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ViPERFX_RE/include/hardware/system/audio.h
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/*
* 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 "audio-base.h"
#include "audio-base-utils.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;
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;
/* v4a_print(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);
}
/**
* Expresses the convention when stereo audio samples are stored interleaved
* in an array. This should improve readability by allowing code to use
* symbolic indices instead of hard-coded [0] and [1].
*
* For multi-channel beyond stereo, the platform convention is that channels
* are interleaved in order from least significant channel mask bit to most
* significant channel mask bit, with unused bits skipped. Any exceptions
* to this convention will be noted at the appropriate API.
*/
enum {
AUDIO_INTERLEAVE_LEFT = 0,
AUDIO_INTERLEAVE_RIGHT = 1,
};
/* 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 */
};
/** Metadata of a record track for an in stream. */
typedef struct playback_track_metadata {
audio_usage_t usage;
audio_content_type_t content_type;
float gain; // Normalized linear volume. 0=silence, 1=0dbfs...
} playback_track_metadata_t;
/** Metadata of a playback track for an out stream. */
typedef struct record_track_metadata {
audio_source_t source;
float gain; // Normalized linear volume. 0=silence, 1=0dbfs...
} record_track_metadata_t;
/******************************
* Helper functions
*****************************/
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_hearing_aid_out_device(audio_devices_t device)
{
return device == AUDIO_DEVICE_OUT_HEARING_AID;
}
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 audio_channel_mask_t audio_channel_mask_in_to_out(audio_channel_mask_t in)
{
switch (in) {
case AUDIO_CHANNEL_IN_MONO:
return AUDIO_CHANNEL_OUT_MONO;
case AUDIO_CHANNEL_IN_STEREO:
return AUDIO_CHANNEL_OUT_STEREO;
case AUDIO_CHANNEL_IN_5POINT1:
return AUDIO_CHANNEL_OUT_5POINT1;
case AUDIO_CHANNEL_IN_3POINT1POINT2:
return AUDIO_CHANNEL_OUT_3POINT1POINT2;
case AUDIO_CHANNEL_IN_3POINT0POINT2:
return AUDIO_CHANNEL_OUT_3POINT0POINT2;
case AUDIO_CHANNEL_IN_2POINT1POINT2:
return AUDIO_CHANNEL_OUT_2POINT1POINT2;
case AUDIO_CHANNEL_IN_2POINT0POINT2:
return AUDIO_CHANNEL_OUT_2POINT0POINT2;
default:
return AUDIO_CHANNEL_INVALID;
}
}
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_AAC_ELD:
case AUDIO_FORMAT_AAC_XHE:
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:
case AUDIO_FORMAT_E_AC3_JOC:
case AUDIO_FORMAT_MAT_1_0:
case AUDIO_FORMAT_MAT_2_0:
case AUDIO_FORMAT_MAT_2_1:
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;
}
static inline size_t audio_bytes_per_frame(uint32_t channel_count, audio_format_t format)
{
// cannot overflow for reasonable channel_count
return channel_count * audio_bytes_per_sample(format);
}
/* 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;
//TODO: audio_microphone_location_t need to move to HAL v4.0
typedef enum {
AUDIO_MICROPHONE_LOCATION_UNKNOWN = 0,
AUDIO_MICROPHONE_LOCATION_MAINBODY = 1,
AUDIO_MICROPHONE_LOCATION_MAINBODY_MOVABLE = 2,
AUDIO_MICROPHONE_LOCATION_PERIPHERAL = 3,
AUDIO_MICROPHONE_LOCATION_CNT = 4,
} audio_microphone_location_t;
//TODO: audio_microphone_directionality_t need to move to HAL v4.0
typedef enum {
AUDIO_MICROPHONE_DIRECTIONALITY_UNKNOWN = 0,
AUDIO_MICROPHONE_DIRECTIONALITY_OMNI = 1,
AUDIO_MICROPHONE_DIRECTIONALITY_BI_DIRECTIONAL = 2,
AUDIO_MICROPHONE_DIRECTIONALITY_CARDIOID = 3,
AUDIO_MICROPHONE_DIRECTIONALITY_HYPER_CARDIOID = 4,
AUDIO_MICROPHONE_DIRECTIONALITY_SUPER_CARDIOID = 5,
AUDIO_MICROPHONE_DIRECTIONALITY_CNT = 6,
} audio_microphone_directionality_t;
/* A 3D point which could be used to represent geometric location
* or orientation of a microphone.
