audio_policy *pol, audio_io_handle_t output);
+
+    /* request an input appriate for record from the supplied device with
+     * supplied parameters. */
+    audio_io_handle_t (*get_input)(struct audio_policy *pol, int inputSource,
+                                   uint32_t samplingRate,
+                                   uint32_t format,
+                                   uint32_t channels,
+                                   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. */
+    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 */
+    int (*set_stream_volume_index)(struct audio_policy *pol,
+                                   audio_stream_type_t stream,
+                                   int index);
+
+    /* retreive current volume index for the specified stream */
+    int (*get_stream_volume_index)(const struct audio_policy *pol,
+                                   audio_stream_type_t stream,
+                                   int *index);
+
+    /* 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 */
+    uint32_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,
+                                            struct effect_descriptor_s *desc);
+
+    int (*register_effect)(struct audio_policy *pol,
+                           struct effect_descriptor_s *desc,
+                           audio_io_handle_t output,
+                           uint32_t strategy,
+                           int session,
+                           int id);
+
+    int (*unregister_effect)(struct audio_policy *pol, int id);
+
+    bool (*is_stream_active)(const struct audio_policy *pol,
+                             int stream,
+                             uint32_t in_past_ms);
+
+    /* dump state */
+    int (*dump)(const struct audio_policy *pol, int fd);
+};
+
+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,
+                                     uint32_t *pDevices,
+                                     uint32_t *pSamplingRate,
+                                     uint32_t *pFormat,
+                                     uint32_t *pChannels,
+                                     uint32_t *pLatencyMs,
+                                     audio_policy_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 */
+    audio_io_handle_t (*open_input)(void *service,
+                                    uint32_t *pDevices,
+                                    uint32_t *pSamplingRate,
+                                    uint32_t *pFormat,
+                                    uint32_t *pChannels,
+                                    uint32_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);
+
+    /* reroute a given stream type to the specified output */
+    int (*set_stream_output)(void *service,
+                             audio_stream_type_t stream,
+                             audio_io_handle_t output);
+
+    /* 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.
+     */
+
+    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,
+                        int session,
+                        audio_io_handle_t src_output,
+                        audio_io_handle_t dst_output);
+};
+
+/**********************************************************************/
+
+/**
+ * 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 {
+    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