*/
struct audio_microphone_coordinate {
float x;
float y;
float z;
};
/* An number to indicate which group the microphone locate. Main body is
* usually group 0. Developer could use this value to group the microphones
* that locate on the same peripheral or attachments.
*/
typedef int audio_microphone_group_t;
typedef enum {
AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED = 0,
AUDIO_MICROPHONE_CHANNEL_MAPPING_DIRECT = 1,
AUDIO_MICROPHONE_CHANNEL_MAPPING_PROCESSED = 2,
AUDIO_MICROPHONE_CHANNEL_MAPPING_CNT = 3,
} audio_microphone_channel_mapping_t;
/* the maximum length for the microphone id */
#define AUDIO_MICROPHONE_ID_MAX_LEN 32
/* max number of frequency responses in a frequency response table */
#define AUDIO_MICROPHONE_MAX_FREQUENCY_RESPONSES 256
/* max number of microphone */
#define AUDIO_MICROPHONE_MAX_COUNT 32
/* the value of unknown spl */
#define AUDIO_MICROPHONE_SPL_UNKNOWN -FLT_MAX
/* the value of unknown sensitivity */
#define AUDIO_MICROPHONE_SENSITIVITY_UNKNOWN -FLT_MAX
/* the value of unknown coordinate */
#define AUDIO_MICROPHONE_COORDINATE_UNKNOWN -FLT_MAX
/* the value used as address when the address of bottom microphone is empty */
#define AUDIO_BOTTOM_MICROPHONE_ADDRESS "bottom"
/* the value used as address when the address of back microphone is empty */
#define AUDIO_BACK_MICROPHONE_ADDRESS "back"
struct audio_microphone_characteristic_t {
char device_id[AUDIO_MICROPHONE_ID_MAX_LEN];
audio_port_handle_t id;
audio_devices_t device;
char address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
audio_microphone_channel_mapping_t channel_mapping[AUDIO_CHANNEL_COUNT_MAX];
audio_microphone_location_t location;
audio_microphone_group_t group;
unsigned int index_in_the_group;
float sensitivity;
float max_spl;
float min_spl;
audio_microphone_directionality_t directionality;
unsigned int num_frequency_responses;
float frequency_responses[2][AUDIO_MICROPHONE_MAX_FREQUENCY_RESPONSES];
struct audio_microphone_coordinate geometric_location;
struct audio_microphone_coordinate orientation;
};
__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"
#define AUDIO_HARDWARE_MODULE_ID_HEARING_AID "hearing_aid"
/**
* 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 */
/* Request the presentation id to be decoded by a next gen audio decoder */
#define AUDIO_PARAMETER_STREAM_PRESENTATION_ID "presentation_id" /* int32_t */
/* Request the program id to be decoded by a next gen audio decoder */
#define AUDIO_PARAMETER_STREAM_PROGRAM_ID "program_id" /* int32_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 "|"
/* Reconfigure offloaded A2DP codec */
#define AUDIO_PARAMETER_RECONFIG_A2DP "reconfigA2dp"
/* Query if HwModule supports reconfiguration of offloaded A2DP codec */
#define AUDIO_PARAMETER_A2DP_RECONFIG_SUPPORTED "isReconfigA2dpSupported"
/**
* 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"
#endif // ANDROID_AUDIO_CORE_H
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